From 45ee5878397822c41b9109d0e3bc2d635c8c8634 Mon Sep 17 00:00:00 2001 From: Luca Sas Date: Thu, 1 Oct 2020 19:17:58 +0100 Subject: Updated to raylib 3.0 --- libs/raylib/src/external/miniaudio.h | 55007 +++++++++++++++++---------------- 1 file changed, 29038 insertions(+), 25969 deletions(-) (limited to 'libs/raylib/src/external/miniaudio.h') diff --git a/libs/raylib/src/external/miniaudio.h b/libs/raylib/src/external/miniaudio.h index 1927d96..7d26cf7 100644 --- a/libs/raylib/src/external/miniaudio.h +++ b/libs/raylib/src/external/miniaudio.h @@ -1,6 +1,6 @@ /* Audio playback and capture library. Choice of public domain or MIT-0. See license statements at the end of this file. -miniaudio (formerly mini_al) - v0.9.8 - 2019-10-07 +miniaudio (formerly mini_al) - v0.xx.xx - 2020-xx-xx David Reid - davidreidsoftware@gmail.com @@ -8,329 +8,385 @@ https://github.com/dr-soft/miniaudio */ /* -MAJOR CHANGES IN VERSION 0.9 +RELEASE NOTES - VERSION 0.10 ============================ -Version 0.9 includes major API changes, centered mostly around full-duplex and the rebrand to "miniaudio". Before I go into -detail about the major changes I would like to apologize. I know it's annoying dealing with breaking API changes, but I think -it's best to get these changes out of the way now while the library is still relatively young and unknown. +Version 0.10 includes major API changes and refactoring, mostly concerned with the data conversion system. Data conversion is performed internally to convert +audio data between the format requested when initializing the `ma_device` object and the format of the internal device used by the backend. The same applies +to the `ma_decoder` object. The previous design has several design flaws and missing features which necessitated a complete redesign. -There's been a lot of refactoring with this release so there's a good chance a few bugs have been introduced. I apologize in -advance for this. You may want to hold off on upgrading for the short term if you're worried. If mini_al v0.8.14 works for -you, and you don't need full-duplex support, you can avoid upgrading (though you won't be getting future bug fixes). +Changes to Data Conversion +-------------------------- +The previous data conversion system used callbacks to deliver input data for conversion. This design works well in some specific situations, but in other +situations it has some major readability and maintenance issues. The decision was made to replace this with a more iterative approach where you just pass in a +pointer to the input data directly rather than dealing with a callback. -Rebranding to "miniaudio" -------------------------- -The decision was made to rename mini_al to miniaudio. Don't worry, it's the same project. The reason for this is simple: +The following are the data conversion APIs that have been removed and their replacements: -1) Having the word "audio" in the title makes it immediately clear that the library is related to audio; and -2) I don't like the look of the underscore. + - ma_format_converter -> ma_convert_pcm_frames_format() + - ma_channel_router -> ma_channel_converter + - ma_src -> ma_resampler + - ma_pcm_converter -> ma_data_converter -This rebrand has necessitated a change in namespace from "mal" to "ma". I know this is annoying, and I apologize, but it's -better to get this out of the road now rather than later. Also, since there are necessary API changes for full-duplex support -I think it's better to just get the namespace change over and done with at the same time as the full-duplex changes. I'm hoping -this will be the last of the major API changes. Fingers crossed! +The previous conversion APIs accepted a callback in their configs. There are no longer any callbacks to deal with. Instead you just pass the data into the +`*_process_pcm_frames()` function as a pointer to a buffer. -The implementation define is now "#define MINIAUDIO_IMPLEMENTATION". You can also use "#define MA_IMPLEMENTATION" if that's -your preference. +The simplest aspect of data conversion is sample format conversion. To convert between two formats, just call `ma_convert_pcm_frames_format()`. Channel +conversion is also simple which you can do with `ma_channel_router` via `ma_channel_router_process_pcm_frames(). +Resampling is more complicated because the number of output frames that are processed is different to the number of input frames that are consumed. When you +call `ma_resampler_process_pcm_frames()` you need to pass in the number of input frames available for processing and the number of output frames you want to +output. Upon returning they will receive the number of input frames that were consumed and the number of output frames that were generated. -Full-Duplex Support -------------------- -The major feature added to version 0.9 is full-duplex. This has necessitated a few API changes. +The `ma_data_converter` API is a wrapper around format, channel and sample rate conversion and handles all of the data conversion you'll need which probably +makes it the best option if you need to do data conversion. -1) The data callback has now changed. Previously there was one type of callback for playback and another for capture. I wanted - to avoid a third callback just for full-duplex so the decision was made to break this API and unify the callbacks. Now, - there is just one callback which is the same for all three modes (playback, capture, duplex). The new callback looks like - the following: +In addition to changes to the API design, a few other changes have been made to the data conversion pipeline: - void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount); + - The sinc resampler has been removed. This was completely broken and never actually worked properly. + - The linear resampler can now uses low-pass filtering to remove aliasing. The quality of the low-pass filter can be controlled via the resampler config with + the `lpfCount` option, which has a maximum value of MA_MAX_RESAMPLER_LPF_FILTERS. + - Data conversion now supports s16 natively which runs through a fixed point pipeline. Previously everything needed to be converted to floating point before + processing, whereas now both s16 and f32 are natively supported. Other formats still require conversion to either s16 or f32 prior to processing, however + `ma_data_converter` will handle this for you. - This callback allows you to move data straight out of the input buffer and into the output buffer in full-duplex mode. In - playback-only mode, pInput will be null. Likewise, pOutput will be null in capture-only mode. The sample count is no longer - returned from the callback since it's not necessary for miniaudio anymore. -2) The device config needed to change in order to support full-duplex. Full-duplex requires the ability to allow the client - to choose a different PCM format for the playback and capture sides. The old ma_device_config object simply did not allow - this and needed to change. With these changes you now specify the device ID, format, channels, channel map and share mode - on a per-playback and per-capture basis (see example below). The sample rate must be the same for playback and capture. +Custom Memory Allocators +------------------------ +miniaudio has always supported macro level customization for memory allocation via MA_MALLOC, MA_REALLOC and MA_FREE, however some scenarios require more +flexibility by allowing a user data pointer to be passed to the custom allocation routines. Support for this has been added to version 0.10 via the +`ma_allocation_callbacks` structure. Anything making use of heap allocations has been updated to accept this new structure. - Since the device config API has changed I have also decided to take the opportunity to simplify device initialization. Now, - the device ID, device type and callback user data are set in the config. ma_device_init() is now simplified down to taking - just the context, device config and a pointer to the device object being initialized. The rationale for this change is that - it just makes more sense to me that these are set as part of the config like everything else. +The `ma_context_config` structure has been updated with a new member called `allocationCallbacks`. Leaving this set to it's defaults returned by +`ma_context_config_init()` will cause it to use MA_MALLOC, MA_REALLOC and MA_FREE. Likewise, The `ma_decoder_config` structure has been updated in the same +way, and leaving everything as-is after `ma_decoder_config_init()` will cause it to use the same defaults. - Example device initialization: +The following APIs have been updated to take a pointer to a `ma_allocation_callbacks` object. Setting this parameter to NULL will cause it to use defaults. +Otherwise they will use the relevant callback in the structure. - ma_device_config config = ma_device_config_init(ma_device_type_duplex); // Or ma_device_type_playback or ma_device_type_capture. - config.playback.pDeviceID = &myPlaybackDeviceID; // Or NULL for the default playback device. - config.playback.format = ma_format_f32; - config.playback.channels = 2; - config.capture.pDeviceID = &myCaptureDeviceID; // Or NULL for the default capture device. - config.capture.format = ma_format_s16; - config.capture.channels = 1; - config.sampleRate = 44100; - config.dataCallback = data_callback; - config.pUserData = &myUserData; + - ma_malloc() + - ma_realloc() + - ma_free() + - ma_aligned_malloc() + - ma_aligned_free() + - ma_rb_init() / ma_rb_init_ex() + - ma_pcm_rb_init() / ma_pcm_rb_init_ex() - result = ma_device_init(&myContext, &config, &device); - if (result != MA_SUCCESS) { - ... handle error ... - } +Note that you can continue to use MA_MALLOC, MA_REALLOC and MA_FREE as per normal. These will continue to be used by default if you do not specify custom +allocation callbacks. - Note that the "onDataCallback" member of ma_device_config has been renamed to "dataCallback". Also, "onStopCallback" has - been renamed to "stopCallback". -This is the first pass for full-duplex and there is a known bug. You will hear crackling on the following backends when sample -rate conversion is required for the playback device: - - Core Audio - - JACK - - AAudio - - OpenSL - - WebAudio +Buffer and Period Configuration Changes +--------------------------------------- +The way in which the size of the internal buffer and periods are specified in the device configuration have changed. In previous versions, the config variables +`bufferSizeInFrames` and `bufferSizeInMilliseconds` defined the size of the entire buffer, with the size of a period being the size of this variable divided by +the period count. This became confusing because people would expect the value of `bufferSizeInFrames` or `bufferSizeInMilliseconds` to independantly determine +latency, when in fact it was that value divided by the period count that determined it. These variables have been removed and replaced with new ones called +`periodSizeInFrames` and `periodSizeInMilliseconds`. -In addition to the above, not all platforms have been absolutely thoroughly tested simply because I lack the hardware for such -thorough testing. If you experience a bug, an issue report on GitHub or an email would be greatly appreciated (and a sample -program that reproduces the issue if possible). +These new configuration variables work in the same way as their predecessors in that if one is set to 0, the other will be used, but the main difference is +that you now set these to you desired latency rather than the size of the entire buffer. The benefit of this is that it's much easier and less confusing to +configure latency. + +The following unused APIs have been removed: + + ma_get_default_buffer_size_in_milliseconds() + ma_get_default_buffer_size_in_frames() + +The following macros have been removed: + + MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_LOW_LATENCY + MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_CONSERVATIVE Other API Changes ----------------- -In addition to the above, the following API changes have been made: +Other less major API changes have also been made in version 0.10. + +`ma_device_set_stop_callback()` has been removed. You now must set the stop callback via the device config just like the data callback. + +The `ma_sine_wave` API has been replaced with a more general API called `ma_waveform`. This supports generation of different types of waveforms, including +sine, square, triangle and sawtooth. Use `ma_waveform_init()` in place of `ma_sine_wave_init()` to initialize the waveform object. This takes the same +parameters, except an additional `ma_waveform_type` value which you would set to `ma_waveform_type_sine`. Use `ma_waveform_read_pcm_frames()` in place of +`ma_sine_wave_read_f32()` and `ma_sine_wave_read_f32_ex()`. + +`ma_convert_frames()` and `ma_convert_frames_ex()` have been changed. Both of these functions now take a new parameter called `frameCountOut` which specifies +the size of the output buffer in PCM frames. This has been added for safety. In addition to this, the parameters for `ma_convert_frames_ex()` have changed to +take a pointer to a `ma_data_converter_config` object to specify the input and output formats to convert between. This was done to make it make it more +flexible, to prevent the parameter list getting too long, and to prevent API breakage whenever a new conversion property is added. + +`ma_calculate_frame_count_after_src()` has been renamed to `ma_calculate_frame_count_after_resampling()` for consistency with the new `ma_resampler` API. -- The log callback is no longer passed to ma_context_config_init(). Instead you need to set it manually after initialization. -- The onLogCallback member of ma_context_config has been renamed to "logCallback". -- The log callback now takes a logLevel parameter. The new callback looks like: void log_callback(ma_context* pContext, ma_device* pDevice, ma_uint32 logLevel, const char* message) - - You can use ma_log_level_to_string() to convert the logLevel to human readable text if you want to log it. -- Some APIs have been renamed: - - mal_decoder_read() -> ma_decoder_read_pcm_frames() - - mal_decoder_seek_to_frame() -> ma_decoder_seek_to_pcm_frame() - - mal_sine_wave_read() -> ma_sine_wave_read_f32() - - mal_sine_wave_read_ex() -> ma_sine_wave_read_f32_ex() -- Some APIs have been removed: - - mal_device_get_buffer_size_in_bytes() - - mal_device_set_recv_callback() - - mal_device_set_send_callback() - - mal_src_set_input_sample_rate() - - mal_src_set_output_sample_rate() -- Error codes have been rearranged. If you're a binding maintainer you will need to update. -- The ma_backend enums have been rearranged to priority order. The rationale for this is to simplify automatic backend selection - and to make it easier to see the priority. If you're a binding maintainer you will need to update. -- ma_dsp has been renamed to ma_pcm_converter. The rationale for this change is that I'm expecting "ma_dsp" to conflict with - some future planned high-level APIs. -- For functions that take a pointer/count combo, such as ma_decoder_read_pcm_frames(), the parameter order has changed so that - the pointer comes before the count. The rationale for this is to keep it consistent with things like memcpy(). + +Biquad and Low-Pass Filters +--------------------------- +A generic biquad filter has been added. This is used via the `ma_biquad` API. The biquad filter is used as the basis for the low-pass filter. The biquad filter +supports 32-bit floating point samples which runs on a floating point pipeline and 16-bit signed integer samples which runs on a 32-bit fixed point pipeline. +Both formats use transposed direct form 2. + +The low-pass filter is just a biquad filter. By itself it's a second order low-pass filter, but it can be extended to higher orders by chaining low-pass +filters together. Low-pass filtering is achieved via the `ma_lpf` API. Since the low-pass filter is just a biquad filter, it supports both 32-bit floating +point and 16-bit signed integer formats. + + +Sine, Square, Triangle and Sawtooth Waveforms +--------------------------------------------- +Previously miniaudio supported only sine wave generation. This has now been generalized to support sine, square, triangle and sawtooth waveforms. The old +`ma_sine_wave` API has been removed and replaced with the `ma_waveform` API. Use `ma_waveform_init()` to initialize the waveform. Here you specify tyhe type of +waveform you want to generated. You then read data using `ma_waveform_read_pcm_frames()`. Miscellaneous Changes --------------------- -The following miscellaneous changes have also been made. +Internal functions have all been made static where possible. If you get warnings about unused functions, please submit a bug report. -- The AAudio backend has been added for Android 8 and above. This is Android's new "High-Performance Audio" API. (For the - record, this is one of the nicest audio APIs out there, just behind the BSD audio APIs). -- The WebAudio backend has been added. This is based on ScriptProcessorNode. This removes the need for SDL. -- The SDL and OpenAL backends have been removed. These were originally implemented to add support for platforms for which miniaudio - was not explicitly supported. These are no longer needed and have therefore been removed. -- Device initialization now fails if the requested share mode is not supported. If you ask for exclusive mode, you either get an - exclusive mode device, or an error. The rationale for this change is to give the client more control over how to handle cases - when the desired shared mode is unavailable. -- A lock-free ring buffer API has been added. There are two varients of this. "ma_rb" operates on bytes, whereas "ma_pcm_rb" - operates on PCM frames. -- The library is now licensed as a choice of Public Domain (Unlicense) _or_ MIT-0 (No Attribution) which is the same as MIT, but - removes the attribution requirement. The rationale for this is to support countries that don't recognize public domain. +The `ma_device` structure is no longer defined as being aligned to MA_SIMD_ALIGNMENT. This resulted in a possible crash when allocating a `ma_device` object on +the heap, but not aligning it to MA_SIMD_ALIGNMENT. This crash would happen due to the compiler seeing the alignment specified on the structure and assuming it +was always aligned as such and thinking it was safe to emit alignment-dependant SIMD instructions. Since miniaudio's philosophy is for things to just work, +this has been removed from all structures. */ + /* -ABOUT -===== -miniaudio is a single file library for audio playback and capture. It's written in C (compilable as -C++) and released into the public domain. - -Supported Backends: - - WASAPI - - DirectSound - - WinMM - - Core Audio (Apple) - - ALSA - - PulseAudio - - JACK - - sndio (OpenBSD) - - audio(4) (NetBSD and OpenBSD) - - OSS (FreeBSD) - - AAudio (Android 8.0+) - - OpenSL|ES (Android only) - - Web Audio (Emscripten) - - Null (Silence) - -Supported Formats: - - Unsigned 8-bit PCM - - Signed 16-bit PCM - - Signed 24-bit PCM (tightly packed) - - Signed 32-bit PCM - - IEEE 32-bit floating point PCM - - -USAGE -===== -miniaudio is a single-file library. To use it, do something like the following in one .c file. - #define MINIAUDIO_IMPLEMENTATION - #include "miniaudio.h" - -You can then #include this file in other parts of the program as you would with any other header file. - -miniaudio uses an asynchronous, callback based API. You initialize a device with a configuration (sample rate, -channel count, etc.) which includes the callback you want to use to handle data transmission to/from the -device. In the callback you either read from a data pointer in the case of playback or write to it in the case -of capture. - -Playback Example ----------------- - void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount) - { - ma_decoder* pDecoder = (ma_decoder*)pDevice->pUserData; - if (pDecoder == NULL) { - return; - } - - ma_decoder_read_pcm_frames(pDecoder, frameCount, pOutput); - } +Introduction +============ +miniaudio is a single file library for audio playback and capture. To use it, do the following in one .c file: - ... + ```c + #define MINIAUDIO_IMPLEMENTATION + #include "miniaudio.h + ``` - ma_device_config config = ma_device_config_init(ma_device_type_playback); - config.playback.format = decoder.outputFormat; - config.playback.channels = decoder.outputChannels; - config.sampleRate = decoder.outputSampleRate; - config.dataCallback = data_callback; - config.pUserData = &decoder; +You can #include miniaudio.h in other parts of the program just like any other header. - ma_device device; - if (ma_device_init(NULL, &config, &device) != MA_SUCCESS) { - ... An error occurred ... - } +miniaudio uses the concept of a "device" as the abstraction for physical devices. The idea is that you choose a physical device to emit or capture audio from, +and then move data to/from the device when miniaudio tells you to. Data is delivered to and from devices asynchronously via a callback which you specify when +initializing the device. - ma_device_start(&device); // The device is sleeping by default so you'll need to start it manually. +When initializing the device you first need to configure it. The device configuration allows you to specify things like the format of the data delivered via +the callback, the size of the internal buffer and the ID of the device you want to emit or capture audio from. - ... +Once you have the device configuration set up you can initialize the device. When initializing a device you need to allocate memory for the device object +beforehand. This gives the application complete control over how the memory is allocated. In the example below we initialize a playback device on the stack, +but you could allocate it on the heap if that suits your situation better. - ma_device_uninit(&device); // This will stop the device so no need to do that manually. + ```c + void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount) + { + // In playback mode copy data to pOutput. In capture mode read data from pInput. In full-duplex mode, both pOutput and pInput will be valid and you can + // move data from pInput into pOutput. Never process more than frameCount frames. + } + ... -BUILDING -======== -miniaudio should Just Work by adding it to your project's source tree. You do not need to download or install -any dependencies. See below for platform-specific details. + ma_device_config config = ma_device_config_init(ma_device_type_playback); + config.playback.format = MY_FORMAT; + config.playback.channels = MY_CHANNEL_COUNT; + config.sampleRate = MY_SAMPLE_RATE; + config.dataCallback = data_callback; + config.pUserData = pMyCustomData; // Can be accessed from the device object (device.pUserData). + + ma_device device; + if (ma_device_init(NULL, &config, &device) != MA_SUCCESS) { + ... An error occurred ... + } + + ma_device_start(&device); // The device is sleeping by default so you'll need to start it manually. + + ... + + ma_device_uninit(&device); // This will stop the device so no need to do that manually. + ``` + +In the example above, `data_callback()` is where audio data is written and read from the device. The idea is in playback mode you cause sound to be emitted +from the speakers by writing audio data to the output buffer (`pOutput` in the example). In capture mode you read data from the input buffer (`pInput`) to +extract sound captured by the microphone. The `frameCount` parameter tells you how many frames can be written to the output buffer and read from the input +buffer. A "frame" is one sample for each channel. For example, in a stereo stream (2 channels), one frame is 2 samples: one for the left, one for the right. +The channel count is defined by the device config. The size in bytes of an individual sample is defined by the sample format which is also specified in the +device config. Multi-channel audio data is always interleaved, which means the samples for each frame are stored next to each other in memory. For example, in +a stereo stream the first pair of samples will be the left and right samples for the first frame, the second pair of samples will be the left and right samples +for the second frame, etc. + +The configuration of the device is defined by the `ma_device_config` structure. The config object is always initialized with `ma_device_config_init()`. It's +important to always initialize the config with this function as it initializes it with logical defaults and ensures your program doesn't break when new members +are added to the `ma_device_config` structure. The example above uses a fairly simple and standard device configuration. The call to `ma_device_config_init()` +takes a single parameter, which is whether or not the device is a playback, capture, duplex or loopback device (loopback devices are not supported on all +backends). The `config.playback.format` member sets the sample format which can be one of the following (all formats are native-endian): + + |---------------|----------------------------------------|---------------------------| + | Symbol | Description | Range | + |---------------|----------------------------------------|---------------------------| + | ma_format_f32 | 32-bit floating point | [-1, 1] | + | ma_format_s16 | 16-bit signed integer | [-32768, 32767] | + | ma_format_s24 | 24-bit signed integer (tightly packed) | [-8388608, 8388607] | + | ma_format_s32 | 32-bit signed integer | [-2147483648, 2147483647] | + | ma_format_u8 | 8-bit unsigned integer | [0, 255] | + |---------------|----------------------------------------|---------------------------| + +The `config.playback.channels` member sets the number of channels to use with the device. The channel count cannot exceed MA_MAX_CHANNELS. The +`config.sampleRate` member sets the sample rate (which must be the same for both playback and capture in full-duplex configurations). This is usually set to +44100 or 48000, but can be set to anything. It's recommended to keep this between 8000 and 384000, however. + +Note that leaving the format, channel count and/or sample rate at their default values will result in the internal device's native configuration being used +which is useful if you want to avoid the overhead of miniaudio's automatic data conversion. + +In addition to the sample format, channel count and sample rate, the data callback and user data pointer are also set via the config. The user data pointer is +not passed into the callback as a parameter, but is instead set to the `pUserData` member of `ma_device` which you can access directly since all miniaudio +structures are transparent. + +Initializing the device is done with `ma_device_init()`. This will return a result code telling you what went wrong, if anything. On success it will return +`MA_SUCCESS`. After initialization is complete the device will be in a stopped state. To start it, use `ma_device_start()`. Uninitializing the device will stop +it, which is what the example above does, but you can also stop the device with `ma_device_stop()`. To resume the device simply call `ma_device_start()` again. +Note that it's important to never stop or start the device from inside the callback. This will result in a deadlock. Instead you set a variable or signal an +event indicating that the device needs to stop and handle it in a different thread. The following APIs must never be called inside the callback: + + ma_device_init() + ma_device_init_ex() + ma_device_uninit() + ma_device_start() + ma_device_stop() + +You must never try uninitializing and reinitializing a device inside the callback. You must also never try to stop and start it from inside the callback. There +are a few other things you shouldn't do in the callback depending on your requirements, however this isn't so much a thread-safety thing, but rather a real- +time processing thing which is beyond the scope of this introduction. + +The example above demonstrates the initialization of a playback device, but it works exactly the same for capture. All you need to do is change the device type +from `ma_device_type_playback` to `ma_device_type_capture` when setting up the config, like so: + + ```c + ma_device_config config = ma_device_config_init(ma_device_type_capture); + config.capture.format = MY_FORMAT; + config.capture.channels = MY_CHANNELS; + ``` + +In the data callback you just read from the input buffer (`pInput` in the example above) and leave the output buffer alone (it will be set to NULL when the +device type is set to `ma_device_type_capture`). + +These are the available device types and how you should handle the buffers in the callback: + + |-------------------------|--------------------------------------------------------| + | Device Type | Callback Behavior | + |-------------------------|--------------------------------------------------------| + | ma_device_type_playback | Write to output buffer, leave input buffer untouched. | + | ma_device_type_capture | Read from input buffer, leave output buffer untouched. | + | ma_device_type_duplex | Read from input buffer, write to output buffer. | + | ma_device_type_loopback | Read from input buffer, leave output buffer untouched. | + |-------------------------|--------------------------------------------------------| + +You will notice in the example above that the sample format and channel count is specified separately for playback and capture. This is to support different +data formats between the playback and capture devices in a full-duplex system. An example may be that you want to capture audio data as a monaural stream (one +channel), but output sound to a stereo speaker system. Note that if you use different formats between playback and capture in a full-duplex configuration you +will need to convert the data yourself. There are functions available to help you do this which will be explained later. + +The example above did not specify a physical device to connect to which means it will use the operating system's default device. If you have multiple physical +devices connected and you want to use a specific one you will need to specify the device ID in the configuration, like so: + + ``` + config.playback.pDeviceID = pMyPlaybackDeviceID; // Only if requesting a playback or duplex device. + config.capture.pDeviceID = pMyCaptureDeviceID; // Only if requesting a capture, duplex or loopback device. + ``` + +To retrieve the device ID you will need to perform device enumeration, however this requires the use of a new concept call the "context". Conceptually speaking +the context sits above the device. There is one context to many devices. The purpose of the context is to represent the backend at a more global level and to +perform operations outside the scope of an individual device. Mainly it is used for performing run-time linking against backend libraries, initializing +backends and enumerating devices. The example below shows how to enumerate devices. + + ```c + ma_context context; + if (ma_context_init(NULL, 0, NULL, &context) != MA_SUCCESS) { + // Error. + } + + ma_device_info* pPlaybackDeviceInfos; + ma_uint32 playbackDeviceCount; + ma_device_info* pCaptureDeviceInfos; + ma_uint32 captureDeviceCount; + if (ma_context_get_devices(&context, &pPlaybackDeviceInfos, &playbackDeviceCount, &pCaptureDeviceInfos, &captureDeviceCount) != MA_SUCCESS) { + // Error. + } + + // Loop over the each device info and do something with it. Here we just print the name with their index. You may want to give the user the + // opportunity to choose which device they'd prefer. + for (ma_uint32 iDevice = 0; iDevice < playbackDeviceCount; iDevice += 1) { + printf("%d - %s\n", iDevice, pPlaybackDeviceInfos[iDevice].name); + } -If you want to disable a specific backend, #define the appropriate MA_NO_* option before the implementation. + ma_device_config config = ma_device_config_init(ma_device_type_playback); + config.playback.pDeviceID = &pPlaybackDeviceInfos[chosenPlaybackDeviceIndex].id; + config.playback.format = MY_FORMAT; + config.playback.channels = MY_CHANNEL_COUNT; + config.sampleRate = MY_SAMPLE_RATE; + config.dataCallback = data_callback; + config.pUserData = pMyCustomData; -Note that GCC and Clang requires "-msse2", "-mavx2", etc. for SIMD optimizations. + ma_device device; + if (ma_device_init(&context, &config, &device) != MA_SUCCESS) { + // Error + } + ... -Building for Windows --------------------- -The Windows build should compile clean on all popular compilers without the need to configure any include paths -nor link to any libraries. + ma_device_uninit(&device); + ma_context_uninit(&context); + ``` -Building for macOS and iOS --------------------------- -The macOS build should compile clean without the need to download any dependencies or link to any libraries or -frameworks. The iOS build needs to be compiled as Objective-C (sorry) and will need to link the relevant frameworks -but should Just Work with Xcode. +The first thing we do in this example is initialize a `ma_context` object with `ma_context_init()`. The first parameter is a pointer to a list of `ma_backend` +values which are used to override the default backend priorities. When this is NULL, as in this example, miniaudio's default priorities are used. The second +parameter is the number of backends listed in the array pointed to by the first paramter. The third parameter is a pointer to a `ma_context_config` object +which can be NULL, in which case defaults are used. The context configuration is used for setting the logging callback, custom memory allocation callbacks, +user-defined data and some backend-specific configurations. -Building for Linux ------------------- -The Linux build only requires linking to -ldl, -lpthread and -lm. You do not need any development packages. +Once the context has been initialized you can enumerate devices. In the example above we use the simpler `ma_context_get_devices()`, however you can also use a +callback for handling devices by using `ma_context_enumerate_devices()`. When using `ma_context_get_devices()` you provide a pointer to a pointer that will, +upon output, be set to a pointer to a buffer containing a list of `ma_device_info` structures. You also provide a pointer to an unsigned integer that will +receive the number of items in the returned buffer. Do not free the returned buffers as their memory is managed internally by miniaudio. -Building for BSD ----------------- -The BSD build only requires linking to -ldl, -lpthread and -lm. NetBSD uses audio(4), OpenBSD uses sndio and -FreeBSD uses OSS. - -Building for Android --------------------- -AAudio is the highest priority backend on Android. This should work out out of the box without needing any kind of -compiler configuration. Support for AAudio starts with Android 8 which means older versions will fall back to -OpenSL|ES which requires API level 16+. - -Building for Emscripten ------------------------ -The Emscripten build emits Web Audio JavaScript directly and should Just Work without any configuration. - - -NOTES -===== -- This library uses an asynchronous API for delivering and requesting audio data. Each device will have - it's own worker thread which is managed by the library. -- If ma_device_init() is called with a device that's not aligned to the 4 bytes on 32-bit or 8 bytes on - 64-bit it will _not_ be thread-safe. The reason for this is that it depends on members of ma_device being - correctly aligned for atomic assignments. -- Sample data is always native-endian and interleaved. For example, ma_format_s16 means signed 16-bit - integer samples, interleaved. Let me know if you need non-interleaved and I'll look into it. -- The sndio backend is currently only enabled on OpenBSD builds. -- The audio(4) backend is supported on OpenBSD, but you may need to disable sndiod before you can use it. -- Automatic stream routing is enabled on a per-backend basis. Support is explicitly enabled for WASAPI - and Core Audio, however other backends such as PulseAudio may naturally support it, though not all have - been tested. -- The contents of the output buffer passed into the data callback will always be pre-initialized to zero - unless the noPreZeroedOutputBuffer config variable in ma_device_config is set to true, in which case - it'll be undefined which will require you to write something to the entire buffer. -- By default miniaudio will automatically clip samples. This only applies when the playback sample format - is configured as ma_format_f32. If you are doing clipping yourself, you can disable this overhead by - setting noClip to true in the device config. - - -BACKEND NUANCES -=============== +The `ma_device_info` structure contains an `id` member which is the ID you pass to the device config. It also contains the name of the device which is useful +for presenting a list of devices to the user via the UI. -WASAPI ------- -- Low-latency shared mode will be disabled when using an application-defined sample rate which is different to the - device's native sample rate. To work around this, set wasapi.noAutoConvertSRC to true in the device config. This - is due to IAudioClient3_InitializeSharedAudioStream() failing when the AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM flag is - specified. Setting wasapi.noAutoConvertSRC will result in miniaudio's lower quality internal resampler being used - instead which will in turn enable the use of low-latency shared mode. +When creating your own context you will want to pass it to `ma_device_init()` when initializing the device. Passing in NULL, like we do in the first example, +will result in miniaudio creating the context for you, which you don't want to do since you've already created a context. Note that internally the context is +only tracked by it's pointer which means you must not change the location of the `ma_context` object. If this is an issue, consider using `malloc()` to +allocate memory for the context. -PulseAudio ----------- -- If you experience bad glitching/noise on Arch Linux, consider this fix from the Arch wiki: - https://wiki.archlinux.org/index.php/PulseAudio/Troubleshooting#Glitches,_skips_or_crackling - Alternatively, consider using a different backend such as ALSA. -Android + +Building +======== +miniaudio should work cleanly out of the box without the need to download or install any dependencies. See below for platform-specific details. + + +Windows ------- -- To capture audio on Android, remember to add the RECORD_AUDIO permission to your manifest: - -- With OpenSL|ES, only a single ma_context can be active at any given time. This is due to a limitation with OpenSL|ES. -- With AAudio, only default devices are enumerated. This is due to AAudio not having an enumeration API (devices are - enumerated through Java). You can however perform your own device enumeration through Java and then set the ID in the - ma_device_id structure (ma_device_id.aaudio) and pass it to ma_device_init(). -- The backend API will perform resampling where possible. The reason for this as opposed to using miniaudio's built-in - resampler is to take advantage of any potential device-specific optimizations the driver may implement. +The Windows build should compile clean on all popular compilers without the need to configure any include paths nor link to any libraries. -UWP +macOS and iOS +------------- +The macOS build should compile clean without the need to download any dependencies or link to any libraries or frameworks. The iOS build needs to be compiled +as Objective-C (sorry) and will need to link the relevant frameworks but should Just Work with Xcode. Compiling through the command line requires linking to +-lpthread and -lm. + +Linux +----- +The Linux build only requires linking to -ldl, -lpthread and -lm. You do not need any development packages. + +BSD --- -- UWP only supports default playback and capture devices. -- UWP requires the Microphone capability to be enabled in the application's manifest (Package.appxmanifest): - - ... - - - - +The BSD build only requires linking to -lpthread and -lm. NetBSD uses audio(4), OpenBSD uses sndio and FreeBSD uses OSS. -Web Audio / Emscripten ----------------------- -- The first time a context is initialized it will create a global object called "miniaudio" whose primary purpose is to act - as a factory for device objects. -- Currently the Web Audio backend uses ScriptProcessorNode's, but this may need to change later as they've been deprecated. -- Google is implementing a policy in their browsers that prevent automatic media output without first receiving some kind - of user input. See here for details: https://developers.google.com/web/updates/2017/09/autoplay-policy-changes. Starting - the device may fail if you try to start playback without first handling some kind of user input. +Android +------- +AAudio is the highest priority backend on Android. This should work out out of the box without needing any kind of compiler configuration. Support for AAudio +starts with Android 8 which means older versions will fall back to OpenSL|ES which requires API level 16+. + +Emscripten +---------- +The Emscripten build emits Web Audio JavaScript directly and should Just Work without any configuration. You cannot use -std=c* compiler flags, nor -ansi. -OPTIONS -======= -#define these options before including this file. +Build Options +------------- +#define these options before including miniaudio.h. #define MA_NO_WASAPI Disables the WASAPI backend. @@ -374,20 +430,12 @@ OPTIONS #define MA_NO_NULL Disables the null backend. -#define MA_DEFAULT_PERIODS - When a period count of 0 is specified when a device is initialized, it will default to this. - -#define MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_LOW_LATENCY -#define MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_CONSERVATIVE - When a buffer size of 0 is specified when a device is initialized it will default to a buffer of this size, depending - on the chosen performance profile. These can be increased or decreased depending on your specific requirements. - #define MA_NO_DECODING Disables the decoding APIs. #define MA_NO_DEVICE_IO - Disables playback and recording. This will disable ma_context and ma_device APIs. This is useful if you only want to - use miniaudio's data conversion and/or decoding APIs. + Disables playback and recording. This will disable ma_context and ma_device APIs. This is useful if you only want to use miniaudio's data conversion and/or + decoding APIs. #define MA_NO_STDIO Disables file IO APIs. @@ -418,10 +466,11 @@ OPTIONS Windows only. The value to pass to internal calls to CoInitializeEx(). Defaults to COINIT_MULTITHREADED. -DEFINITIONS + +Definitions =========== -This section defines common terms used throughout miniaudio. Unfortunately there is often ambiguity in the use of terms -throughout the audio space, so this section is intended to clarify how miniaudio uses each term. +This section defines common terms used throughout miniaudio. Unfortunately there is often ambiguity in the use of terms throughout the audio space, so this +section is intended to clarify how miniaudio uses each term. Sample ------ @@ -429,1661 +478,2324 @@ A sample is a single unit of audio data. If the sample format is f32, then one s Frame / PCM Frame ----------------- -A frame is a groups of samples equal to the number of channels. For a stereo stream a frame is 2 samples, a mono frame -is 1 sample, a 5.1 surround sound frame is 6 samples, etc. The terms "frame" and "PCM frame" are the same thing in -miniaudio. Note that this is different to a compressed frame. If ever miniaudio needs to refer to a compressed frame, such -as a FLAC frame, it will always clarify what it's referring to with something like "FLAC frame" or whatnot. +A frame is a groups of samples equal to the number of channels. For a stereo stream a frame is 2 samples, a mono frame is 1 sample, a 5.1 surround sound frame +is 6 samples, etc. The terms "frame" and "PCM frame" are the same thing in miniaudio. Note that this is different to a compressed frame. If ever miniaudio +needs to refer to a compressed frame, such as a FLAC frame, it will always clarify what it's referring to with something like "FLAC frame" or whatnot. Channel ------- -A stream of monaural audio that is emitted from an individual speaker in a speaker system, or received from an individual -microphone in a microphone system. A stereo stream has two channels (a left channel, and a right channel), a 5.1 surround -sound system has 6 channels, etc. Some audio systems refer to a channel as a complex audio stream that's mixed with other -channels to produce the final mix - this is completely different to miniaudio's use of the term "channel" and should not be -confused. +A stream of monaural audio that is emitted from an individual speaker in a speaker system, or received from an individual microphone in a microphone system. A +stereo stream has two channels (a left channel, and a right channel), a 5.1 surround sound system has 6 channels, etc. Some audio systems refer to a channel as +a complex audio stream that's mixed with other channels to produce the final mix - this is completely different to miniaudio's use of the term "channel" and +should not be confused. Sample Rate ----------- -The sample rate in miniaudio is always expressed in Hz, such as 44100, 48000, etc. It's the number of PCM frames that are -processed per second. +The sample rate in miniaudio is always expressed in Hz, such as 44100, 48000, etc. It's the number of PCM frames that are processed per second. Formats ------- Throughout miniaudio you will see references to different sample formats: - Symbol | Description | Range - -------|----------------------------------------|--------------------------- - u8 | Unsigned 8-bit integer | [0, 255] - s16 | Signed 16-bit integer | [-32768, 32767] - s24 | Signed 24-bit integer (tightly packed) | [-8388608, 8388607] - s32 | Signed 32-bit integer | [-2147483648, 2147483647] - f32 | 32-bit floating point | [-1, 1] + |---------------|----------------------------------------|---------------------------| + | Symbol | Description | Range | + |---------------|----------------------------------------|---------------------------| + | ma_format_f32 | 32-bit floating point | [-1, 1] | + | ma_format_s16 | 16-bit signed integer | [-32768, 32767] | + | ma_format_s24 | 24-bit signed integer (tightly packed) | [-8388608, 8388607] | + | ma_format_s32 | 32-bit signed integer | [-2147483648, 2147483647] | + | ma_format_u8 | 8-bit unsigned integer | [0, 255] | + |---------------|----------------------------------------|---------------------------| All formats are native-endian. -*/ -#ifndef miniaudio_h -#define miniaudio_h -#ifdef __cplusplus -extern "C" { -#endif -#if defined(_MSC_VER) && !defined(__clang__) - #pragma warning(push) - #pragma warning(disable:4201) /* nonstandard extension used: nameless struct/union */ - #pragma warning(disable:4324) /* structure was padded due to alignment specifier */ -#else - #pragma GCC diagnostic push - #pragma GCC diagnostic ignored "-Wpedantic" /* For ISO C99 doesn't support unnamed structs/unions [-Wpedantic] */ - #if defined(__clang__) - #pragma GCC diagnostic ignored "-Wc11-extensions" /* anonymous unions are a C11 extension */ - #endif -#endif +Decoding +======== +The `ma_decoder` API is used for reading audio files. To enable a decoder you must #include the header of the relevant backend library before the +implementation of miniaudio. You can find copies of these in the "extras" folder in the miniaudio repository (https://github.com/dr-soft/miniaudio). -/* Platform/backend detection. */ -#ifdef _WIN32 - #define MA_WIN32 - #if defined(WINAPI_FAMILY) && (WINAPI_FAMILY == WINAPI_FAMILY_PC_APP || WINAPI_FAMILY == WINAPI_FAMILY_PHONE_APP) - #define MA_WIN32_UWP - #else - #define MA_WIN32_DESKTOP - #endif -#else - #define MA_POSIX - #include /* Unfortunate #include, but needed for pthread_t, pthread_mutex_t and pthread_cond_t types. */ +The table below are the supported decoding backends: - #ifdef __unix__ - #define MA_UNIX - #if defined(__DragonFly__) || defined(__FreeBSD__) || defined(__NetBSD__) || defined(__OpenBSD__) - #define MA_BSD - #endif - #endif - #ifdef __linux__ - #define MA_LINUX - #endif - #ifdef __APPLE__ - #define MA_APPLE - #endif - #ifdef __ANDROID__ - #define MA_ANDROID - #endif - #ifdef __EMSCRIPTEN__ - #define MA_EMSCRIPTEN - #endif -#endif + |--------|-----------------| + | Type | Backend Library | + |--------|-----------------| + | WAV | dr_wav.h | + | FLAC | dr_flac.h | + | MP3 | dr_mp3.h | + | Vorbis | stb_vorbis.c | + |--------|-----------------| -#include /* For size_t. */ +The code below is an example of how to enable decoding backends: -/* Sized types. Prefer built-in types. Fall back to stdint. */ -#ifdef _MSC_VER - #if defined(__clang__) - #pragma GCC diagnostic push - #pragma GCC diagnostic ignored "-Wlanguage-extension-token" - #pragma GCC diagnostic ignored "-Wlong-long" - #pragma GCC diagnostic ignored "-Wc++11-long-long" - #endif - typedef signed __int8 ma_int8; - typedef unsigned __int8 ma_uint8; - typedef signed __int16 ma_int16; - typedef unsigned __int16 ma_uint16; - typedef signed __int32 ma_int32; - typedef unsigned __int32 ma_uint32; - typedef signed __int64 ma_int64; - typedef unsigned __int64 ma_uint64; - #if defined(__clang__) - #pragma GCC diagnostic pop - #endif -#else - #define MA_HAS_STDINT - #include - typedef int8_t ma_int8; - typedef uint8_t ma_uint8; - typedef int16_t ma_int16; - typedef uint16_t ma_uint16; - typedef int32_t ma_int32; - typedef uint32_t ma_uint32; - typedef int64_t ma_int64; - typedef uint64_t ma_uint64; -#endif + ```c + #include "dr_flac.h" // Enables FLAC decoding. + #include "dr_mp3.h" // Enables MP3 decoding. + #include "dr_wav.h" // Enables WAV decoding. -#ifdef MA_HAS_STDINT - typedef uintptr_t ma_uintptr; -#else - #if defined(_WIN32) - #if defined(_WIN64) - typedef ma_uint64 ma_uintptr; - #else - typedef ma_uint32 ma_uintptr; - #endif - #elif defined(__GNUC__) - #if defined(__LP64__) - typedef ma_uint64 ma_uintptr; - #else - typedef ma_uint32 ma_uintptr; - #endif - #else - typedef ma_uint64 ma_uintptr; /* Fallback. */ - #endif -#endif + #define MINIAUDIO_IMPLEMENTATION + #include "miniaudio.h" + ``` -typedef ma_uint8 ma_bool8; -typedef ma_uint32 ma_bool32; -#define MA_TRUE 1 -#define MA_FALSE 0 +A decoder can be initialized from a file with `ma_decoder_init_file()`, a block of memory with `ma_decoder_init_memory()`, or from data delivered via callbacks +with `ma_decoder_init()`. Here is an example for loading a decoder from a file: -typedef void* ma_handle; -typedef void* ma_ptr; -typedef void (* ma_proc)(void); + ```c + ma_decoder decoder; + ma_result result = ma_decoder_init_file("MySong.mp3", NULL, &decoder); + if (result != MA_SUCCESS) { + return false; // An error occurred. + } -#if defined(_MSC_VER) && !defined(_WCHAR_T_DEFINED) -typedef ma_uint16 wchar_t; -#endif + ... -/* Define NULL for some compilers. */ -#ifndef NULL -#define NULL 0 -#endif + ma_decoder_uninit(&decoder); + ``` -#if defined(SIZE_MAX) - #define MA_SIZE_MAX SIZE_MAX -#else - #define MA_SIZE_MAX 0xFFFFFFFF /* When SIZE_MAX is not defined by the standard library just default to the maximum 32-bit unsigned integer. */ -#endif +When initializing a decoder, you can optionally pass in a pointer to a ma_decoder_config object (the NULL argument in the example above) which allows you to +configure the output format, channel count, sample rate and channel map: + ```c + ma_decoder_config config = ma_decoder_config_init(ma_format_f32, 2, 48000); + ``` -#ifdef _MSC_VER - #define MA_INLINE __forceinline -#elif defined(__GNUC__) - /* - I've had a bug report where GCC is emitting warnings about functions possibly not being inlineable. This warning happens when - the __attribute__((always_inline)) attribute is defined without an "inline" statement. I think therefore there must be some - case where "__inline__" is not always defined, thus the compiler emitting these warnings. When using -std=c89 or -ansi on the - command line, we cannot use the "inline" keyword and instead need to use "__inline__". In an attempt to work around this issue - I am using "__inline__" only when we're compiling in strict ANSI mode. - */ - #if defined(__STRICT_ANSI__) - #define MA_INLINE __inline__ __attribute__((always_inline)) - #else - #define MA_INLINE inline __attribute__((always_inline)) - #endif -#else - #define MA_INLINE -#endif +When passing in NULL for decoder config in `ma_decoder_init*()`, the output format will be the same as that defined by the decoding backend. -#if defined(_MSC_VER) - #if _MSC_VER >= 1400 - #define MA_ALIGN(alignment) __declspec(align(alignment)) - #endif -#elif !defined(__DMC__) - #define MA_ALIGN(alignment) __attribute__((aligned(alignment))) -#endif -#ifndef MA_ALIGN - #define MA_ALIGN(alignment) -#endif +Data is read from the decoder as PCM frames: -#ifdef _MSC_VER -#define MA_ALIGNED_STRUCT(alignment) MA_ALIGN(alignment) struct -#else -#define MA_ALIGNED_STRUCT(alignment) struct MA_ALIGN(alignment) -#endif + ```c + ma_uint64 framesRead = ma_decoder_read_pcm_frames(pDecoder, pFrames, framesToRead); + ``` -/* SIMD alignment in bytes. Currently set to 64 bytes in preparation for future AVX-512 optimizations. */ -#define MA_SIMD_ALIGNMENT 64 +You can also seek to a specific frame like so: + ```c + ma_result result = ma_decoder_seek_to_pcm_frame(pDecoder, targetFrame); + if (result != MA_SUCCESS) { + return false; // An error occurred. + } + ``` -/* Logging levels */ -#define MA_LOG_LEVEL_VERBOSE 4 -#define MA_LOG_LEVEL_INFO 3 -#define MA_LOG_LEVEL_WARNING 2 -#define MA_LOG_LEVEL_ERROR 1 +When loading a decoder, miniaudio uses a trial and error technique to find the appropriate decoding backend. This can be unnecessarily inefficient if the type +is already known. In this case you can use the `_wav`, `_mp3`, etc. varients of the aforementioned initialization APIs: -#ifndef MA_LOG_LEVEL -#define MA_LOG_LEVEL MA_LOG_LEVEL_ERROR -#endif + ```c + ma_decoder_init_wav() + ma_decoder_init_mp3() + ma_decoder_init_memory_wav() + ma_decoder_init_memory_mp3() + ma_decoder_init_file_wav() + ma_decoder_init_file_mp3() + etc. + ``` -typedef struct ma_context ma_context; -typedef struct ma_device ma_device; +The `ma_decoder_init_file()` API will try using the file extension to determine which decoding backend to prefer. -typedef ma_uint8 ma_channel; -#define MA_CHANNEL_NONE 0 -#define MA_CHANNEL_MONO 1 -#define MA_CHANNEL_FRONT_LEFT 2 -#define MA_CHANNEL_FRONT_RIGHT 3 -#define MA_CHANNEL_FRONT_CENTER 4 -#define MA_CHANNEL_LFE 5 -#define MA_CHANNEL_BACK_LEFT 6 -#define MA_CHANNEL_BACK_RIGHT 7 -#define MA_CHANNEL_FRONT_LEFT_CENTER 8 -#define MA_CHANNEL_FRONT_RIGHT_CENTER 9 -#define MA_CHANNEL_BACK_CENTER 10 -#define MA_CHANNEL_SIDE_LEFT 11 -#define MA_CHANNEL_SIDE_RIGHT 12 -#define MA_CHANNEL_TOP_CENTER 13 -#define MA_CHANNEL_TOP_FRONT_LEFT 14 -#define MA_CHANNEL_TOP_FRONT_CENTER 15 -#define MA_CHANNEL_TOP_FRONT_RIGHT 16 -#define MA_CHANNEL_TOP_BACK_LEFT 17 -#define MA_CHANNEL_TOP_BACK_CENTER 18 -#define MA_CHANNEL_TOP_BACK_RIGHT 19 -#define MA_CHANNEL_AUX_0 20 -#define MA_CHANNEL_AUX_1 21 -#define MA_CHANNEL_AUX_2 22 -#define MA_CHANNEL_AUX_3 23 -#define MA_CHANNEL_AUX_4 24 -#define MA_CHANNEL_AUX_5 25 -#define MA_CHANNEL_AUX_6 26 -#define MA_CHANNEL_AUX_7 27 -#define MA_CHANNEL_AUX_8 28 -#define MA_CHANNEL_AUX_9 29 -#define MA_CHANNEL_AUX_10 30 -#define MA_CHANNEL_AUX_11 31 -#define MA_CHANNEL_AUX_12 32 -#define MA_CHANNEL_AUX_13 33 -#define MA_CHANNEL_AUX_14 34 -#define MA_CHANNEL_AUX_15 35 -#define MA_CHANNEL_AUX_16 36 -#define MA_CHANNEL_AUX_17 37 -#define MA_CHANNEL_AUX_18 38 -#define MA_CHANNEL_AUX_19 39 -#define MA_CHANNEL_AUX_20 40 -#define MA_CHANNEL_AUX_21 41 -#define MA_CHANNEL_AUX_22 42 -#define MA_CHANNEL_AUX_23 43 -#define MA_CHANNEL_AUX_24 44 -#define MA_CHANNEL_AUX_25 45 -#define MA_CHANNEL_AUX_26 46 -#define MA_CHANNEL_AUX_27 47 -#define MA_CHANNEL_AUX_28 48 -#define MA_CHANNEL_AUX_29 49 -#define MA_CHANNEL_AUX_30 50 -#define MA_CHANNEL_AUX_31 51 -#define MA_CHANNEL_LEFT MA_CHANNEL_FRONT_LEFT -#define MA_CHANNEL_RIGHT MA_CHANNEL_FRONT_RIGHT -#define MA_CHANNEL_POSITION_COUNT MA_CHANNEL_AUX_31 + 1 -typedef int ma_result; -#define MA_SUCCESS 0 +Sample Format Conversion +======================== +Conversion between sample formats is achieved with the `ma_pcm_*_to_*()`, `ma_pcm_convert()` and `ma_convert_pcm_frames_format()` APIs. Use `ma_pcm_*_to_*()` +to convert between two specific formats. Use `ma_pcm_convert()` to convert based on a `ma_format` variable. Use `ma_convert_pcm_frames_format()` to convert +PCM frames where you want to specify the frame count and channel count as a variable instead of the total sample count. -/* General errors. */ -#define MA_ERROR -1 /* A generic error. */ -#define MA_INVALID_ARGS -2 -#define MA_INVALID_OPERATION -3 -#define MA_OUT_OF_MEMORY -4 -#define MA_ACCESS_DENIED -5 -#define MA_TOO_LARGE -6 -#define MA_TIMEOUT -7 +Dithering +--------- +Dithering can be set using ditherMode parmater. -/* General miniaudio-specific errors. */ -#define MA_FORMAT_NOT_SUPPORTED -100 -#define MA_DEVICE_TYPE_NOT_SUPPORTED -101 -#define MA_SHARE_MODE_NOT_SUPPORTED -102 -#define MA_NO_BACKEND -103 -#define MA_NO_DEVICE -104 -#define MA_API_NOT_FOUND -105 -#define MA_INVALID_DEVICE_CONFIG -106 +The different dithering modes include the following, in order of efficiency: -/* State errors. */ -#define MA_DEVICE_BUSY -200 -#define MA_DEVICE_NOT_INITIALIZED -201 -#define MA_DEVICE_NOT_STARTED -202 -#define MA_DEVICE_UNAVAILABLE -203 + |-----------|--------------------------| + | Type | Enum Token | + |-----------|--------------------------| + | None | ma_dither_mode_none | + | Rectangle | ma_dither_mode_rectangle | + | Triangle | ma_dither_mode_triangle | + |-----------|--------------------------| -/* Operation errors. */ -#define MA_FAILED_TO_MAP_DEVICE_BUFFER -300 -#define MA_FAILED_TO_UNMAP_DEVICE_BUFFER -301 -#define MA_FAILED_TO_INIT_BACKEND -302 -#define MA_FAILED_TO_READ_DATA_FROM_CLIENT -303 -#define MA_FAILED_TO_READ_DATA_FROM_DEVICE -304 -#define MA_FAILED_TO_SEND_DATA_TO_CLIENT -305 -#define MA_FAILED_TO_SEND_DATA_TO_DEVICE -306 -#define MA_FAILED_TO_OPEN_BACKEND_DEVICE -307 -#define MA_FAILED_TO_START_BACKEND_DEVICE -308 -#define MA_FAILED_TO_STOP_BACKEND_DEVICE -309 -#define MA_FAILED_TO_CONFIGURE_BACKEND_DEVICE -310 -#define MA_FAILED_TO_CREATE_MUTEX -311 -#define MA_FAILED_TO_CREATE_EVENT -312 -#define MA_FAILED_TO_CREATE_THREAD -313 +Note that even if the dither mode is set to something other than `ma_dither_mode_none`, it will be ignored for conversions where dithering is not needed. +Dithering is available for the following conversions: + s16 -> u8 + s24 -> u8 + s32 -> u8 + f32 -> u8 + s24 -> s16 + s32 -> s16 + f32 -> s16 -/* Standard sample rates. */ -#define MA_SAMPLE_RATE_8000 8000 -#define MA_SAMPLE_RATE_11025 11025 -#define MA_SAMPLE_RATE_16000 16000 -#define MA_SAMPLE_RATE_22050 22050 -#define MA_SAMPLE_RATE_24000 24000 -#define MA_SAMPLE_RATE_32000 32000 -#define MA_SAMPLE_RATE_44100 44100 -#define MA_SAMPLE_RATE_48000 48000 -#define MA_SAMPLE_RATE_88200 88200 -#define MA_SAMPLE_RATE_96000 96000 -#define MA_SAMPLE_RATE_176400 176400 -#define MA_SAMPLE_RATE_192000 192000 -#define MA_SAMPLE_RATE_352800 352800 -#define MA_SAMPLE_RATE_384000 384000 +Note that it is not an error to pass something other than ma_dither_mode_none for conversions where dither is not used. It will just be ignored. -#define MA_MIN_PCM_SAMPLE_SIZE_IN_BYTES 1 /* For simplicity, miniaudio does not support PCM samples that are not byte aligned. */ -#define MA_MAX_PCM_SAMPLE_SIZE_IN_BYTES 8 -#define MA_MIN_CHANNELS 1 -#define MA_MAX_CHANNELS 32 -#define MA_MIN_SAMPLE_RATE MA_SAMPLE_RATE_8000 -#define MA_MAX_SAMPLE_RATE MA_SAMPLE_RATE_384000 -#define MA_SRC_SINC_MIN_WINDOW_WIDTH 2 -#define MA_SRC_SINC_MAX_WINDOW_WIDTH 32 -#define MA_SRC_SINC_DEFAULT_WINDOW_WIDTH 32 -#define MA_SRC_SINC_LOOKUP_TABLE_RESOLUTION 8 -#define MA_SRC_INPUT_BUFFER_SIZE_IN_SAMPLES 256 -typedef enum -{ - ma_stream_format_pcm = 0 -} ma_stream_format; -typedef enum -{ - ma_stream_layout_interleaved = 0, - ma_stream_layout_deinterleaved -} ma_stream_layout; +Channel Conversion +================== +Channel conversion is used for channel rearrangement and conversion from one channel count to another. The `ma_channel_converter` API is used for channel +conversion. Below is an example of initializing a simple channel converter which converts from mono to stereo. -typedef enum -{ - ma_dither_mode_none = 0, - ma_dither_mode_rectangle, - ma_dither_mode_triangle -} ma_dither_mode; + ```c + ma_channel_converter_config config = ma_channel_converter_config_init(ma_format, 1, NULL, 2, NULL, ma_channel_mix_mode_default, NULL); + result = ma_channel_converter_init(&config, &converter); + if (result != MA_SUCCESS) { + // Error. + } + ``` -typedef enum -{ - /* - I like to keep these explicitly defined because they're used as a key into a lookup table. When items are - added to this, make sure there are no gaps and that they're added to the lookup table in ma_get_bytes_per_sample(). - */ - ma_format_unknown = 0, /* Mainly used for indicating an error, but also used as the default for the output format for decoders. */ - ma_format_u8 = 1, - ma_format_s16 = 2, /* Seems to be the most widely supported format. */ - ma_format_s24 = 3, /* Tightly packed. 3 bytes per sample. */ - ma_format_s32 = 4, - ma_format_f32 = 5, - ma_format_count -} ma_format; +To process perform the conversion simply call `ma_channel_converter_process_pcm_frames()` like so: -typedef enum -{ - ma_channel_mix_mode_rectangular = 0, /* Simple averaging based on the plane(s) the channel is sitting on. */ - ma_channel_mix_mode_simple, /* Drop excess channels; zeroed out extra channels. */ - ma_channel_mix_mode_custom_weights, /* Use custom weights specified in ma_channel_router_config. */ - ma_channel_mix_mode_planar_blend = ma_channel_mix_mode_rectangular, - ma_channel_mix_mode_default = ma_channel_mix_mode_planar_blend -} ma_channel_mix_mode; + ```c + ma_result result = ma_channel_converter_process_pcm_frames(&converter, pFramesOut, pFramesIn, frameCount); + if (result != MA_SUCCESS) { + // Error. + } + ``` + +It is up to the caller to ensure the output buffer is large enough to accomodate the new PCM frames. + +The only formats supported are `ma_format_s16` and `ma_format_f32`. If you need another format you need to convert your data manually which you can do with +`ma_pcm_convert()`, etc. + +Input and output PCM frames are always interleaved. Deinterleaved layouts are not supported. + + +Channel Mapping +--------------- +In addition to converting from one channel count to another, like the example above, The channel converter can also be used to rearrange channels. When +initializing the channel converter, you can optionally pass in channel maps for both the input and output frames. If the channel counts are the same, and each +channel map contains the same channel positions with the exception that they're in a different order, a simple shuffling of the channels with be performed. If, +however, there is not a 1:1 mapping of channel positions, or the channel counts differ, the input channels will be mixed based on a mixing +mode which is specified when initializing the `ma_channel_converter_config` object. + +When converting from mono to multi-channel, the mono channel is simply copied to each output channel. When going the other way around, the audio of each output +channel is simply averaged and copied to the mono channel. + +In more complicated cases blending is used. The `ma_channel_mix_mode_simple` mode will drop excess channels and silence extra channels. For example, converting +from 4 to 2 channels, the 3rd and 4th channels will be dropped, whereas converting from 2 to 4 channels will put silence into the 3rd and 4th channels. + +The `ma_channel_mix_mode_rectangle` mode uses spacial locality based on a rectangle to compute a simple distribution between input and output. Imagine sitting +in the middle of a room, with speakers on the walls representing channel positions. The MA_CHANNEL_FRONT_LEFT position can be thought of as being in the corner +of the front and left walls. + +Finally, the `ma_channel_mix_mode_custom_weights` mode can be used to use custom user-defined weights. Custom weights can be passed in as the last parameter of +`ma_channel_converter_config_init()`. + +Predefined channel maps can be retrieved with `ma_get_standard_channel_map()`. This takes a `ma_standard_channel_map` enum as it's first parameter, which can +be one of the following: + + |-----------------------------------|-----------------------------------------------------------| + | Name | Description | + |-----------------------------------|-----------------------------------------------------------| + | ma_standard_channel_map_default | Default channel map used by miniaudio. See below. | + | ma_standard_channel_map_microsoft | Channel map used by Microsoft's bitfield channel maps. | + | ma_standard_channel_map_alsa | Default ALSA channel map. | + | ma_standard_channel_map_rfc3551 | RFC 3551. Based on AIFF. | + | ma_standard_channel_map_flac | FLAC channel map. | + | ma_standard_channel_map_vorbis | Vorbis channel map. | + | ma_standard_channel_map_sound4 | FreeBSD's sound(4). | + | ma_standard_channel_map_sndio | sndio channel map. www.sndio.org/tips.html | + | ma_standard_channel_map_webaudio | https://webaudio.github.io/web-audio-api/#ChannelOrdering | + |-----------------------------------|-----------------------------------------------------------| + +Below are the channel maps used by default in miniaudio (ma_standard_channel_map_default): + + |---------------|------------------------------| + | Channel Count | Mapping | + |---------------|------------------------------| + | 1 (Mono) | 0: MA_CHANNEL_MONO | + |---------------|------------------------------| + | 2 (Stereo) | 0: MA_CHANNEL_FRONT_LEFT | + | | 1: MA_CHANNEL_FRONT_RIGHT | + |---------------|------------------------------| + | 3 | 0: MA_CHANNEL_FRONT_LEFT | + | | 1: MA_CHANNEL_FRONT_RIGHT | + | | 2: MA_CHANNEL_FRONT_CENTER | + |---------------|------------------------------| + | 4 (Surround) | 0: MA_CHANNEL_FRONT_LEFT | + | | 1: MA_CHANNEL_FRONT_RIGHT | + | | 2: MA_CHANNEL_FRONT_CENTER | + | | 3: MA_CHANNEL_BACK_CENTER | + |---------------|------------------------------| + | 5 | 0: MA_CHANNEL_FRONT_LEFT | + | | 1: MA_CHANNEL_FRONT_RIGHT | + | | 2: MA_CHANNEL_FRONT_CENTER | + | | 3: MA_CHANNEL_BACK_LEFT | + | | 4: MA_CHANNEL_BACK_RIGHT | + |---------------|------------------------------| + | 6 (5.1) | 0: MA_CHANNEL_FRONT_LEFT | + | | 1: MA_CHANNEL_FRONT_RIGHT | + | | 2: MA_CHANNEL_FRONT_CENTER | + | | 3: MA_CHANNEL_LFE | + | | 4: MA_CHANNEL_SIDE_LEFT | + | | 5: MA_CHANNEL_SIDE_RIGHT | + |---------------|------------------------------| + | 7 | 0: MA_CHANNEL_FRONT_LEFT | + | | 1: MA_CHANNEL_FRONT_RIGHT | + | | 2: MA_CHANNEL_FRONT_CENTER | + | | 3: MA_CHANNEL_LFE | + | | 4: MA_CHANNEL_BACK_CENTER | + | | 4: MA_CHANNEL_SIDE_LEFT | + | | 5: MA_CHANNEL_SIDE_RIGHT | + |---------------|------------------------------| + | 8 (7.1) | 0: MA_CHANNEL_FRONT_LEFT | + | | 1: MA_CHANNEL_FRONT_RIGHT | + | | 2: MA_CHANNEL_FRONT_CENTER | + | | 3: MA_CHANNEL_LFE | + | | 4: MA_CHANNEL_BACK_LEFT | + | | 5: MA_CHANNEL_BACK_RIGHT | + | | 6: MA_CHANNEL_SIDE_LEFT | + | | 7: MA_CHANNEL_SIDE_RIGHT | + |---------------|------------------------------| + | Other | All channels set to 0. This | + | | is equivalent to the same | + | | mapping as the device. | + |---------------|------------------------------| + + + +Resampling +========== +Resampling is achieved with the `ma_resampler` object. To create a resampler object, do something like the following: -typedef enum -{ - ma_standard_channel_map_microsoft, - ma_standard_channel_map_alsa, - ma_standard_channel_map_rfc3551, /* Based off AIFF. */ - ma_standard_channel_map_flac, - ma_standard_channel_map_vorbis, - ma_standard_channel_map_sound4, /* FreeBSD's sound(4). */ - ma_standard_channel_map_sndio, /* www.sndio.org/tips.html */ - ma_standard_channel_map_webaudio = ma_standard_channel_map_flac, /* https://webaudio.github.io/web-audio-api/#ChannelOrdering. Only 1, 2, 4 and 6 channels are defined, but can fill in the gaps with logical assumptions. */ - ma_standard_channel_map_default = ma_standard_channel_map_microsoft -} ma_standard_channel_map; + ```c + ma_resampler_config config = ma_resampler_config_init(ma_format_s16, channels, sampleRateIn, sampleRateOut, ma_resample_algorithm_linear); + ma_resampler resampler; + ma_result result = ma_resampler_init(&config, &resampler); + if (result != MA_SUCCESS) { + // An error occurred... + } + ``` -typedef enum -{ - ma_performance_profile_low_latency = 0, - ma_performance_profile_conservative -} ma_performance_profile; +Do the following to uninitialize the resampler: + ```c + ma_resampler_uninit(&resampler); + ``` -typedef struct ma_format_converter ma_format_converter; -typedef ma_uint32 (* ma_format_converter_read_proc) (ma_format_converter* pConverter, ma_uint32 frameCount, void* pFramesOut, void* pUserData); -typedef ma_uint32 (* ma_format_converter_read_deinterleaved_proc)(ma_format_converter* pConverter, ma_uint32 frameCount, void** ppSamplesOut, void* pUserData); - -typedef struct -{ - ma_format formatIn; - ma_format formatOut; - ma_uint32 channels; - ma_stream_format streamFormatIn; - ma_stream_format streamFormatOut; - ma_dither_mode ditherMode; - ma_bool32 noSSE2 : 1; - ma_bool32 noAVX2 : 1; - ma_bool32 noAVX512 : 1; - ma_bool32 noNEON : 1; - ma_format_converter_read_proc onRead; - ma_format_converter_read_deinterleaved_proc onReadDeinterleaved; - void* pUserData; -} ma_format_converter_config; - -struct ma_format_converter -{ - ma_format_converter_config config; - ma_bool32 useSSE2 : 1; - ma_bool32 useAVX2 : 1; - ma_bool32 useAVX512 : 1; - ma_bool32 useNEON : 1; - void (* onConvertPCM)(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode); - void (* onInterleavePCM)(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels); - void (* onDeinterleavePCM)(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels); -}; +The following example shows how data can be processed + ```c + ma_uint64 frameCountIn = 1000; + ma_uint64 frameCountOut = 2000; + ma_result result = ma_resampler_process_pcm_frames(&resampler, pFramesIn, &frameCountIn, pFramesOut, &frameCountOut); + if (result != MA_SUCCESS) { + // An error occurred... + } + // At this point, frameCountIn contains the number of input frames that were consumed and frameCountOut contains the number of output frames written. + ``` -typedef struct ma_channel_router ma_channel_router; -typedef ma_uint32 (* ma_channel_router_read_deinterleaved_proc)(ma_channel_router* pRouter, ma_uint32 frameCount, void** ppSamplesOut, void* pUserData); +To initialize the resampler you first need to set up a config (`ma_resampler_config`) with `ma_resampler_config_init()`. You need to specify the sample format +you want to use, the number of channels, the input and output sample rate, and the algorithm. -typedef struct -{ - ma_uint32 channelsIn; - ma_uint32 channelsOut; - ma_channel channelMapIn[MA_MAX_CHANNELS]; - ma_channel channelMapOut[MA_MAX_CHANNELS]; - ma_channel_mix_mode mixingMode; - float weights[MA_MAX_CHANNELS][MA_MAX_CHANNELS]; /* [in][out]. Only used when mixingMode is set to ma_channel_mix_mode_custom_weights. */ - ma_bool32 noSSE2 : 1; - ma_bool32 noAVX2 : 1; - ma_bool32 noAVX512 : 1; - ma_bool32 noNEON : 1; - ma_channel_router_read_deinterleaved_proc onReadDeinterleaved; - void* pUserData; -} ma_channel_router_config; +The sample format can be either `ma_format_s16` or `ma_format_f32`. If you need a different format you will need to perform pre- and post-conversions yourself +where necessary. Note that the format is the same for both input and output. The format cannot be changed after initialization. -struct ma_channel_router -{ - ma_channel_router_config config; - ma_bool32 isPassthrough : 1; - ma_bool32 isSimpleShuffle : 1; - ma_bool32 isSimpleMonoExpansion : 1; - ma_bool32 isStereoToMono : 1; - ma_bool32 useSSE2 : 1; - ma_bool32 useAVX2 : 1; - ma_bool32 useAVX512 : 1; - ma_bool32 useNEON : 1; - ma_uint8 shuffleTable[MA_MAX_CHANNELS]; -}; +The resampler supports multiple channels and is always interleaved (both input and output). The channel count cannot be changed after initialization. +The sample rates can be anything other than zero, and are always specified in hertz. They should be set to something like 44100, etc. The sample rate is the +only configuration property that can be changed after initialization. +The miniaudio resampler supports multiple algorithms: -typedef struct ma_src ma_src; -typedef ma_uint32 (* ma_src_read_deinterleaved_proc)(ma_src* pSRC, ma_uint32 frameCount, void** ppSamplesOut, void* pUserData); /* Returns the number of frames that were read. */ + |-----------|------------------------------| + | Algorithm | Enum Token | + |-----------|------------------------------| + | Linear | ma_resample_algorithm_linear | + | Speex | ma_resample_algorithm_speex | + |-----------|------------------------------| -typedef enum -{ - ma_src_algorithm_linear = 0, - ma_src_algorithm_sinc, - ma_src_algorithm_none, - ma_src_algorithm_default = ma_src_algorithm_linear -} ma_src_algorithm; +Because Speex is not public domain it is strictly opt-in and the code is stored in separate files. if you opt-in to the Speex backend you will need to consider +it's license, the text of which can be found in it's source files in "extras/speex_resampler". Details on how to opt-in to the Speex resampler is explained in +the Speex Resampler section below. -typedef enum -{ - ma_src_sinc_window_function_hann = 0, - ma_src_sinc_window_function_rectangular, - ma_src_sinc_window_function_default = ma_src_sinc_window_function_hann -} ma_src_sinc_window_function; +The algorithm cannot be changed after initialization. -typedef struct -{ - ma_src_sinc_window_function windowFunction; - ma_uint32 windowWidth; -} ma_src_config_sinc; +Processing always happens on a per PCM frame basis and always assumes interleaved input and output. De-interleaved processing is not supported. To process +frames, use `ma_resampler_process_pcm_frames()`. On input, this function takes the number of output frames you can fit in the output buffer and the number of +input frames contained in the input buffer. On output these variables contain the number of output frames that were written to the output buffer and the +number of input frames that were consumed in the process. You can pass in NULL for the input buffer in which case it will be treated as an infinitely large +buffer of zeros. The output buffer can also be NULL, in which case the processing will be treated as seek. -typedef struct -{ - ma_uint32 sampleRateIn; - ma_uint32 sampleRateOut; - ma_uint32 channels; - ma_src_algorithm algorithm; - ma_bool32 neverConsumeEndOfInput : 1; - ma_bool32 noSSE2 : 1; - ma_bool32 noAVX2 : 1; - ma_bool32 noAVX512 : 1; - ma_bool32 noNEON : 1; - ma_src_read_deinterleaved_proc onReadDeinterleaved; - void* pUserData; - ma_src_config_sinc sinc; -} ma_src_config; +The sample rate can be changed dynamically on the fly. You can change this with explicit sample rates with `ma_resampler_set_rate()` and also with a decimal +ratio with `ma_resampler_set_rate_ratio()`. The ratio is in/out. -MA_ALIGNED_STRUCT(MA_SIMD_ALIGNMENT) ma_src -{ - union - { - struct - { - MA_ALIGN(MA_SIMD_ALIGNMENT) float input[MA_MAX_CHANNELS][MA_SRC_INPUT_BUFFER_SIZE_IN_SAMPLES]; - float timeIn; - ma_uint32 leftoverFrames; - } linear; +Sometimes it's useful to know exactly how many input frames will be required to output a specific number of frames. You can calculate this with +`ma_resampler_get_required_input_frame_count()`. Likewise, it's sometimes useful to know exactly how many frames would be output given a certain number of +input frames. You can do this with `ma_resampler_get_expected_output_frame_count()`. - struct - { - MA_ALIGN(MA_SIMD_ALIGNMENT) float input[MA_MAX_CHANNELS][MA_SRC_SINC_MAX_WINDOW_WIDTH*2 + MA_SRC_INPUT_BUFFER_SIZE_IN_SAMPLES]; - float timeIn; - ma_uint32 inputFrameCount; /* The number of frames sitting in the input buffer, not including the first half of the window. */ - ma_uint32 windowPosInSamples; /* An offset of . */ - float table[MA_SRC_SINC_MAX_WINDOW_WIDTH*1 * MA_SRC_SINC_LOOKUP_TABLE_RESOLUTION]; /* Precomputed lookup table. The +1 is used to avoid the need for an overflow check. */ - } sinc; - }; +Due to the nature of how resampling works, the resampler introduces some latency. This can be retrieved in terms of both the input rate and the output rate +with `ma_resampler_get_input_latency()` and `ma_resampler_get_output_latency()`. - ma_src_config config; - ma_bool32 isEndOfInputLoaded : 1; - ma_bool32 useSSE2 : 1; - ma_bool32 useAVX2 : 1; - ma_bool32 useAVX512 : 1; - ma_bool32 useNEON : 1; -}; -typedef struct ma_pcm_converter ma_pcm_converter; -typedef ma_uint32 (* ma_pcm_converter_read_proc)(ma_pcm_converter* pDSP, void* pFramesOut, ma_uint32 frameCount, void* pUserData); +Resampling Algorithms +--------------------- +The choice of resampling algorithm depends on your situation and requirements. The linear resampler is the most efficient and has the least amount of latency, +but at the expense of poorer quality. The Speex resampler is higher quality, but slower with more latency. It also performs several heap applications +internally for memory management. -typedef struct -{ - ma_format formatIn; - ma_uint32 channelsIn; - ma_uint32 sampleRateIn; - ma_channel channelMapIn[MA_MAX_CHANNELS]; - ma_format formatOut; - ma_uint32 channelsOut; - ma_uint32 sampleRateOut; - ma_channel channelMapOut[MA_MAX_CHANNELS]; - ma_channel_mix_mode channelMixMode; - ma_dither_mode ditherMode; - ma_src_algorithm srcAlgorithm; - ma_bool32 allowDynamicSampleRate; - ma_bool32 neverConsumeEndOfInput : 1; /* <-- For SRC. */ - ma_bool32 noSSE2 : 1; - ma_bool32 noAVX2 : 1; - ma_bool32 noAVX512 : 1; - ma_bool32 noNEON : 1; - ma_pcm_converter_read_proc onRead; - void* pUserData; - union - { - ma_src_config_sinc sinc; - }; -} ma_pcm_converter_config; -MA_ALIGNED_STRUCT(MA_SIMD_ALIGNMENT) ma_pcm_converter -{ - ma_pcm_converter_read_proc onRead; - void* pUserData; - ma_format_converter formatConverterIn; /* For converting data to f32 in preparation for further processing. */ - ma_format_converter formatConverterOut; /* For converting data to the requested output format. Used as the final step in the processing pipeline. */ - ma_channel_router channelRouter; /* For channel conversion. */ - ma_src src; /* For sample rate conversion. */ - ma_bool32 isDynamicSampleRateAllowed : 1; /* ma_pcm_converter_set_input_sample_rate() and ma_pcm_converter_set_output_sample_rate() will fail if this is set to false. */ - ma_bool32 isPreFormatConversionRequired : 1; - ma_bool32 isPostFormatConversionRequired : 1; - ma_bool32 isChannelRoutingRequired : 1; - ma_bool32 isSRCRequired : 1; - ma_bool32 isChannelRoutingAtStart : 1; - ma_bool32 isPassthrough : 1; /* <-- Will be set to true when the conversion pipeline is an optimized passthrough. */ -}; +Linear Resampling +----------------- +The linear resampler is the fastest, but comes at the expense of poorer quality. There is, however, some control over the quality of the linear resampler which +may make it a suitable option depending on your requirements. +The linear resampler performs low-pass filtering before or after downsampling or upsampling, depending on the sample rates you're converting between. When +decreasing the sample rate, the low-pass filter will be applied before downsampling. When increasing the rate it will be performed after upsampling. By default +a second order low-pass filter will be applied. To improve quality you can chain low-pass filters together, up to a maximum of `MA_MAX_RESAMPLER_LPF_FILTERS`. +This comes at the expense of increased computational cost and latency. You can also disable filtering altogether by setting the filter count to 0. The filter +count is controlled with the `lpfCount` config variable. -/************************************************************************************************************************************************************ -************************************************************************************************************************************************************* +The low-pass filter has a cutoff frequency which defaults to half the sample rate of the lowest of the input and output sample rates (Nyquist Frequency). This +can be controlled with the `lpfNyquistFactor` config variable. This defaults to 1, and should be in the range of 0..1, although a value of 0 does not make +sense and should be avoided. A value of 1 will use the Nyquist Frequency as the cutoff. A value of 0.5 will use half the Nyquist Frequency as the cutoff, etc. +Values less than 1 will result in more washed out sound due to more of the higher frequencies being removed. This config variable has no impact on performance +and is a purely perceptual configuration. -DATA CONVERSION -=============== +The API for the linear resampler is the same as the main resampler API, only it's called `ma_linear_resampler`. -This section contains the APIs for data conversion. You will find everything here for channel mapping, sample format conversion, resampling, etc. -************************************************************************************************************************************************************* -************************************************************************************************************************************************************/ +Speex Resampling +---------------- +The Speex resampler is made up of third party code which is released under the BSD license. Because it is licensed differently to miniaudio, which is public +domain, it is strictly opt-in and all of it's code is stored in separate files. If you opt-in to the Speex resampler you must consider the license text in it's +source files. To opt-in, you must first #include the following file before the implementation of miniaudio.h: -/************************************************************************************************************************************************************ + #include "extras/speex_resampler/ma_speex_resampler.h" -Channel Maps -============ +Both the header and implementation is contained within the same file. To implementation can be included in your program like so: -Below is the channel map used by ma_standard_channel_map_default: - -|---------------|------------------------------| -| Channel Count | Mapping | -|---------------|------------------------------| -| 1 (Mono) | 0: MA_CHANNEL_MONO | -|---------------|------------------------------| -| 2 (Stereo) | 0: MA_CHANNEL_FRONT_LEFT | -| | 1: MA_CHANNEL_FRONT_RIGHT | -|---------------|------------------------------| -| 3 | 0: MA_CHANNEL_FRONT_LEFT | -| | 1: MA_CHANNEL_FRONT_RIGHT | -| | 2: MA_CHANNEL_FRONT_CENTER | -|---------------|------------------------------| -| 4 (Surround) | 0: MA_CHANNEL_FRONT_LEFT | -| | 1: MA_CHANNEL_FRONT_RIGHT | -| | 2: MA_CHANNEL_FRONT_CENTER | -| | 3: MA_CHANNEL_BACK_CENTER | -|---------------|------------------------------| -| 5 | 0: MA_CHANNEL_FRONT_LEFT | -| | 1: MA_CHANNEL_FRONT_RIGHT | -| | 2: MA_CHANNEL_FRONT_CENTER | -| | 3: MA_CHANNEL_BACK_LEFT | -| | 4: MA_CHANNEL_BACK_RIGHT | -|---------------|------------------------------| -| 6 (5.1) | 0: MA_CHANNEL_FRONT_LEFT | -| | 1: MA_CHANNEL_FRONT_RIGHT | -| | 2: MA_CHANNEL_FRONT_CENTER | -| | 3: MA_CHANNEL_LFE | -| | 4: MA_CHANNEL_SIDE_LEFT | -| | 5: MA_CHANNEL_SIDE_RIGHT | -|---------------|------------------------------| -| 7 | 0: MA_CHANNEL_FRONT_LEFT | -| | 1: MA_CHANNEL_FRONT_RIGHT | -| | 2: MA_CHANNEL_FRONT_CENTER | -| | 3: MA_CHANNEL_LFE | -| | 4: MA_CHANNEL_BACK_CENTER | -| | 4: MA_CHANNEL_SIDE_LEFT | -| | 5: MA_CHANNEL_SIDE_RIGHT | -|---------------|------------------------------| -| 8 (7.1) | 0: MA_CHANNEL_FRONT_LEFT | -| | 1: MA_CHANNEL_FRONT_RIGHT | -| | 2: MA_CHANNEL_FRONT_CENTER | -| | 3: MA_CHANNEL_LFE | -| | 4: MA_CHANNEL_BACK_LEFT | -| | 5: MA_CHANNEL_BACK_RIGHT | -| | 6: MA_CHANNEL_SIDE_LEFT | -| | 7: MA_CHANNEL_SIDE_RIGHT | -|---------------|------------------------------| -| Other | All channels set to 0. This | -| | is equivalent to the same | -| | mapping as the device. | -|---------------|------------------------------| + #define MINIAUDIO_SPEEX_RESAMPLER_IMPLEMENTATION + #include "extras/speex_resampler/ma_speex_resampler.h" -************************************************************************************************************************************************************/ +Note that even if you opt-in to the Speex backend, miniaudio won't use it unless you explicitly ask for it in the respective config of the object you are +initializing. If you try to use the Speex resampler without opting in, initialization of the `ma_resampler` object will fail with `MA_NO_BACKEND`. -/* -Helper for retrieving a standard channel map. -*/ -void ma_get_standard_channel_map(ma_standard_channel_map standardChannelMap, ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]); +The only configuration option to consider with the Speex resampler is the `speex.quality` config variable. This is a value between 0 and 10, with 0 being +the worst/fastest and 10 being the best/slowest. The default value is 3. -/* -Copies a channel map. -*/ -void ma_channel_map_copy(ma_channel* pOut, const ma_channel* pIn, ma_uint32 channels); -/* -Determines whether or not a channel map is valid. -A blank channel map is valid (all channels set to MA_CHANNEL_NONE). The way a blank channel map is handled is context specific, but -is usually treated as a passthrough. +General Data Conversion +======================= +The `ma_data_converter` API can be used to wrap sample format conversion, channel conversion and resampling into one operation. This is what miniaudio uses +internally to convert between the format requested when the device was initialized and the format of the backend's native device. The API for general data +conversion is very similar to the resampling API. Create a `ma_data_converter` object like this: -Invalid channel maps: - - A channel map with no channels - - A channel map with more than one channel and a mono channel -*/ -ma_bool32 ma_channel_map_valid(ma_uint32 channels, const ma_channel channelMap[MA_MAX_CHANNELS]); + ```c + ma_data_converter_config config = ma_data_converter_config_init(inputFormat, outputFormat, inputChannels, outputChannels, inputSampleRate, outputSampleRate); + ma_data_converter converter; + ma_result result = ma_data_converter_init(&config, &converter); + if (result != MA_SUCCESS) { + // An error occurred... + } + ``` -/* -Helper for comparing two channel maps for equality. +In the example above we use `ma_data_converter_config_init()` to initialize the config, however there's many more properties that can be configured, such as +channel maps and resampling quality. Something like the following may be more suitable depending on your requirements: -This assumes the channel count is the same between the two. -*/ -ma_bool32 ma_channel_map_equal(ma_uint32 channels, const ma_channel channelMapA[MA_MAX_CHANNELS], const ma_channel channelMapB[MA_MAX_CHANNELS]); + ```c + ma_data_converter_config config = ma_data_converter_config_init_default(); + config.formatIn = inputFormat; + config.formatOut = outputFormat; + config.channelsIn = inputChannels; + config.channelsOut = outputChannels; + config.sampleRateIn = inputSampleRate; + config.sampleRateOut = outputSampleRate; + ma_get_standard_channel_map(ma_standard_channel_map_flac, config.channelCountIn, config.channelMapIn); + config.resampling.linear.lpfCount = MA_MAX_RESAMPLER_LPF_FILTERS; + ``` -/* -Helper for determining if a channel map is blank (all channels set to MA_CHANNEL_NONE). -*/ -ma_bool32 ma_channel_map_blank(ma_uint32 channels, const ma_channel channelMap[MA_MAX_CHANNELS]); +Do the following to uninitialize the data converter: -/* -Helper for determining whether or not a channel is present in the given channel map. -*/ -ma_bool32 ma_channel_map_contains_channel_position(ma_uint32 channels, const ma_channel channelMap[MA_MAX_CHANNELS], ma_channel channelPosition); + ```c + ma_data_converter_uninit(&converter); + ``` +The following example shows how data can be processed -/************************************************************************************************************************************************************ + ```c + ma_uint64 frameCountIn = 1000; + ma_uint64 frameCountOut = 2000; + ma_result result = ma_data_converter_process_pcm_frames(&converter, pFramesIn, &frameCountIn, pFramesOut, &frameCountOut); + if (result != MA_SUCCESS) { + // An error occurred... + } -Format Conversion -================= -The format converter serves two purposes: - 1) Conversion between data formats (u8 to f32, etc.) - 2) Interleaving and deinterleaving + // At this point, frameCountIn contains the number of input frames that were consumed and frameCountOut contains the number of output frames written. + ``` -When initializing a converter, you specify the input and output formats (u8, s16, etc.) and read callbacks. There are two read callbacks - one for -interleaved input data (onRead) and another for deinterleaved input data (onReadDeinterleaved). You implement whichever is most convenient for you. You -can implement both, but it's not recommended as it just introduces unnecessary complexity. +The data converter supports multiple channels and is always interleaved (both input and output). The channel count cannot be changed after initialization. -To read data as interleaved samples, use ma_format_converter_read(). Otherwise use ma_format_converter_read_deinterleaved(). +The sample rates can be anything other than zero, and are always specified in hertz. They should be set to something like 44100, etc. The sample rate is the +only configuration property that can be changed after initialization, but only if the `resampling.allowDynamicSampleRate` member of `ma_data_converter_config` +is set to MA_TRUE. To change the sample rate, use `ma_data_converter_set_rate()` or `ma_data_converter_set_rate_ratio()`. The ratio must be in/out. The +resampling algorithm cannot be changed after initialization. -Dithering ---------- -The format converter also supports dithering. Dithering can be set using ditherMode variable in the config, like so. +Processing always happens on a per PCM frame basis and always assumes interleaved input and output. De-interleaved processing is not supported. To process +frames, use `ma_data_converter_process_pcm_frames()`. On input, this function takes the number of output frames you can fit in the output buffer and the number +of input frames contained in the input buffer. On output these variables contain the number of output frames that were written to the output buffer and the +number of input frames that were consumed in the process. You can pass in NULL for the input buffer in which case it will be treated as an infinitely large +buffer of zeros. The output buffer can also be NULL, in which case the processing will be treated as seek. - pConfig->ditherMode = ma_dither_mode_rectangle; +Sometimes it's useful to know exactly how many input frames will be required to output a specific number of frames. You can calculate this with +`ma_data_converter_get_required_input_frame_count()`. Likewise, it's sometimes useful to know exactly how many frames would be output given a certain number of +input frames. You can do this with `ma_data_converter_get_expected_output_frame_count()`. -The different dithering modes include the following, in order of efficiency: - - None: ma_dither_mode_none - - Rectangle: ma_dither_mode_rectangle - - Triangle: ma_dither_mode_triangle +Due to the nature of how resampling works, the data converter introduces some latency if resampling is required. This can be retrieved in terms of both the +input rate and the output rate with `ma_data_converter_get_input_latency()` and `ma_data_converter_get_output_latency()`. -Note that even if the dither mode is set to something other than ma_dither_mode_none, it will be ignored for conversions where dithering is not needed. -Dithering is available for the following conversions: - - s16 -> u8 - - s24 -> u8 - - s32 -> u8 - - f32 -> u8 - - s24 -> s16 - - s32 -> s16 - - f32 -> s16 -Note that it is not an error to pass something other than ma_dither_mode_none for conversions where dither is not used. It will just be ignored. -************************************************************************************************************************************************************/ -/* -Initializes a format converter. -*/ -ma_result ma_format_converter_init(const ma_format_converter_config* pConfig, ma_format_converter* pConverter); +Biquad Filtering +================ +Biquad filtering is achieved with the `ma_biquad` API. Example: -/* -Reads data from the format converter as interleaved channels. -*/ -ma_uint64 ma_format_converter_read(ma_format_converter* pConverter, ma_uint64 frameCount, void* pFramesOut, void* pUserData); + ```c + ma_biquad_config config = ma_biquad_config_init(ma_format_f32, channels, b0, b1, b2, a0, a1, a2); + ma_result result = ma_biquad_init(&config, &biquad); + if (result != MA_SUCCESS) { + // Error. + } -/* -Reads data from the format converter as deinterleaved channels. -*/ -ma_uint64 ma_format_converter_read_deinterleaved(ma_format_converter* pConverter, ma_uint64 frameCount, void** ppSamplesOut, void* pUserData); + ... -/* -Helper for initializing a format converter config. -*/ -ma_format_converter_config ma_format_converter_config_init_new(void); -ma_format_converter_config ma_format_converter_config_init(ma_format formatIn, ma_format formatOut, ma_uint32 channels, ma_format_converter_read_proc onRead, void* pUserData); -ma_format_converter_config ma_format_converter_config_init_deinterleaved(ma_format formatIn, ma_format formatOut, ma_uint32 channels, ma_format_converter_read_deinterleaved_proc onReadDeinterleaved, void* pUserData); + ma_biquad_process_pcm_frames(&biquad, pFramesOut, pFramesIn, frameCount); + ``` +Biquad filtering is implemented using transposed direct form 2. The numerator coefficients are b0, b1 and b2, and the denominator coefficients are a0, a1 and +a2. The a0 coefficient is required and coefficients must not be pre-normalized. +Supported formats are `ma_format_s16` and `ma_format_f32`. If you need to use a different format you need to convert it yourself beforehand. When using +`ma_format_s16` the biquad filter will use fixed point arithmetic. When using `ma_format_f32`, floating point arithmetic will be used. -/************************************************************************************************************************************************************ +Input and output frames are always interleaved. -Channel Routing -=============== -There are two main things you can do with the channel router: - 1) Rearrange channels - 2) Convert from one channel count to another +Filtering can be applied in-place by passing in the same pointer for both the input and output buffers, like so: -Channel Rearrangement ---------------------- -A simple example of channel rearrangement may be swapping the left and right channels in a stereo stream. To do this you just pass in the same channel -count for both the input and output with channel maps that contain the same channels (in a different order). + ```c + ma_biquad_process_pcm_frames(&biquad, pMyData, pMyData, frameCount); + ``` -Channel Conversion ------------------- -The channel router can also convert from one channel count to another, such as converting a 5.1 stream to stero. When changing the channel count, the -router will first perform a 1:1 mapping of channel positions that are present in both the input and output channel maps. The second thing it will do -is distribute the input mono channel (if any) across all output channels, excluding any None and LFE channels. If there is an output mono channel, all -input channels will be averaged, excluding any None and LFE channels. - -The last case to consider is when a channel position in the input channel map is not present in the output channel map, and vice versa. In this case the -channel router will perform a blend of other related channels to produce an audible channel. There are several blending modes. - 1) Simple - Unmatched channels are silenced. - 2) Planar Blending - Channels are blended based on a set of planes that each speaker emits audio from. - -Rectangular / Planar Blending ------------------------------ -In this mode, channel positions are associated with a set of planes where the channel conceptually emits audio from. An example is the front/left speaker. -This speaker is positioned to the front of the listener, so you can think of it as emitting audio from the front plane. It is also positioned to the left -of the listener so you can think of it as also emitting audio from the left plane. Now consider the (unrealistic) situation where the input channel map -contains only the front/left channel position, but the output channel map contains both the front/left and front/center channel. When deciding on the audio -data to send to the front/center speaker (which has no 1:1 mapping with an input channel) we need to use some logic based on our available input channel -positions. - -As mentioned earlier, our front/left speaker is, conceptually speaking, emitting audio from the front _and_ the left planes. Similarly, the front/center -speaker is emitting audio from _only_ the front plane. What these two channels have in common is that they are both emitting audio from the front plane. -Thus, it makes sense that the front/center speaker should receive some contribution from the front/left channel. How much contribution depends on their -planar relationship (thus the name of this blending technique). - -Because the front/left channel is emitting audio from two planes (front and left), you can think of it as though it's willing to dedicate 50% of it's total -volume to each of it's planes (a channel position emitting from 1 plane would be willing to given 100% of it's total volume to that plane, and a channel -position emitting from 3 planes would be willing to given 33% of it's total volume to each plane). Similarly, the front/center speaker is emitting audio -from only one plane so you can think of it as though it's willing to _take_ 100% of it's volume from front plane emissions. Now, since the front/left -channel is willing to _give_ 50% of it's total volume to the front plane, and the front/center speaker is willing to _take_ 100% of it's total volume -from the front, you can imagine that 50% of the front/left speaker will be given to the front/center speaker. - -Usage ------ -To use the channel router you need to specify three things: - 1) The input channel count and channel map - 2) The output channel count and channel map - 3) The mixing mode to use in the case where a 1:1 mapping is unavailable +If you need to change the values of the coefficients, but maintain the values in the registers you can do so with `ma_biquad_reinit()`. This is useful if you +need to change the properties of the filter while keeping the values of registers valid to avoid glitching or whatnot. Do not use `ma_biquad_init()` for this +as it will do a full initialization which involves clearing the registers to 0. Note that changing the format or channel count after initialization is invalid +and will result in an error. -Note that input and output data is always deinterleaved 32-bit floating point. -Initialize the channel router with ma_channel_router_init(). You will need to pass in a config object which specifies the input and output configuration, -mixing mode and a callback for sending data to the router. This callback will be called when input data needs to be sent to the router for processing. Note -that the mixing mode is only used when a 1:1 mapping is unavailable. This includes the custom weights mode. -Read data from the channel router with ma_channel_router_read_deinterleaved(). Output data is always 32-bit floating point. +Low-Pass, High-Pass and Band-Pass Filtering +=========================================== +Low-pass, high-pass and band-pass filtering is achieved with the `ma_lpf`, `ma_hpf` and `ma_bpf` APIs respective. Low-pass filter example: -************************************************************************************************************************************************************/ + ```c + ma_lpf_config config = ma_lpf_config_init(ma_format_f32, channels, sampleRate, cutoffFrequency); + ma_result result = ma_lpf_init(&config, &lpf); + if (result != MA_SUCCESS) { + // Error. + } -/* -Initializes a channel router where it is assumed that the input data is non-interleaved. -*/ -ma_result ma_channel_router_init(const ma_channel_router_config* pConfig, ma_channel_router* pRouter); + ... -/* -Reads data from the channel router as deinterleaved channels. -*/ -ma_uint64 ma_channel_router_read_deinterleaved(ma_channel_router* pRouter, ma_uint64 frameCount, void** ppSamplesOut, void* pUserData); + ma_lpf_process_pcm_frames(&lpf, pFramesOut, pFramesIn, frameCount); + ``` -/* -Helper for initializing a channel router config. -*/ -ma_channel_router_config ma_channel_router_config_init(ma_uint32 channelsIn, const ma_channel channelMapIn[MA_MAX_CHANNELS], ma_uint32 channelsOut, const ma_channel channelMapOut[MA_MAX_CHANNELS], ma_channel_mix_mode mixingMode, ma_channel_router_read_deinterleaved_proc onRead, void* pUserData); +Supported formats are `ma_format_s16` and` ma_format_f32`. If you need to use a different format you need to convert it yourself beforehand. Input and output +frames are always interleaved. +Filtering can be applied in-place by passing in the same pointer for both the input and output buffers, like so: -/************************************************************************************************************************************************************ + ```c + ma_lpf_process_pcm_frames(&lpf, pMyData, pMyData, frameCount); + ``` -Sample Rate Conversion -====================== +These filters are implemented as a biquad filter. If you need to increase the filter order, simply chain multiple filters together. -************************************************************************************************************************************************************/ + ```c + for (iFilter = 0; iFilter < filterCount; iFilter += 1) { + ma_lpf_process_pcm_frames(&lpf[iFilter], pMyData, pMyData, frameCount); + } + ``` -/* -Initializes a sample rate conversion object. -*/ -ma_result ma_src_init(const ma_src_config* pConfig, ma_src* pSRC); +If you need to change the configuration of the filter, but need to maintain the state of internal registers you can do so with `ma_lpf_reinit()`. This may be +useful if you need to change the sample rate and/or cutoff frequency dynamically while maintaing smooth transitions. Note that changing the format or channel +count after initialization is invalid and will result in an error. -/* -Dynamically adjusts the sample rate. +The example code above is for low-pass filters, but the same applies for high-pass and band-pass filters, only you should use the `ma_hpf` and `ma_bpf` APIs +instead. -This is useful for dynamically adjust pitch. Keep in mind, however, that this will speed up or slow down the sound. If this -is not acceptable you will need to use your own algorithm. -*/ -ma_result ma_src_set_sample_rate(ma_src* pSRC, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut); -/* -Reads a number of frames. -Returns the number of frames actually read. -*/ -ma_uint64 ma_src_read_deinterleaved(ma_src* pSRC, ma_uint64 frameCount, void** ppSamplesOut, void* pUserData); +Waveforms +========= +miniaudio supports generation of sine, square, triangle and sawtooth waveforms. This is achieved with the `ma_waveform` API. Example: -/* -Helper for creating a sample rate conversion config. -*/ -ma_src_config ma_src_config_init_new(void); -ma_src_config ma_src_config_init(ma_uint32 sampleRateIn, ma_uint32 sampleRateOut, ma_uint32 channels, ma_src_read_deinterleaved_proc onReadDeinterleaved, void* pUserData); + ```c + ma_waveform waveform; + ma_result result = ma_waveform_init(ma_waveform_type_sine, amplitude, frequency, sampleRate, &waveform); + if (result != MA_SUCCESS) { + // Error. + } + ... -/************************************************************************************************************************************************************ + ma_waveform_read_pcm_frames(&waveform, pOutput, frameCount, FORMAT, CHANNELS); + ``` -Conversion +The amplitude, frequency and sample rate can be changed dynamically with `ma_waveform_set_amplitude()`, `ma_waveform_set_frequency()` and +`ma_waveform_set_sample_rate()` respectively. -************************************************************************************************************************************************************/ -/* -Initializes a DSP object. -*/ -ma_result ma_pcm_converter_init(const ma_pcm_converter_config* pConfig, ma_pcm_converter* pDSP); -/* -Dynamically adjusts the input sample rate. +Ring Buffers +============ +miniaudio supports lock free (single producer, single consumer) ring buffers which are exposed via the `ma_rb` and `ma_pcm_rb` APIs. The `ma_rb` API operates +on bytes, whereas the `ma_pcm_rb` operates on PCM frames. They are otherwise identical as `ma_pcm_rb` is just a wrapper around `ma_rb`. -This will fail is the DSP was not initialized with allowDynamicSampleRate. +Unlike most other APIs in miniaudio, ring buffers support both interleaved and deinterleaved streams. The caller can also allocate their own backing memory for +the ring buffer to use internally for added flexibility. Otherwise the ring buffer will manage it's internal memory for you. -DEPRECATED. Use ma_pcm_converter_set_sample_rate() instead. -*/ -ma_result ma_pcm_converter_set_input_sample_rate(ma_pcm_converter* pDSP, ma_uint32 sampleRateOut); +The examples below use the PCM frame variant of the ring buffer since that's most likely the one you will want to use. To initialize a ring buffer, do +something like the following: -/* -Dynamically adjusts the output sample rate. + ```c + ma_pcm_rb rb; + ma_result result = ma_pcm_rb_init(FORMAT, CHANNELS, BUFFER_SIZE_IN_FRAMES, NULL, NULL, &rb); + if (result != MA_SUCCESS) { + // Error + } + ``` -This is useful for dynamically adjust pitch. Keep in mind, however, that this will speed up or slow down the sound. If this -is not acceptable you will need to use your own algorithm. +The `ma_pcm_rb_init()` function takes the sample format and channel count as parameters because it's the PCM varient of the ring buffer API. For the regular +ring buffer that operates on bytes you would call `ma_rb_init()` which leaves these out and just takes the size of the buffer in bytes instead of frames. The +fourth parameter is an optional pre-allocated buffer and the fifth parameter is a pointer to a `ma_allocation_callbacks` structure for custom memory allocation +routines. Passing in NULL for this results in MA_MALLOC() and MA_FREE() being used. -This will fail is the DSP was not initialized with allowDynamicSampleRate. +Use `ma_pcm_rb_init_ex()` if you need a deinterleaved buffer. The data for each sub-buffer is offset from each other based on the stride. To manage your sub- +buffers you can use `ma_pcm_rb_get_subbuffer_stride()`, `ma_pcm_rb_get_subbuffer_offset()` and `ma_pcm_rb_get_subbuffer_ptr()`. -DEPRECATED. Use ma_pcm_converter_set_sample_rate() instead. -*/ -ma_result ma_pcm_converter_set_output_sample_rate(ma_pcm_converter* pDSP, ma_uint32 sampleRateOut); +Use 'ma_pcm_rb_acquire_read()` and `ma_pcm_rb_acquire_write()` to retrieve a pointer to a section of the ring buffer. You specify the number of frames you +need, and on output it will set to what was actually acquired. If the read or write pointer is positioned such that the number of frames requested will require +a loop, it will be clamped to the end of the buffer. Therefore, the number of frames you're given may be less than the number you requested. -/* -Dynamically adjusts the output sample rate. +After calling `ma_pcm_rb_acquire_read()` or `ma_pcm_rb_acquire_write()`, you do your work on the buffer and then "commit" it with `ma_pcm_rb_commit_read()` or +`ma_pcm_rb_commit_write()`. This is where the read/write pointers are updated. When you commit you need to pass in the buffer that was returned by the earlier +call to `ma_pcm_rb_acquire_read()` or `ma_pcm_rb_acquire_write()` and is only used for validation. The number of frames passed to `ma_pcm_rb_commit_read()` and +`ma_pcm_rb_commit_write()` is what's used to increment the pointers. -This is useful for dynamically adjust pitch. Keep in mind, however, that this will speed up or slow down the sound. If this -is not acceptable you will need to use your own algorithm. +If you want to correct for drift between the write pointer and the read pointer you can use a combination of `ma_pcm_rb_pointer_distance()`, +`ma_pcm_rb_seek_read()` and `ma_pcm_rb_seek_write()`. Note that you can only move the pointers forward, and you should only move the read pointer forward via +the consumer thread, and the write pointer forward by the producer thread. If there is too much space between the pointers, move the read pointer forward. If +there is too little space between the pointers, move the write pointer forward. -This will fail if the DSP was not initialized with allowDynamicSampleRate. -*/ -ma_result ma_pcm_converter_set_sample_rate(ma_pcm_converter* pDSP, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut); +You can use a ring buffer at the byte level instead of the PCM frame level by using the `ma_rb` API. This is exactly the sample, only you will use the `ma_rb` +functions instead of `ma_pcm_rb` and instead of frame counts you'll pass around byte counts. -/* -Reads a number of frames and runs them through the DSP processor. -*/ -ma_uint64 ma_pcm_converter_read(ma_pcm_converter* pDSP, void* pFramesOut, ma_uint64 frameCount); +The maximum size of the buffer in bytes is 0x7FFFFFFF-(MA_SIMD_ALIGNMENT-1) due to the most significant bit being used to encode a flag and the internally +managed buffers always being aligned to MA_SIMD_ALIGNMENT. -/* -Helper for initializing a ma_pcm_converter_config object. -*/ -ma_pcm_converter_config ma_pcm_converter_config_init_new(void); -ma_pcm_converter_config ma_pcm_converter_config_init(ma_format formatIn, ma_uint32 channelsIn, ma_uint32 sampleRateIn, ma_format formatOut, ma_uint32 channelsOut, ma_uint32 sampleRateOut, ma_pcm_converter_read_proc onRead, void* pUserData); -ma_pcm_converter_config ma_pcm_converter_config_init_ex(ma_format formatIn, ma_uint32 channelsIn, ma_uint32 sampleRateIn, ma_channel channelMapIn[MA_MAX_CHANNELS], ma_format formatOut, ma_uint32 channelsOut, ma_uint32 sampleRateOut, ma_channel channelMapOut[MA_MAX_CHANNELS], ma_pcm_converter_read_proc onRead, void* pUserData); +Note that the ring buffer is only thread safe when used by a single consumer thread and single producer thread. -/* -High-level helper for doing a full format conversion in one go. Returns the number of output frames. Call this with pOut set to NULL to -determine the required size of the output buffer. -A return value of 0 indicates an error. -This function is useful for one-off bulk conversions, but if you're streaming data you should use the ma_pcm_converter APIs instead. -*/ -ma_uint64 ma_convert_frames(void* pOut, ma_format formatOut, ma_uint32 channelsOut, ma_uint32 sampleRateOut, const void* pIn, ma_format formatIn, ma_uint32 channelsIn, ma_uint32 sampleRateIn, ma_uint64 frameCount); -ma_uint64 ma_convert_frames_ex(void* pOut, ma_format formatOut, ma_uint32 channelsOut, ma_uint32 sampleRateOut, ma_channel channelMapOut[MA_MAX_CHANNELS], const void* pIn, ma_format formatIn, ma_uint32 channelsIn, ma_uint32 sampleRateIn, ma_channel channelMapIn[MA_MAX_CHANNELS], ma_uint64 frameCount); +Backends +======== +The following backends are supported by miniaudio. + + |-------------|-----------------------|--------------------------------------------------------| + | Name | Enum Name | Supported Operating Systems | + |-------------|-----------------------|--------------------------------------------------------| + | WASAPI | ma_backend_wasapi | Windows Vista+ | + | DirectSound | ma_backend_dsound | Windows XP+ | + | WinMM | ma_backend_winmm | Windows XP+ (may work on older versions, but untested) | + | Core Audio | ma_backend_coreaudio | macOS, iOS | + | ALSA | ma_backend_alsa | Linux | + | PulseAudio | ma_backend_pulseaudio | Cross Platform (disabled on Windows, BSD and Android) | + | JACK | ma_backend_jack | Cross Platform (disabled on BSD and Android) | + | sndio | ma_backend_sndio | OpenBSD | + | audio(4) | ma_backend_audio4 | NetBSD, OpenBSD | + | OSS | ma_backend_oss | FreeBSD | + | AAudio | ma_backend_aaudio | Android 8+ | + | OpenSL|ES | ma_backend_opensl | Android (API level 16+) | + | Web Audio | ma_backend_webaudio | Web (via Emscripten) | + | Null | ma_backend_null | Cross Platform (not used on Web) | + |-------------|-----------------------|--------------------------------------------------------| + +Some backends have some nuance details you may want to be aware of. +WASAPI +------ +- Low-latency shared mode will be disabled when using an application-defined sample rate which is different to the device's native sample rate. To work around + this, set wasapi.noAutoConvertSRC to true in the device config. This is due to IAudioClient3_InitializeSharedAudioStream() failing when the + AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM flag is specified. Setting wasapi.noAutoConvertSRC will result in miniaudio's lower quality internal resampler being used + instead which will in turn enable the use of low-latency shared mode. -/************************************************************************************************************************************************************ +PulseAudio +---------- +- If you experience bad glitching/noise on Arch Linux, consider this fix from the Arch wiki: + https://wiki.archlinux.org/index.php/PulseAudio/Troubleshooting#Glitches,_skips_or_crackling + Alternatively, consider using a different backend such as ALSA. -Ring Buffer -=========== +Android +------- +- To capture audio on Android, remember to add the RECORD_AUDIO permission to your manifest: + +- With OpenSL|ES, only a single ma_context can be active at any given time. This is due to a limitation with OpenSL|ES. +- With AAudio, only default devices are enumerated. This is due to AAudio not having an enumeration API (devices are enumerated through Java). You can however + perform your own device enumeration through Java and then set the ID in the ma_device_id structure (ma_device_id.aaudio) and pass it to ma_device_init(). +- The backend API will perform resampling where possible. The reason for this as opposed to using miniaudio's built-in resampler is to take advantage of any + potential device-specific optimizations the driver may implement. -Features --------- -- Lock free (assuming single producer, single consumer) -- Support for interleaved and deinterleaved streams -- Allows the caller to allocate their own block of memory +UWP +--- +- UWP only supports default playback and capture devices. +- UWP requires the Microphone capability to be enabled in the application's manifest (Package.appxmanifest): + + ... + + + + -Usage ------ -- Call ma_rb_init() to initialize a simple buffer, with an optional pre-allocated buffer. If you pass in NULL - for the pre-allocated buffer, it will be allocated for you and free()'d in ma_rb_uninit(). If you pass in - your own pre-allocated buffer, free()-ing is left to you. - -- Call ma_rb_init_ex() if you need a deinterleaved buffer. The data for each sub-buffer is offset from each - other based on the stride. Use ma_rb_get_subbuffer_stride(), ma_rb_get_subbuffer_offset() and - ma_rb_get_subbuffer_ptr() to manage your sub-buffers. - -- Use ma_rb_acquire_read() and ma_rb_acquire_write() to retrieve a pointer to a section of the ring buffer. - You specify the number of bytes you need, and on output it will set to what was actually acquired. If the - read or write pointer is positioned such that the number of bytes requested will require a loop, it will be - clamped to the end of the buffer. Therefore, the number of bytes you're given may be less than the number - you requested. - -- After calling ma_rb_acquire_read/write(), you do your work on the buffer and then "commit" it with - ma_rb_commit_read/write(). This is where the read/write pointers are updated. When you commit you need to - pass in the buffer that was returned by the earlier call to ma_rb_acquire_read/write() and is only used - for validation. The number of bytes passed to ma_rb_commit_read/write() is what's used to increment the - pointers. - -- If you want to correct for drift between the write pointer and the read pointer you can use a combination - of ma_rb_pointer_distance(), ma_rb_seek_read() and ma_rb_seek_write(). Note that you can only move the - pointers forward, and you should only move the read pointer forward via the consumer thread, and the write - pointer forward by the producer thread. If there is too much space between the pointers, move the read - pointer forward. If there is too little space between the pointers, move the write pointer forward. - - -Notes ------ -- Thread safety depends on a single producer, single consumer model. Only one thread is allowed to write, and - only one thread is allowed to read. The producer is the only one allowed to move the write pointer, and the - consumer is the only one allowed to move the read pointer. -- Operates on bytes. Use ma_pcm_rb to operate in terms of PCM frames. -- Maximum buffer size in bytes is 0x7FFFFFFF-(MA_SIMD_ALIGNMENT-1) because of reasons. +Web Audio / Emscripten +---------------------- +- You cannot use -std=c* compiler flags, nor -ansi. This only applies to the Emscripten build. +- The first time a context is initialized it will create a global object called "miniaudio" whose primary purpose is to act as a factory for device objects. +- Currently the Web Audio backend uses ScriptProcessorNode's, but this may need to change later as they've been deprecated. +- Google has implemented a policy in their browsers that prevent automatic media output without first receiving some kind of user input. The following web page + has additional details: https://developers.google.com/web/updates/2017/09/autoplay-policy-changes. Starting the device may fail if you try to start playback + without first handling some kind of user input. -PCM Ring Buffer -=============== -This is the same as the regular ring buffer, except that it works on PCM frames instead of bytes. -************************************************************************************************************************************************************/ -typedef struct -{ - void* pBuffer; - ma_uint32 subbufferSizeInBytes; - ma_uint32 subbufferCount; - ma_uint32 subbufferStrideInBytes; - volatile ma_uint32 encodedReadOffset; /* Most significant bit is the loop flag. Lower 31 bits contains the actual offset in bytes. */ - volatile ma_uint32 encodedWriteOffset; /* Most significant bit is the loop flag. Lower 31 bits contains the actual offset in bytes. */ - ma_bool32 ownsBuffer : 1; /* Used to know whether or not miniaudio is responsible for free()-ing the buffer. */ - ma_bool32 clearOnWriteAcquire : 1; /* When set, clears the acquired write buffer before returning from ma_rb_acquire_write(). */ -} ma_rb; +Miscellaneous Notes +=================== +- Automatic stream routing is enabled on a per-backend basis. Support is explicitly enabled for WASAPI and Core Audio, however other backends such as + PulseAudio may naturally support it, though not all have been tested. +- The contents of the output buffer passed into the data callback will always be pre-initialized to zero unless the noPreZeroedOutputBuffer config variable in + ma_device_config is set to true, in which case it'll be undefined which will require you to write something to the entire buffer. +- By default miniaudio will automatically clip samples. This only applies when the playback sample format is configured as ma_format_f32. If you are doing + clipping yourself, you can disable this overhead by setting noClip to true in the device config. +- The sndio backend is currently only enabled on OpenBSD builds. +- The audio(4) backend is supported on OpenBSD, but you may need to disable sndiod before you can use it. +- Note that GCC and Clang requires "-msse2", "-mavx2", etc. for SIMD optimizations. +*/ -ma_result ma_rb_init_ex(size_t subbufferSizeInBytes, size_t subbufferCount, size_t subbufferStrideInBytes, void* pOptionalPreallocatedBuffer, ma_rb* pRB); -ma_result ma_rb_init(size_t bufferSizeInBytes, void* pOptionalPreallocatedBuffer, ma_rb* pRB); -void ma_rb_uninit(ma_rb* pRB); -void ma_rb_reset(ma_rb* pRB); -ma_result ma_rb_acquire_read(ma_rb* pRB, size_t* pSizeInBytes, void** ppBufferOut); -ma_result ma_rb_commit_read(ma_rb* pRB, size_t sizeInBytes, void* pBufferOut); -ma_result ma_rb_acquire_write(ma_rb* pRB, size_t* pSizeInBytes, void** ppBufferOut); -ma_result ma_rb_commit_write(ma_rb* pRB, size_t sizeInBytes, void* pBufferOut); -ma_result ma_rb_seek_read(ma_rb* pRB, size_t offsetInBytes); -ma_result ma_rb_seek_write(ma_rb* pRB, size_t offsetInBytes); -ma_int32 ma_rb_pointer_distance(ma_rb* pRB); /* Returns the distance between the write pointer and the read pointer. Should never be negative for a correct program. Will return the number of bytes that can be read before the read pointer hits the write pointer. */ -ma_uint32 ma_rb_available_read(ma_rb* pRB); -ma_uint32 ma_rb_available_write(ma_rb* pRB); -size_t ma_rb_get_subbuffer_size(ma_rb* pRB); -size_t ma_rb_get_subbuffer_stride(ma_rb* pRB); -size_t ma_rb_get_subbuffer_offset(ma_rb* pRB, size_t subbufferIndex); -void* ma_rb_get_subbuffer_ptr(ma_rb* pRB, size_t subbufferIndex, void* pBuffer); +#ifndef miniaudio_h +#define miniaudio_h +#ifdef __cplusplus +extern "C" { +#endif -typedef struct -{ - ma_rb rb; - ma_format format; - ma_uint32 channels; -} ma_pcm_rb; - -ma_result ma_pcm_rb_init_ex(ma_format format, ma_uint32 channels, ma_uint32 subbufferSizeInFrames, ma_uint32 subbufferCount, ma_uint32 subbufferStrideInFrames, void* pOptionalPreallocatedBuffer, ma_pcm_rb* pRB); -ma_result ma_pcm_rb_init(ma_format format, ma_uint32 channels, ma_uint32 bufferSizeInFrames, void* pOptionalPreallocatedBuffer, ma_pcm_rb* pRB); -void ma_pcm_rb_uninit(ma_pcm_rb* pRB); -void ma_pcm_rb_reset(ma_pcm_rb* pRB); -ma_result ma_pcm_rb_acquire_read(ma_pcm_rb* pRB, ma_uint32* pSizeInFrames, void** ppBufferOut); -ma_result ma_pcm_rb_commit_read(ma_pcm_rb* pRB, ma_uint32 sizeInFrames, void* pBufferOut); -ma_result ma_pcm_rb_acquire_write(ma_pcm_rb* pRB, ma_uint32* pSizeInFrames, void** ppBufferOut); -ma_result ma_pcm_rb_commit_write(ma_pcm_rb* pRB, ma_uint32 sizeInFrames, void* pBufferOut); -ma_result ma_pcm_rb_seek_read(ma_pcm_rb* pRB, ma_uint32 offsetInFrames); -ma_result ma_pcm_rb_seek_write(ma_pcm_rb* pRB, ma_uint32 offsetInFrames); -ma_int32 ma_pcm_rb_pointer_disance(ma_pcm_rb* pRB); /* Return value is in frames. */ -ma_uint32 ma_pcm_rb_available_read(ma_pcm_rb* pRB); -ma_uint32 ma_pcm_rb_available_write(ma_pcm_rb* pRB); -ma_uint32 ma_pcm_rb_get_subbuffer_size(ma_pcm_rb* pRB); -ma_uint32 ma_pcm_rb_get_subbuffer_stride(ma_pcm_rb* pRB); -ma_uint32 ma_pcm_rb_get_subbuffer_offset(ma_pcm_rb* pRB, ma_uint32 subbufferIndex); -void* ma_pcm_rb_get_subbuffer_ptr(ma_pcm_rb* pRB, ma_uint32 subbufferIndex, void* pBuffer); - - -/************************************************************************************************************************************************************ - -Miscellaneous Helpers - -************************************************************************************************************************************************************/ - -/* -malloc(). Calls MA_MALLOC(). -*/ -void* ma_malloc(size_t sz); - -/* -realloc(). Calls MA_REALLOC(). -*/ -void* ma_realloc(void* p, size_t sz); - -/* -free(). Calls MA_FREE(). -*/ -void ma_free(void* p); +#if defined(_MSC_VER) && !defined(__clang__) + #pragma warning(push) + #pragma warning(disable:4201) /* nonstandard extension used: nameless struct/union */ + #pragma warning(disable:4324) /* structure was padded due to alignment specifier */ +#else + #pragma GCC diagnostic push + #pragma GCC diagnostic ignored "-Wpedantic" /* For ISO C99 doesn't support unnamed structs/unions [-Wpedantic] */ + #if defined(__clang__) + #pragma GCC diagnostic ignored "-Wc11-extensions" /* anonymous unions are a C11 extension */ + #endif +#endif -/* -Performs an aligned malloc, with the assumption that the alignment is a power of 2. -*/ -void* ma_aligned_malloc(size_t sz, size_t alignment); +/* Platform/backend detection. */ +#ifdef _WIN32 + #define MA_WIN32 + #if defined(WINAPI_FAMILY) && (WINAPI_FAMILY == WINAPI_FAMILY_PC_APP || WINAPI_FAMILY == WINAPI_FAMILY_PHONE_APP) + #define MA_WIN32_UWP + #else + #define MA_WIN32_DESKTOP + #endif +#else + #define MA_POSIX + #include /* Unfortunate #include, but needed for pthread_t, pthread_mutex_t and pthread_cond_t types. */ + #include -/* -Free's an aligned malloc'd buffer. -*/ -void ma_aligned_free(void* p); + #ifdef __unix__ + #define MA_UNIX + #if defined(__DragonFly__) || defined(__FreeBSD__) || defined(__NetBSD__) || defined(__OpenBSD__) + #define MA_BSD + #endif + #endif + #ifdef __linux__ + #define MA_LINUX + #endif + #ifdef __APPLE__ + #define MA_APPLE + #endif + #ifdef __ANDROID__ + #define MA_ANDROID + #endif + #ifdef __EMSCRIPTEN__ + #define MA_EMSCRIPTEN + #endif +#endif -/* -Retrieves a friendly name for a format. -*/ -const char* ma_get_format_name(ma_format format); +#include /* For size_t. */ -/* -Blends two frames in floating point format. -*/ -void ma_blend_f32(float* pOut, float* pInA, float* pInB, float factor, ma_uint32 channels); +/* Sized types. Prefer built-in types. Fall back to stdint. */ +#ifdef _MSC_VER + #if defined(__clang__) + #pragma GCC diagnostic push + #pragma GCC diagnostic ignored "-Wlanguage-extension-token" + #pragma GCC diagnostic ignored "-Wlong-long" + #pragma GCC diagnostic ignored "-Wc++11-long-long" + #endif + typedef signed __int8 ma_int8; + typedef unsigned __int8 ma_uint8; + typedef signed __int16 ma_int16; + typedef unsigned __int16 ma_uint16; + typedef signed __int32 ma_int32; + typedef unsigned __int32 ma_uint32; + typedef signed __int64 ma_int64; + typedef unsigned __int64 ma_uint64; + #if defined(__clang__) + #pragma GCC diagnostic pop + #endif +#else + #define MA_HAS_STDINT + #include + typedef int8_t ma_int8; + typedef uint8_t ma_uint8; + typedef int16_t ma_int16; + typedef uint16_t ma_uint16; + typedef int32_t ma_int32; + typedef uint32_t ma_uint32; + typedef int64_t ma_int64; + typedef uint64_t ma_uint64; +#endif -/* -Retrieves the size of a sample in bytes for the given format. +#ifdef MA_HAS_STDINT + typedef uintptr_t ma_uintptr; +#else + #if defined(_WIN32) + #if defined(_WIN64) + typedef ma_uint64 ma_uintptr; + #else + typedef ma_uint32 ma_uintptr; + #endif + #elif defined(__GNUC__) + #if defined(__LP64__) + typedef ma_uint64 ma_uintptr; + #else + typedef ma_uint32 ma_uintptr; + #endif + #else + typedef ma_uint64 ma_uintptr; /* Fallback. */ + #endif +#endif -This API is efficient and is implemented using a lookup table. +typedef ma_uint8 ma_bool8; +typedef ma_uint32 ma_bool32; +#define MA_TRUE 1 +#define MA_FALSE 0 -Thread Safety: SAFE - This API is pure. -*/ -ma_uint32 ma_get_bytes_per_sample(ma_format format); -static MA_INLINE ma_uint32 ma_get_bytes_per_frame(ma_format format, ma_uint32 channels) { return ma_get_bytes_per_sample(format) * channels; } +typedef void* ma_handle; +typedef void* ma_ptr; +typedef void (* ma_proc)(void); -/* -Converts a log level to a string. -*/ -const char* ma_log_level_to_string(ma_uint32 logLevel); +#if defined(_MSC_VER) && !defined(_WCHAR_T_DEFINED) +typedef ma_uint16 wchar_t; +#endif +/* Define NULL for some compilers. */ +#ifndef NULL +#define NULL 0 +#endif -/************************************************************************************************************************************************************ +#if defined(SIZE_MAX) + #define MA_SIZE_MAX SIZE_MAX +#else + #define MA_SIZE_MAX 0xFFFFFFFF /* When SIZE_MAX is not defined by the standard library just default to the maximum 32-bit unsigned integer. */ +#endif -Format Conversion -************************************************************************************************************************************************************/ -void ma_pcm_u8_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_u8_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_u8_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_u8_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_s16_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_s16_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_s16_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_s16_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_s24_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_s24_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_s24_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_s24_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_s32_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_s32_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_s32_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_s32_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_f32_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_f32_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_f32_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_f32_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); -void ma_pcm_convert(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 sampleCount, ma_dither_mode ditherMode); +#ifdef _MSC_VER + #define MA_INLINE __forceinline +#elif defined(__GNUC__) + /* + I've had a bug report where GCC is emitting warnings about functions possibly not being inlineable. This warning happens when + the __attribute__((always_inline)) attribute is defined without an "inline" statement. I think therefore there must be some + case where "__inline__" is not always defined, thus the compiler emitting these warnings. When using -std=c89 or -ansi on the + command line, we cannot use the "inline" keyword and instead need to use "__inline__". In an attempt to work around this issue + I am using "__inline__" only when we're compiling in strict ANSI mode. + */ + #if defined(__STRICT_ANSI__) + #define MA_INLINE __inline__ __attribute__((always_inline)) + #else + #define MA_INLINE inline __attribute__((always_inline)) + #endif +#else + #define MA_INLINE +#endif -/* -Deinterleaves an interleaved buffer. -*/ -void ma_deinterleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void* pInterleavedPCMFrames, void** ppDeinterleavedPCMFrames); +#if defined(_MSC_VER) + #if _MSC_VER >= 1400 + #define MA_ALIGN(alignment) __declspec(align(alignment)) + #endif +#elif !defined(__DMC__) + #define MA_ALIGN(alignment) __attribute__((aligned(alignment))) +#endif +#ifndef MA_ALIGN + #define MA_ALIGN(alignment) +#endif -/* -Interleaves a group of deinterleaved buffers. -*/ -void ma_interleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void** ppDeinterleavedPCMFrames, void* pInterleavedPCMFrames); +/* SIMD alignment in bytes. Currently set to 64 bytes in preparation for future AVX-512 optimizations. */ +#define MA_SIMD_ALIGNMENT 64 -/************************************************************************************************************************************************************ -************************************************************************************************************************************************************* +/* Logging levels */ +#define MA_LOG_LEVEL_VERBOSE 4 +#define MA_LOG_LEVEL_INFO 3 +#define MA_LOG_LEVEL_WARNING 2 +#define MA_LOG_LEVEL_ERROR 1 -DEVICE I/O -========== +#ifndef MA_LOG_LEVEL +#define MA_LOG_LEVEL MA_LOG_LEVEL_ERROR +#endif -This section contains the APIs for device playback and capture. Here is where you'll find ma_device_init(), etc. +typedef struct ma_context ma_context; +typedef struct ma_device ma_device; -************************************************************************************************************************************************************* -************************************************************************************************************************************************************/ -#ifndef MA_NO_DEVICE_IO -/* Some backends are only supported on certain platforms. */ -#if defined(MA_WIN32) - #define MA_SUPPORT_WASAPI - #if defined(MA_WIN32_DESKTOP) /* DirectSound and WinMM backends are only supported on desktops. */ - #define MA_SUPPORT_DSOUND - #define MA_SUPPORT_WINMM - #define MA_SUPPORT_JACK /* JACK is technically supported on Windows, but I don't know how many people use it in practice... */ - #endif -#endif -#if defined(MA_UNIX) - #if defined(MA_LINUX) - #if !defined(MA_ANDROID) /* ALSA is not supported on Android. */ - #define MA_SUPPORT_ALSA - #endif - #endif - #if !defined(MA_BSD) && !defined(MA_ANDROID) && !defined(MA_EMSCRIPTEN) - #define MA_SUPPORT_PULSEAUDIO - #define MA_SUPPORT_JACK - #endif - #if defined(MA_ANDROID) - #define MA_SUPPORT_AAUDIO - #define MA_SUPPORT_OPENSL - #endif - #if defined(__OpenBSD__) /* <-- Change this to "#if defined(MA_BSD)" to enable sndio on all BSD flavors. */ - #define MA_SUPPORT_SNDIO /* sndio is only supported on OpenBSD for now. May be expanded later if there's demand. */ - #endif - #if defined(__NetBSD__) || defined(__OpenBSD__) - #define MA_SUPPORT_AUDIO4 /* Only support audio(4) on platforms with known support. */ - #endif - #if defined(__FreeBSD__) || defined(__DragonFly__) - #define MA_SUPPORT_OSS /* Only support OSS on specific platforms with known support. */ - #endif -#endif -#if defined(MA_APPLE) - #define MA_SUPPORT_COREAUDIO -#endif -#if defined(MA_EMSCRIPTEN) - #define MA_SUPPORT_WEBAUDIO -#endif +typedef ma_uint8 ma_channel; +#define MA_CHANNEL_NONE 0 +#define MA_CHANNEL_MONO 1 +#define MA_CHANNEL_FRONT_LEFT 2 +#define MA_CHANNEL_FRONT_RIGHT 3 +#define MA_CHANNEL_FRONT_CENTER 4 +#define MA_CHANNEL_LFE 5 +#define MA_CHANNEL_BACK_LEFT 6 +#define MA_CHANNEL_BACK_RIGHT 7 +#define MA_CHANNEL_FRONT_LEFT_CENTER 8 +#define MA_CHANNEL_FRONT_RIGHT_CENTER 9 +#define MA_CHANNEL_BACK_CENTER 10 +#define MA_CHANNEL_SIDE_LEFT 11 +#define MA_CHANNEL_SIDE_RIGHT 12 +#define MA_CHANNEL_TOP_CENTER 13 +#define MA_CHANNEL_TOP_FRONT_LEFT 14 +#define MA_CHANNEL_TOP_FRONT_CENTER 15 +#define MA_CHANNEL_TOP_FRONT_RIGHT 16 +#define MA_CHANNEL_TOP_BACK_LEFT 17 +#define MA_CHANNEL_TOP_BACK_CENTER 18 +#define MA_CHANNEL_TOP_BACK_RIGHT 19 +#define MA_CHANNEL_AUX_0 20 +#define MA_CHANNEL_AUX_1 21 +#define MA_CHANNEL_AUX_2 22 +#define MA_CHANNEL_AUX_3 23 +#define MA_CHANNEL_AUX_4 24 +#define MA_CHANNEL_AUX_5 25 +#define MA_CHANNEL_AUX_6 26 +#define MA_CHANNEL_AUX_7 27 +#define MA_CHANNEL_AUX_8 28 +#define MA_CHANNEL_AUX_9 29 +#define MA_CHANNEL_AUX_10 30 +#define MA_CHANNEL_AUX_11 31 +#define MA_CHANNEL_AUX_12 32 +#define MA_CHANNEL_AUX_13 33 +#define MA_CHANNEL_AUX_14 34 +#define MA_CHANNEL_AUX_15 35 +#define MA_CHANNEL_AUX_16 36 +#define MA_CHANNEL_AUX_17 37 +#define MA_CHANNEL_AUX_18 38 +#define MA_CHANNEL_AUX_19 39 +#define MA_CHANNEL_AUX_20 40 +#define MA_CHANNEL_AUX_21 41 +#define MA_CHANNEL_AUX_22 42 +#define MA_CHANNEL_AUX_23 43 +#define MA_CHANNEL_AUX_24 44 +#define MA_CHANNEL_AUX_25 45 +#define MA_CHANNEL_AUX_26 46 +#define MA_CHANNEL_AUX_27 47 +#define MA_CHANNEL_AUX_28 48 +#define MA_CHANNEL_AUX_29 49 +#define MA_CHANNEL_AUX_30 50 +#define MA_CHANNEL_AUX_31 51 +#define MA_CHANNEL_LEFT MA_CHANNEL_FRONT_LEFT +#define MA_CHANNEL_RIGHT MA_CHANNEL_FRONT_RIGHT +#define MA_CHANNEL_POSITION_COUNT (MA_CHANNEL_AUX_31 + 1) -/* Explicitly disable the Null backend for Emscripten because it uses a background thread which is not properly supported right now. */ -#if !defined(MA_EMSCRIPTEN) -#define MA_SUPPORT_NULL -#endif +typedef int ma_result; +#define MA_SUCCESS 0 -#if !defined(MA_NO_WASAPI) && defined(MA_SUPPORT_WASAPI) - #define MA_ENABLE_WASAPI -#endif -#if !defined(MA_NO_DSOUND) && defined(MA_SUPPORT_DSOUND) - #define MA_ENABLE_DSOUND -#endif -#if !defined(MA_NO_WINMM) && defined(MA_SUPPORT_WINMM) - #define MA_ENABLE_WINMM -#endif -#if !defined(MA_NO_ALSA) && defined(MA_SUPPORT_ALSA) - #define MA_ENABLE_ALSA -#endif -#if !defined(MA_NO_PULSEAUDIO) && defined(MA_SUPPORT_PULSEAUDIO) - #define MA_ENABLE_PULSEAUDIO -#endif -#if !defined(MA_NO_JACK) && defined(MA_SUPPORT_JACK) - #define MA_ENABLE_JACK -#endif -#if !defined(MA_NO_COREAUDIO) && defined(MA_SUPPORT_COREAUDIO) - #define MA_ENABLE_COREAUDIO -#endif -#if !defined(MA_NO_SNDIO) && defined(MA_SUPPORT_SNDIO) - #define MA_ENABLE_SNDIO -#endif -#if !defined(MA_NO_AUDIO4) && defined(MA_SUPPORT_AUDIO4) - #define MA_ENABLE_AUDIO4 -#endif -#if !defined(MA_NO_OSS) && defined(MA_SUPPORT_OSS) - #define MA_ENABLE_OSS -#endif -#if !defined(MA_NO_AAUDIO) && defined(MA_SUPPORT_AAUDIO) - #define MA_ENABLE_AAUDIO -#endif -#if !defined(MA_NO_OPENSL) && defined(MA_SUPPORT_OPENSL) - #define MA_ENABLE_OPENSL -#endif -#if !defined(MA_NO_WEBAUDIO) && defined(MA_SUPPORT_WEBAUDIO) - #define MA_ENABLE_WEBAUDIO -#endif -#if !defined(MA_NO_NULL) && defined(MA_SUPPORT_NULL) - #define MA_ENABLE_NULL -#endif +/* General errors. */ +#define MA_ERROR -1 /* A generic error. */ +#define MA_INVALID_ARGS -2 +#define MA_INVALID_OPERATION -3 +#define MA_OUT_OF_MEMORY -4 +#define MA_ACCESS_DENIED -5 +#define MA_TOO_LARGE -6 +#define MA_TIMEOUT -7 -#ifdef MA_SUPPORT_WASAPI -/* We need a IMMNotificationClient object for WASAPI. */ -typedef struct -{ - void* lpVtbl; - ma_uint32 counter; - ma_device* pDevice; -} ma_IMMNotificationClient; -#endif +/* General miniaudio-specific errors. */ +#define MA_FORMAT_NOT_SUPPORTED -100 +#define MA_DEVICE_TYPE_NOT_SUPPORTED -101 +#define MA_SHARE_MODE_NOT_SUPPORTED -102 +#define MA_NO_BACKEND -103 +#define MA_NO_DEVICE -104 +#define MA_API_NOT_FOUND -105 +#define MA_INVALID_DEVICE_CONFIG -106 + +/* State errors. */ +#define MA_DEVICE_BUSY -200 +#define MA_DEVICE_NOT_INITIALIZED -201 +#define MA_DEVICE_NOT_STARTED -202 +#define MA_DEVICE_UNAVAILABLE -203 + +/* Operation errors. */ +#define MA_FAILED_TO_MAP_DEVICE_BUFFER -300 +#define MA_FAILED_TO_UNMAP_DEVICE_BUFFER -301 +#define MA_FAILED_TO_INIT_BACKEND -302 +#define MA_FAILED_TO_READ_DATA_FROM_CLIENT -303 +#define MA_FAILED_TO_READ_DATA_FROM_DEVICE -304 +#define MA_FAILED_TO_SEND_DATA_TO_CLIENT -305 +#define MA_FAILED_TO_SEND_DATA_TO_DEVICE -306 +#define MA_FAILED_TO_OPEN_BACKEND_DEVICE -307 +#define MA_FAILED_TO_START_BACKEND_DEVICE -308 +#define MA_FAILED_TO_STOP_BACKEND_DEVICE -309 +#define MA_FAILED_TO_CONFIGURE_BACKEND_DEVICE -310 +#define MA_FAILED_TO_CREATE_MUTEX -311 +#define MA_FAILED_TO_CREATE_EVENT -312 +#define MA_FAILED_TO_CREATE_SEMAPHORE -313 +#define MA_FAILED_TO_CREATE_THREAD -314 + + +/* Standard sample rates. */ +#define MA_SAMPLE_RATE_8000 8000 +#define MA_SAMPLE_RATE_11025 11025 +#define MA_SAMPLE_RATE_16000 16000 +#define MA_SAMPLE_RATE_22050 22050 +#define MA_SAMPLE_RATE_24000 24000 +#define MA_SAMPLE_RATE_32000 32000 +#define MA_SAMPLE_RATE_44100 44100 +#define MA_SAMPLE_RATE_48000 48000 +#define MA_SAMPLE_RATE_88200 88200 +#define MA_SAMPLE_RATE_96000 96000 +#define MA_SAMPLE_RATE_176400 176400 +#define MA_SAMPLE_RATE_192000 192000 +#define MA_SAMPLE_RATE_352800 352800 +#define MA_SAMPLE_RATE_384000 384000 + +#define MA_MIN_CHANNELS 1 +#define MA_MAX_CHANNELS 32 +#define MA_MIN_SAMPLE_RATE MA_SAMPLE_RATE_8000 +#define MA_MAX_SAMPLE_RATE MA_SAMPLE_RATE_384000 -/* Backend enums must be in priority order. */ typedef enum { - ma_backend_wasapi, - ma_backend_dsound, - ma_backend_winmm, - ma_backend_coreaudio, - ma_backend_sndio, - ma_backend_audio4, - ma_backend_oss, - ma_backend_pulseaudio, - ma_backend_alsa, - ma_backend_jack, - ma_backend_aaudio, - ma_backend_opensl, - ma_backend_webaudio, - ma_backend_null /* <-- Must always be the last item. Lowest priority, and used as the terminator for backend enumeration. */ -} ma_backend; + ma_stream_format_pcm = 0 +} ma_stream_format; -/* Thread priorties should be ordered such that the default priority of the worker thread is 0. */ typedef enum { - ma_thread_priority_idle = -5, - ma_thread_priority_lowest = -4, - ma_thread_priority_low = -3, - ma_thread_priority_normal = -2, - ma_thread_priority_high = -1, - ma_thread_priority_highest = 0, - ma_thread_priority_realtime = 1, - ma_thread_priority_default = 0 -} ma_thread_priority; + ma_stream_layout_interleaved = 0, + ma_stream_layout_deinterleaved +} ma_stream_layout; -typedef struct +typedef enum { - ma_context* pContext; + ma_dither_mode_none = 0, + ma_dither_mode_rectangle, + ma_dither_mode_triangle +} ma_dither_mode; - union - { -#ifdef MA_WIN32 - struct - { - /*HANDLE*/ ma_handle hThread; - } win32; -#endif -#ifdef MA_POSIX - struct - { - pthread_t thread; - } posix; +typedef enum +{ + /* + I like to keep these explicitly defined because they're used as a key into a lookup table. When items are + added to this, make sure there are no gaps and that they're added to the lookup table in ma_get_bytes_per_sample(). + */ + ma_format_unknown = 0, /* Mainly used for indicating an error, but also used as the default for the output format for decoders. */ + ma_format_u8 = 1, + ma_format_s16 = 2, /* Seems to be the most widely supported format. */ + ma_format_s24 = 3, /* Tightly packed. 3 bytes per sample. */ + ma_format_s32 = 4, + ma_format_f32 = 5, + ma_format_count +} ma_format; + +typedef enum +{ + ma_channel_mix_mode_rectangular = 0, /* Simple averaging based on the plane(s) the channel is sitting on. */ + ma_channel_mix_mode_simple, /* Drop excess channels; zeroed out extra channels. */ + ma_channel_mix_mode_custom_weights, /* Use custom weights specified in ma_channel_router_config. */ + ma_channel_mix_mode_planar_blend = ma_channel_mix_mode_rectangular, + ma_channel_mix_mode_default = ma_channel_mix_mode_planar_blend +} ma_channel_mix_mode; + +typedef enum +{ + ma_standard_channel_map_microsoft, + ma_standard_channel_map_alsa, + ma_standard_channel_map_rfc3551, /* Based off AIFF. */ + ma_standard_channel_map_flac, + ma_standard_channel_map_vorbis, + ma_standard_channel_map_sound4, /* FreeBSD's sound(4). */ + ma_standard_channel_map_sndio, /* www.sndio.org/tips.html */ + ma_standard_channel_map_webaudio = ma_standard_channel_map_flac, /* https://webaudio.github.io/web-audio-api/#ChannelOrdering. Only 1, 2, 4 and 6 channels are defined, but can fill in the gaps with logical assumptions. */ + ma_standard_channel_map_default = ma_standard_channel_map_microsoft +} ma_standard_channel_map; + +typedef enum +{ + ma_performance_profile_low_latency = 0, + ma_performance_profile_conservative +} ma_performance_profile; + + +typedef struct +{ + void* pUserData; + void* (* onMalloc)(size_t sz, void* pUserData); + void* (* onRealloc)(void* p, size_t sz, void* pUserData); + void (* onFree)(void* p, void* pUserData); +} ma_allocation_callbacks; + + +/************************************************************************************************************************************************************** + +Biquad Filtering + +**************************************************************************************************************************************************************/ +typedef union +{ + float f32; + ma_int32 s32; +} ma_biquad_coefficient; + +typedef struct +{ + ma_format format; + ma_uint32 channels; + double b0; + double b1; + double b2; + double a0; + double a1; + double a2; +} ma_biquad_config; + +ma_biquad_config ma_biquad_config_init(ma_format format, ma_uint32 channels, double b0, double b1, double b2, double a0, double a1, double a2); + +typedef struct +{ + ma_format format; + ma_uint32 channels; + ma_biquad_coefficient b0; + ma_biquad_coefficient b1; + ma_biquad_coefficient b2; + ma_biquad_coefficient a1; + ma_biquad_coefficient a2; + ma_biquad_coefficient r1[MA_MAX_CHANNELS]; + ma_biquad_coefficient r2[MA_MAX_CHANNELS]; +} ma_biquad; + +ma_result ma_biquad_init(const ma_biquad_config* pConfig, ma_biquad* pBQ); +ma_result ma_biquad_reinit(const ma_biquad_config* pConfig, ma_biquad* pBQ); +ma_result ma_biquad_process_pcm_frames(ma_biquad* pBQ, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); +ma_uint32 ma_biquad_get_latency(ma_biquad* pBQ); + + +/************************************************************************************************************************************************************** + +Low-Pass Filtering + +**************************************************************************************************************************************************************/ +typedef struct +{ + ma_format format; + ma_uint32 channels; + ma_uint32 sampleRate; + double cutoffFrequency; +} ma_lpf_config; + +ma_lpf_config ma_lpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency); + +typedef struct +{ + ma_biquad bq; /* The low-pass filter is implemented as a biquad filter. */ +} ma_lpf; + +ma_result ma_lpf_init(const ma_lpf_config* pConfig, ma_lpf* pLPF); +ma_result ma_lpf_reinit(const ma_lpf_config* pConfig, ma_lpf* pLPF); +ma_result ma_lpf_process_pcm_frames(ma_lpf* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); +ma_uint32 ma_lpf_get_latency(ma_lpf* pLPF); + + +/************************************************************************************************************************************************************** + +High-Pass Filtering + +**************************************************************************************************************************************************************/ +typedef struct +{ + ma_format format; + ma_uint32 channels; + ma_uint32 sampleRate; + double cutoffFrequency; +} ma_hpf_config; + +ma_hpf_config ma_hpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency); + +typedef struct +{ + ma_biquad bq; /* The high-pass filter is implemented as a biquad filter. */ +} ma_hpf; + +ma_result ma_hpf_init(const ma_hpf_config* pConfig, ma_hpf* pHPF); +ma_result ma_hpf_reinit(const ma_hpf_config* pConfig, ma_hpf* pHPF); +ma_result ma_hpf_process_pcm_frames(ma_hpf* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); +ma_uint32 ma_hpf_get_latency(ma_hpf* pHPF); + + +/************************************************************************************************************************************************************** + +Band-Pass Filtering + +**************************************************************************************************************************************************************/ +typedef struct +{ + ma_format format; + ma_uint32 channels; + ma_uint32 sampleRate; + double cutoffFrequency; +} ma_bpf_config; + +ma_bpf_config ma_bpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency); + +typedef struct +{ + ma_biquad bq; /* The band-pass filter is implemented as a biquad filter. */ +} ma_bpf; + +ma_result ma_bpf_init(const ma_bpf_config* pConfig, ma_bpf* pBPF); +ma_result ma_bpf_reinit(const ma_bpf_config* pConfig, ma_bpf* pBPF); +ma_result ma_bpf_process_pcm_frames(ma_bpf* pBPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); +ma_uint32 ma_bpf_get_latency(ma_bpf* pBPF); + + +/************************************************************************************************************************************************************ +************************************************************************************************************************************************************* + +DATA CONVERSION +=============== + +This section contains the APIs for data conversion. You will find everything here for channel mapping, sample format conversion, resampling, etc. + +************************************************************************************************************************************************************* +************************************************************************************************************************************************************/ + +/************************************************************************************************************************************************************** + +Resampling + +**************************************************************************************************************************************************************/ +#ifndef MA_MAX_RESAMPLER_LPF_FILTERS +#define MA_MAX_RESAMPLER_LPF_FILTERS 4 #endif - int _unused; - }; -} ma_thread; typedef struct { - ma_context* pContext; + ma_format format; + ma_uint32 channels; + ma_uint32 sampleRateIn; + ma_uint32 sampleRateOut; + ma_uint32 lpfCount; /* How many low-pass filters to chain together. A single low-pass filter is second order. Setting this to 0 will disable low-pass filtering. */ + double lpfNyquistFactor; /* 0..1. Defaults to 1. 1 = Half the sampling frequency (Nyquist Frequency), 0.5 = Quarter the sampling frequency (half Nyquest Frequency), etc. */ +} ma_linear_resampler_config; +ma_linear_resampler_config ma_linear_resampler_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut); + +typedef struct +{ + ma_linear_resampler_config config; + ma_uint32 inAdvanceInt; + ma_uint32 inAdvanceFrac; + ma_uint32 inTimeInt; + ma_uint32 inTimeFrac; union { -#ifdef MA_WIN32 - struct - { - /*HANDLE*/ ma_handle hMutex; - } win32; -#endif -#ifdef MA_POSIX - struct - { - pthread_mutex_t mutex; - } posix; -#endif - int _unused; - }; -} ma_mutex; + float f32[MA_MAX_CHANNELS]; + ma_int16 s16[MA_MAX_CHANNELS]; + } x0; /* The previous input frame. */ + union + { + float f32[MA_MAX_CHANNELS]; + ma_int16 s16[MA_MAX_CHANNELS]; + } x1; /* The next input frame. */ + ma_lpf lpf[MA_MAX_RESAMPLER_LPF_FILTERS]; +} ma_linear_resampler; + +ma_result ma_linear_resampler_init(const ma_linear_resampler_config* pConfig, ma_linear_resampler* pResampler); +void ma_linear_resampler_uninit(ma_linear_resampler* pResampler); +ma_result ma_linear_resampler_process_pcm_frames(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut); +ma_result ma_linear_resampler_set_rate(ma_linear_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut); +ma_result ma_linear_resampler_set_rate_ratio(ma_linear_resampler* pResampler, float ratioInOut); +ma_uint64 ma_linear_resampler_get_required_input_frame_count(ma_linear_resampler* pResampler, ma_uint64 outputFrameCount); +ma_uint64 ma_linear_resampler_get_expected_output_frame_count(ma_linear_resampler* pResampler, ma_uint64 inputFrameCount); +ma_uint64 ma_linear_resampler_get_input_latency(ma_linear_resampler* pResampler); +ma_uint64 ma_linear_resampler_get_output_latency(ma_linear_resampler* pResampler); + +typedef enum +{ + ma_resample_algorithm_linear = 0, /* Fastest, lowest quality. Optional low-pass filtering. Default. */ + ma_resample_algorithm_speex +} ma_resample_algorithm; typedef struct { - ma_context* pContext; + ma_format format; /* Must be either ma_format_f32 or ma_format_s16. */ + ma_uint32 channels; + ma_uint32 sampleRateIn; + ma_uint32 sampleRateOut; + ma_resample_algorithm algorithm; + struct + { + ma_uint32 lpfCount; + double lpfNyquistFactor; + } linear; + struct + { + int quality; /* 0 to 10. Defaults to 3. */ + } speex; +} ma_resampler_config; + +ma_resampler_config ma_resampler_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut, ma_resample_algorithm algorithm); +typedef struct +{ + ma_resampler_config config; union { -#ifdef MA_WIN32 - struct - { - /*HANDLE*/ ma_handle hEvent; - } win32; -#endif -#ifdef MA_POSIX + ma_linear_resampler linear; struct { - pthread_mutex_t mutex; - pthread_cond_t condition; - ma_uint32 value; - } posix; -#endif - int _unused; - }; -} ma_event; + void* pSpeexResamplerState; /* SpeexResamplerState* */ + } speex; + } state; +} ma_resampler; +/* +Initializes a new resampler object from a config. +*/ +ma_result ma_resampler_init(const ma_resampler_config* pConfig, ma_resampler* pResampler); /* -The callback for processing audio data from the device. +Uninitializes a resampler. +*/ +void ma_resampler_uninit(ma_resampler* pResampler); + +/* +Converts the given input data. + +Both the input and output frames must be in the format specified in the config when the resampler was initilized. -pOutput is a pointer to a buffer that will receive audio data that will later be played back through the speakers. This will be non-null -for a playback or full-duplex device and null for a capture device. +On input, [pFrameCountOut] contains the number of output frames to process. On output it contains the number of output frames that +were actually processed, which may be less than the requested amount which will happen if there's not enough input data. You can use +ma_resampler_get_expected_output_frame_count() to know how many output frames will be processed for a given number of input frames. -pInput is a pointer to a buffer containing input data from the device. This will be non-null for a capture or full-duplex device, and -null for a playback device. +On input, [pFrameCountIn] contains the number of input frames contained in [pFramesIn]. On output it contains the number of whole +input frames that were actually processed. You can use ma_resampler_get_required_input_frame_count() to know how many input frames +you should provide for a given number of output frames. [pFramesIn] can be NULL, in which case zeroes will be used instead. -frameCount is the number of PCM frames to process. If an output buffer is provided (pOutput is not null), applications should write out -to the entire output buffer. Note that frameCount will not necessarily be exactly what you asked for when you initialized the deviced. -The bufferSizeInFrames and bufferSizeInMilliseconds members of the device config are just hints, and are not necessarily exactly what -you'll get. +If [pFramesOut] is NULL, a seek is performed. In this case, if [pFrameCountOut] is not NULL it will seek by the specified number of +output frames. Otherwise, if [pFramesCountOut] is NULL and [pFrameCountIn] is not NULL, it will seek by the specified number of input +frames. When seeking, [pFramesIn] is allowed to NULL, in which case the internal timing state will be updated, but no input will be +processed. In this case, any internal filter state will be updated as if zeroes were passed in. -Do _not_ call any miniaudio APIs from the callback. Attempting the stop the device can result in a deadlock. The proper way to stop the -device is to call ma_device_stop() from a different thread, normally the main application thread. +It is an error for [pFramesOut] to be non-NULL and [pFrameCountOut] to be NULL. + +It is an error for both [pFrameCountOut] and [pFrameCountIn] to be NULL. */ -typedef void (* ma_device_callback_proc)(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount); +ma_result ma_resampler_process_pcm_frames(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut); + /* -The callback for when the device has been stopped. +Sets the input and output sample sample rate. +*/ +ma_result ma_resampler_set_rate(ma_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut); -This will be called when the device is stopped explicitly with ma_device_stop() and also called implicitly when the device is stopped -through external forces such as being unplugged or an internal error occuring. +/* +Sets the input and output sample rate as a ratio. -Do not restart the device from the callback. +The ration is in/out. */ -typedef void (* ma_stop_proc)(ma_device* pDevice); +ma_result ma_resampler_set_rate_ratio(ma_resampler* pResampler, float ratio); + /* -The callback for handling log messages. +Calculates the number of whole input frames that would need to be read from the client in order to output the specified +number of output frames. -It is possible for pDevice to be null in which case the log originated from the context. If it is non-null you can assume the message -came from the device. +The returned value does not include cached input frames. It only returns the number of extra frames that would need to be +read from the input buffer in order to output the specified number of output frames. +*/ +ma_uint64 ma_resampler_get_required_input_frame_count(ma_resampler* pResampler, ma_uint64 outputFrameCount); -logLevel is one of the following: - MA_LOG_LEVEL_VERBOSE - MA_LOG_LEVEL_INFO - MA_LOG_LEVEL_WARNING - MA_LOG_LEVEL_ERROR +/* +Calculates the number of whole output frames that would be output after fully reading and consuming the specified number of +input frames. */ -typedef void (* ma_log_proc)(ma_context* pContext, ma_device* pDevice, ma_uint32 logLevel, const char* message); +ma_uint64 ma_resampler_get_expected_output_frame_count(ma_resampler* pResampler, ma_uint64 inputFrameCount); -typedef enum -{ - ma_device_type_playback = 1, - ma_device_type_capture = 2, - ma_device_type_duplex = ma_device_type_playback | ma_device_type_capture, /* 3 */ - ma_device_type_loopback = 4 -} ma_device_type; -typedef enum +/* +Retrieves the latency introduced by the resampler in input frames. +*/ +ma_uint64 ma_resampler_get_input_latency(ma_resampler* pResampler); + +/* +Retrieves the latency introduced by the resampler in output frames. +*/ +ma_uint64 ma_resampler_get_output_latency(ma_resampler* pResampler); + + + +/************************************************************************************************************************************************************** + +Channel Conversion + +**************************************************************************************************************************************************************/ +typedef struct { - ma_share_mode_shared = 0, - ma_share_mode_exclusive -} ma_share_mode; + ma_format format; + ma_uint32 channelsIn; + ma_uint32 channelsOut; + ma_channel channelMapIn[MA_MAX_CHANNELS]; + ma_channel channelMapOut[MA_MAX_CHANNELS]; + ma_channel_mix_mode mixingMode; + float weights[MA_MAX_CHANNELS][MA_MAX_CHANNELS]; /* [in][out]. Only used when mixingMode is set to ma_channel_mix_mode_custom_weights. */ +} ma_channel_converter_config; -typedef union +ma_channel_converter_config ma_channel_converter_config_init(ma_format format, ma_uint32 channelsIn, const ma_channel channelMapIn[MA_MAX_CHANNELS], ma_uint32 channelsOut, const ma_channel channelMapOut[MA_MAX_CHANNELS], ma_channel_mix_mode mixingMode); + +typedef struct { -#ifdef MA_SUPPORT_WASAPI - wchar_t wasapi[64]; /* WASAPI uses a wchar_t string for identification. */ -#endif -#ifdef MA_SUPPORT_DSOUND - ma_uint8 dsound[16]; /* DirectSound uses a GUID for identification. */ -#endif -#ifdef MA_SUPPORT_WINMM - /*UINT_PTR*/ ma_uint32 winmm; /* When creating a device, WinMM expects a Win32 UINT_PTR for device identification. In practice it's actually just a UINT. */ -#endif -#ifdef MA_SUPPORT_ALSA - char alsa[256]; /* ALSA uses a name string for identification. */ -#endif -#ifdef MA_SUPPORT_PULSEAUDIO - char pulse[256]; /* PulseAudio uses a name string for identification. */ -#endif -#ifdef MA_SUPPORT_JACK - int jack; /* JACK always uses default devices. */ -#endif -#ifdef MA_SUPPORT_COREAUDIO - char coreaudio[256]; /* Core Audio uses a string for identification. */ -#endif -#ifdef MA_SUPPORT_SNDIO - char sndio[256]; /* "snd/0", etc. */ -#endif -#ifdef MA_SUPPORT_AUDIO4 - char audio4[256]; /* "/dev/audio", etc. */ -#endif -#ifdef MA_SUPPORT_OSS - char oss[64]; /* "dev/dsp0", etc. "dev/dsp" for the default device. */ -#endif -#ifdef MA_SUPPORT_AAUDIO - ma_int32 aaudio; /* AAudio uses a 32-bit integer for identification. */ -#endif -#ifdef MA_SUPPORT_OPENSL - ma_uint32 opensl; /* OpenSL|ES uses a 32-bit unsigned integer for identification. */ -#endif -#ifdef MA_SUPPORT_WEBAUDIO - char webaudio[32]; /* Web Audio always uses default devices for now, but if this changes it'll be a GUID. */ -#endif -#ifdef MA_SUPPORT_NULL - int nullbackend; /* The null backend uses an integer for device IDs. */ -#endif -} ma_device_id; + ma_format format; + ma_uint32 channelsIn; + ma_uint32 channelsOut; + ma_channel channelMapIn[MA_MAX_CHANNELS]; + ma_channel channelMapOut[MA_MAX_CHANNELS]; + ma_channel_mix_mode mixingMode; + union + { + float f32[MA_MAX_CHANNELS][MA_MAX_CHANNELS]; + ma_int32 s16[MA_MAX_CHANNELS][MA_MAX_CHANNELS]; + } weights; + ma_bool32 isPassthrough : 1; + ma_bool32 isSimpleShuffle : 1; + ma_bool32 isSimpleMonoExpansion : 1; + ma_bool32 isStereoToMono : 1; + ma_uint8 shuffleTable[MA_MAX_CHANNELS]; +} ma_channel_converter; -typedef struct -{ - /* Basic info. This is the only information guaranteed to be filled in during device enumeration. */ - ma_device_id id; - char name[256]; +ma_result ma_channel_converter_init(const ma_channel_converter_config* pConfig, ma_channel_converter* pConverter); +void ma_channel_converter_uninit(ma_channel_converter* pConverter); +ma_result ma_channel_converter_process_pcm_frames(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount); - /* - Detailed info. As much of this is filled as possible with ma_context_get_device_info(). Note that you are allowed to initialize - a device with settings outside of this range, but it just means the data will be converted using miniaudio's data conversion - pipeline before sending the data to/from the device. Most programs will need to not worry about these values, but it's provided - here mainly for informational purposes or in the rare case that someone might find it useful. - - These will be set to 0 when returned by ma_context_enumerate_devices() or ma_context_get_devices(). - */ - ma_uint32 formatCount; - ma_format formats[ma_format_count]; - ma_uint32 minChannels; - ma_uint32 maxChannels; - ma_uint32 minSampleRate; - ma_uint32 maxSampleRate; -} ma_device_info; -typedef union -{ - ma_int64 counter; - double counterD; -} ma_timer; +/************************************************************************************************************************************************************** +Data Conversion + +**************************************************************************************************************************************************************/ typedef struct { - ma_device_type deviceType; - ma_uint32 sampleRate; - ma_uint32 bufferSizeInFrames; - ma_uint32 bufferSizeInMilliseconds; - ma_uint32 periods; - ma_performance_profile performanceProfile; - ma_bool32 noPreZeroedOutputBuffer; /* When set to true, the contents of the output buffer passed into the data callback will be left undefined rather than initialized to zero. */ - ma_bool32 noClip; /* When set to true, the contents of the output buffer passed into the data callback will be clipped after returning. Only applies when the playback sample format is f32. */ - ma_device_callback_proc dataCallback; - ma_stop_proc stopCallback; - void* pUserData; - struct - { - ma_device_id* pDeviceID; - ma_format format; - ma_uint32 channels; - ma_channel channelMap[MA_MAX_CHANNELS]; - ma_share_mode shareMode; - } playback; + ma_format formatIn; + ma_format formatOut; + ma_uint32 channelsIn; + ma_uint32 channelsOut; + ma_uint32 sampleRateIn; + ma_uint32 sampleRateOut; + ma_channel channelMapIn[MA_MAX_CHANNELS]; + ma_channel channelMapOut[MA_MAX_CHANNELS]; + ma_dither_mode ditherMode; + ma_channel_mix_mode channelMixMode; + float channelWeights[MA_MAX_CHANNELS][MA_MAX_CHANNELS]; /* [in][out]. Only used when channelMixMode is set to ma_channel_mix_mode_custom_weights. */ struct { - ma_device_id* pDeviceID; - ma_format format; - ma_uint32 channels; - ma_channel channelMap[MA_MAX_CHANNELS]; - ma_share_mode shareMode; - } capture; + ma_resample_algorithm algorithm; + ma_bool32 allowDynamicSampleRate; + struct + { + ma_uint32 lpfCount; + double lpfNyquistFactor; + } linear; + struct + { + int quality; + } speex; + } resampling; +} ma_data_converter_config; - struct - { - ma_bool32 noAutoConvertSRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM. */ - ma_bool32 noDefaultQualitySRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY. */ - } wasapi; - struct - { - ma_bool32 noMMap; /* Disables MMap mode. */ - } alsa; - struct - { - const char* pStreamNamePlayback; - const char* pStreamNameCapture; - } pulse; -} ma_device_config; +ma_data_converter_config ma_data_converter_config_init_default(void); +ma_data_converter_config ma_data_converter_config_init(ma_format formatIn, ma_format formatOut, ma_uint32 channelsIn, ma_uint32 channelsOut, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut); typedef struct { - ma_log_proc logCallback; - ma_thread_priority threadPriority; - void* pUserData; + ma_data_converter_config config; + ma_channel_converter channelConverter; + ma_resampler resampler; + ma_bool32 hasPreFormatConversion : 1; + ma_bool32 hasPostFormatConversion : 1; + ma_bool32 hasChannelConverter : 1; + ma_bool32 hasResampler : 1; + ma_bool32 isPassthrough : 1; +} ma_data_converter; + +ma_result ma_data_converter_init(const ma_data_converter_config* pConfig, ma_data_converter* pConverter); +void ma_data_converter_uninit(ma_data_converter* pConverter); +ma_result ma_data_converter_process_pcm_frames(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut); +ma_result ma_data_converter_set_rate(ma_data_converter* pConverter, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut); +ma_result ma_data_converter_set_rate_ratio(ma_data_converter* pConverter, float ratioInOut); +ma_uint64 ma_data_converter_get_required_input_frame_count(ma_data_converter* pConverter, ma_uint64 outputFrameCount); +ma_uint64 ma_data_converter_get_expected_output_frame_count(ma_data_converter* pConverter, ma_uint64 inputFrameCount); +ma_uint64 ma_data_converter_get_input_latency(ma_data_converter* pConverter); +ma_uint64 ma_data_converter_get_output_latency(ma_data_converter* pConverter); - struct - { - ma_bool32 useVerboseDeviceEnumeration; - } alsa; - struct - { - const char* pApplicationName; - const char* pServerName; - ma_bool32 tryAutoSpawn; /* Enables autospawning of the PulseAudio daemon if necessary. */ - } pulse; - struct - { - const char* pClientName; - ma_bool32 tryStartServer; - } jack; -} ma_context_config; -typedef ma_bool32 (* ma_enum_devices_callback_proc)(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pInfo, void* pUserData); +/************************************************************************************************************************************************************ -struct ma_context -{ - ma_backend backend; /* DirectSound, ALSA, etc. */ - ma_log_proc logCallback; - ma_thread_priority threadPriority; - void* pUserData; - ma_mutex deviceEnumLock; /* Used to make ma_context_get_devices() thread safe. */ - ma_mutex deviceInfoLock; /* Used to make ma_context_get_device_info() thread safe. */ - ma_uint32 deviceInfoCapacity; /* Total capacity of pDeviceInfos. */ - ma_uint32 playbackDeviceInfoCount; - ma_uint32 captureDeviceInfoCount; - ma_device_info* pDeviceInfos; /* Playback devices first, then capture. */ - ma_bool32 isBackendAsynchronous : 1; /* Set when the context is initialized. Set to 1 for asynchronous backends such as Core Audio and JACK. Do not modify. */ +Format Conversion - ma_result (* onUninit )(ma_context* pContext); - ma_bool32 (* onDeviceIDEqual )(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1); - ma_result (* onEnumDevices )(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData); /* Return false from the callback to stop enumeration. */ - ma_result (* onGetDeviceInfo )(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo); - ma_result (* onDeviceInit )(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice); - void (* onDeviceUninit )(ma_device* pDevice); - ma_result (* onDeviceStart )(ma_device* pDevice); - ma_result (* onDeviceStop )(ma_device* pDevice); - ma_result (* onDeviceMainLoop)(ma_device* pDevice); +************************************************************************************************************************************************************/ +void ma_pcm_u8_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_u8_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_u8_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_u8_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_s16_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_s16_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_s16_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_s16_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_s24_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_s24_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_s24_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_s24_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_s32_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_s32_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_s32_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_s32_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_f32_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_f32_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_f32_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_f32_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode); +void ma_pcm_convert(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 sampleCount, ma_dither_mode ditherMode); +void ma_convert_pcm_frames_format(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 frameCount, ma_uint32 channels, ma_dither_mode ditherMode); - union - { -#ifdef MA_SUPPORT_WASAPI - struct - { - int _unused; - } wasapi; -#endif -#ifdef MA_SUPPORT_DSOUND - struct - { - ma_handle hDSoundDLL; - ma_proc DirectSoundCreate; - ma_proc DirectSoundEnumerateA; - ma_proc DirectSoundCaptureCreate; - ma_proc DirectSoundCaptureEnumerateA; - } dsound; -#endif -#ifdef MA_SUPPORT_WINMM - struct - { - ma_handle hWinMM; - ma_proc waveOutGetNumDevs; - ma_proc waveOutGetDevCapsA; - ma_proc waveOutOpen; - ma_proc waveOutClose; - ma_proc waveOutPrepareHeader; - ma_proc waveOutUnprepareHeader; - ma_proc waveOutWrite; - ma_proc waveOutReset; - ma_proc waveInGetNumDevs; - ma_proc waveInGetDevCapsA; - ma_proc waveInOpen; - ma_proc waveInClose; - ma_proc waveInPrepareHeader; - ma_proc waveInUnprepareHeader; - ma_proc waveInAddBuffer; - ma_proc waveInStart; - ma_proc waveInReset; - } winmm; -#endif -#ifdef MA_SUPPORT_ALSA - struct - { - ma_handle asoundSO; - ma_proc snd_pcm_open; - ma_proc snd_pcm_close; - ma_proc snd_pcm_hw_params_sizeof; - ma_proc snd_pcm_hw_params_any; - ma_proc snd_pcm_hw_params_set_format; - ma_proc snd_pcm_hw_params_set_format_first; - ma_proc snd_pcm_hw_params_get_format_mask; - ma_proc snd_pcm_hw_params_set_channels_near; - ma_proc snd_pcm_hw_params_set_rate_resample; - ma_proc snd_pcm_hw_params_set_rate_near; - ma_proc snd_pcm_hw_params_set_buffer_size_near; - ma_proc snd_pcm_hw_params_set_periods_near; - ma_proc snd_pcm_hw_params_set_access; - ma_proc snd_pcm_hw_params_get_format; +/* +Deinterleaves an interleaved buffer. +*/ +void ma_deinterleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void* pInterleavedPCMFrames, void** ppDeinterleavedPCMFrames); + +/* +Interleaves a group of deinterleaved buffers. +*/ +void ma_interleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void** ppDeinterleavedPCMFrames, void* pInterleavedPCMFrames); + +/************************************************************************************************************************************************************ + +Channel Maps + +************************************************************************************************************************************************************/ + +/* +Helper for retrieving a standard channel map. +*/ +void ma_get_standard_channel_map(ma_standard_channel_map standardChannelMap, ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]); + +/* +Copies a channel map. +*/ +void ma_channel_map_copy(ma_channel* pOut, const ma_channel* pIn, ma_uint32 channels); + + +/* +Determines whether or not a channel map is valid. + +A blank channel map is valid (all channels set to MA_CHANNEL_NONE). The way a blank channel map is handled is context specific, but +is usually treated as a passthrough. + +Invalid channel maps: + - A channel map with no channels + - A channel map with more than one channel and a mono channel +*/ +ma_bool32 ma_channel_map_valid(ma_uint32 channels, const ma_channel channelMap[MA_MAX_CHANNELS]); + +/* +Helper for comparing two channel maps for equality. + +This assumes the channel count is the same between the two. +*/ +ma_bool32 ma_channel_map_equal(ma_uint32 channels, const ma_channel channelMapA[MA_MAX_CHANNELS], const ma_channel channelMapB[MA_MAX_CHANNELS]); + +/* +Helper for determining if a channel map is blank (all channels set to MA_CHANNEL_NONE). +*/ +ma_bool32 ma_channel_map_blank(ma_uint32 channels, const ma_channel channelMap[MA_MAX_CHANNELS]); + +/* +Helper for determining whether or not a channel is present in the given channel map. +*/ +ma_bool32 ma_channel_map_contains_channel_position(ma_uint32 channels, const ma_channel channelMap[MA_MAX_CHANNELS], ma_channel channelPosition); + + +/************************************************************************************************************************************************************ + +Conversion Helpers + +************************************************************************************************************************************************************/ + +/* +High-level helper for doing a full format conversion in one go. Returns the number of output frames. Call this with pOut set to NULL to +determine the required size of the output buffer. frameCountOut should be set to the capacity of pOut. If pOut is NULL, frameCountOut is +ignored. + +A return value of 0 indicates an error. + +This function is useful for one-off bulk conversions, but if you're streaming data you should use the ma_data_converter APIs instead. +*/ +ma_uint64 ma_convert_frames(void* pOut, ma_uint64 frameCountOut, ma_format formatOut, ma_uint32 channelsOut, ma_uint32 sampleRateOut, const void* pIn, ma_uint64 frameCountIn, ma_format formatIn, ma_uint32 channelsIn, ma_uint32 sampleRateIn); +ma_uint64 ma_convert_frames_ex(void* pOut, ma_uint64 frameCountOut, const void* pIn, ma_uint64 frameCountIn, const ma_data_converter_config* pConfig); + + +/************************************************************************************************************************************************************ + +Ring Buffer + +************************************************************************************************************************************************************/ +typedef struct +{ + void* pBuffer; + ma_uint32 subbufferSizeInBytes; + ma_uint32 subbufferCount; + ma_uint32 subbufferStrideInBytes; + volatile ma_uint32 encodedReadOffset; /* Most significant bit is the loop flag. Lower 31 bits contains the actual offset in bytes. */ + volatile ma_uint32 encodedWriteOffset; /* Most significant bit is the loop flag. Lower 31 bits contains the actual offset in bytes. */ + ma_bool32 ownsBuffer : 1; /* Used to know whether or not miniaudio is responsible for free()-ing the buffer. */ + ma_bool32 clearOnWriteAcquire : 1; /* When set, clears the acquired write buffer before returning from ma_rb_acquire_write(). */ + ma_allocation_callbacks allocationCallbacks; +} ma_rb; + +ma_result ma_rb_init_ex(size_t subbufferSizeInBytes, size_t subbufferCount, size_t subbufferStrideInBytes, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_rb* pRB); +ma_result ma_rb_init(size_t bufferSizeInBytes, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_rb* pRB); +void ma_rb_uninit(ma_rb* pRB); +void ma_rb_reset(ma_rb* pRB); +ma_result ma_rb_acquire_read(ma_rb* pRB, size_t* pSizeInBytes, void** ppBufferOut); +ma_result ma_rb_commit_read(ma_rb* pRB, size_t sizeInBytes, void* pBufferOut); +ma_result ma_rb_acquire_write(ma_rb* pRB, size_t* pSizeInBytes, void** ppBufferOut); +ma_result ma_rb_commit_write(ma_rb* pRB, size_t sizeInBytes, void* pBufferOut); +ma_result ma_rb_seek_read(ma_rb* pRB, size_t offsetInBytes); +ma_result ma_rb_seek_write(ma_rb* pRB, size_t offsetInBytes); +ma_int32 ma_rb_pointer_distance(ma_rb* pRB); /* Returns the distance between the write pointer and the read pointer. Should never be negative for a correct program. Will return the number of bytes that can be read before the read pointer hits the write pointer. */ +ma_uint32 ma_rb_available_read(ma_rb* pRB); +ma_uint32 ma_rb_available_write(ma_rb* pRB); +size_t ma_rb_get_subbuffer_size(ma_rb* pRB); +size_t ma_rb_get_subbuffer_stride(ma_rb* pRB); +size_t ma_rb_get_subbuffer_offset(ma_rb* pRB, size_t subbufferIndex); +void* ma_rb_get_subbuffer_ptr(ma_rb* pRB, size_t subbufferIndex, void* pBuffer); + + +typedef struct +{ + ma_rb rb; + ma_format format; + ma_uint32 channels; +} ma_pcm_rb; + +ma_result ma_pcm_rb_init_ex(ma_format format, ma_uint32 channels, ma_uint32 subbufferSizeInFrames, ma_uint32 subbufferCount, ma_uint32 subbufferStrideInFrames, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_pcm_rb* pRB); +ma_result ma_pcm_rb_init(ma_format format, ma_uint32 channels, ma_uint32 bufferSizeInFrames, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_pcm_rb* pRB); +void ma_pcm_rb_uninit(ma_pcm_rb* pRB); +void ma_pcm_rb_reset(ma_pcm_rb* pRB); +ma_result ma_pcm_rb_acquire_read(ma_pcm_rb* pRB, ma_uint32* pSizeInFrames, void** ppBufferOut); +ma_result ma_pcm_rb_commit_read(ma_pcm_rb* pRB, ma_uint32 sizeInFrames, void* pBufferOut); +ma_result ma_pcm_rb_acquire_write(ma_pcm_rb* pRB, ma_uint32* pSizeInFrames, void** ppBufferOut); +ma_result ma_pcm_rb_commit_write(ma_pcm_rb* pRB, ma_uint32 sizeInFrames, void* pBufferOut); +ma_result ma_pcm_rb_seek_read(ma_pcm_rb* pRB, ma_uint32 offsetInFrames); +ma_result ma_pcm_rb_seek_write(ma_pcm_rb* pRB, ma_uint32 offsetInFrames); +ma_int32 ma_pcm_rb_pointer_disance(ma_pcm_rb* pRB); /* Return value is in frames. */ +ma_uint32 ma_pcm_rb_available_read(ma_pcm_rb* pRB); +ma_uint32 ma_pcm_rb_available_write(ma_pcm_rb* pRB); +ma_uint32 ma_pcm_rb_get_subbuffer_size(ma_pcm_rb* pRB); +ma_uint32 ma_pcm_rb_get_subbuffer_stride(ma_pcm_rb* pRB); +ma_uint32 ma_pcm_rb_get_subbuffer_offset(ma_pcm_rb* pRB, ma_uint32 subbufferIndex); +void* ma_pcm_rb_get_subbuffer_ptr(ma_pcm_rb* pRB, ma_uint32 subbufferIndex, void* pBuffer); + + +/************************************************************************************************************************************************************ + +Miscellaneous Helpers + +************************************************************************************************************************************************************/ + +/* +malloc(). Calls MA_MALLOC(). +*/ +void* ma_malloc(size_t sz, const ma_allocation_callbacks* pAllocationCallbacks); + +/* +realloc(). Calls MA_REALLOC(). +*/ +void* ma_realloc(void* p, size_t sz, const ma_allocation_callbacks* pAllocationCallbacks); + +/* +free(). Calls MA_FREE(). +*/ +void ma_free(void* p, const ma_allocation_callbacks* pAllocationCallbacks); + +/* +Performs an aligned malloc, with the assumption that the alignment is a power of 2. +*/ +void* ma_aligned_malloc(size_t sz, size_t alignment, const ma_allocation_callbacks* pAllocationCallbacks); + +/* +Free's an aligned malloc'd buffer. +*/ +void ma_aligned_free(void* p, const ma_allocation_callbacks* pAllocationCallbacks); + +/* +Retrieves a friendly name for a format. +*/ +const char* ma_get_format_name(ma_format format); + +/* +Blends two frames in floating point format. +*/ +void ma_blend_f32(float* pOut, float* pInA, float* pInB, float factor, ma_uint32 channels); + +/* +Retrieves the size of a sample in bytes for the given format. + +This API is efficient and is implemented using a lookup table. + +Thread Safety: SAFE + This API is pure. +*/ +ma_uint32 ma_get_bytes_per_sample(ma_format format); +static MA_INLINE ma_uint32 ma_get_bytes_per_frame(ma_format format, ma_uint32 channels) { return ma_get_bytes_per_sample(format) * channels; } + +/* +Converts a log level to a string. +*/ +const char* ma_log_level_to_string(ma_uint32 logLevel); + + + +/************************************************************************************************************************************************************ +************************************************************************************************************************************************************* + +DEVICE I/O +========== + +This section contains the APIs for device playback and capture. Here is where you'll find ma_device_init(), etc. + +************************************************************************************************************************************************************* +************************************************************************************************************************************************************/ +#ifndef MA_NO_DEVICE_IO +/* Some backends are only supported on certain platforms. */ +#if defined(MA_WIN32) + #define MA_SUPPORT_WASAPI + #if defined(MA_WIN32_DESKTOP) /* DirectSound and WinMM backends are only supported on desktops. */ + #define MA_SUPPORT_DSOUND + #define MA_SUPPORT_WINMM + #define MA_SUPPORT_JACK /* JACK is technically supported on Windows, but I don't know how many people use it in practice... */ + #endif +#endif +#if defined(MA_UNIX) + #if defined(MA_LINUX) + #if !defined(MA_ANDROID) /* ALSA is not supported on Android. */ + #define MA_SUPPORT_ALSA + #endif + #endif + #if !defined(MA_BSD) && !defined(MA_ANDROID) && !defined(MA_EMSCRIPTEN) + #define MA_SUPPORT_PULSEAUDIO + #define MA_SUPPORT_JACK + #endif + #if defined(MA_ANDROID) + #define MA_SUPPORT_AAUDIO + #define MA_SUPPORT_OPENSL + #endif + #if defined(__OpenBSD__) /* <-- Change this to "#if defined(MA_BSD)" to enable sndio on all BSD flavors. */ + #define MA_SUPPORT_SNDIO /* sndio is only supported on OpenBSD for now. May be expanded later if there's demand. */ + #endif + #if defined(__NetBSD__) || defined(__OpenBSD__) + #define MA_SUPPORT_AUDIO4 /* Only support audio(4) on platforms with known support. */ + #endif + #if defined(__FreeBSD__) || defined(__DragonFly__) + #define MA_SUPPORT_OSS /* Only support OSS on specific platforms with known support. */ + #endif +#endif +#if defined(MA_APPLE) + #define MA_SUPPORT_COREAUDIO +#endif +#if defined(MA_EMSCRIPTEN) + #define MA_SUPPORT_WEBAUDIO +#endif + +/* Explicitly disable the Null backend for Emscripten because it uses a background thread which is not properly supported right now. */ +#if !defined(MA_EMSCRIPTEN) +#define MA_SUPPORT_NULL +#endif + + +#if !defined(MA_NO_WASAPI) && defined(MA_SUPPORT_WASAPI) + #define MA_ENABLE_WASAPI +#endif +#if !defined(MA_NO_DSOUND) && defined(MA_SUPPORT_DSOUND) + #define MA_ENABLE_DSOUND +#endif +#if !defined(MA_NO_WINMM) && defined(MA_SUPPORT_WINMM) + #define MA_ENABLE_WINMM +#endif +#if !defined(MA_NO_ALSA) && defined(MA_SUPPORT_ALSA) + #define MA_ENABLE_ALSA +#endif +#if !defined(MA_NO_PULSEAUDIO) && defined(MA_SUPPORT_PULSEAUDIO) + #define MA_ENABLE_PULSEAUDIO +#endif +#if !defined(MA_NO_JACK) && defined(MA_SUPPORT_JACK) + #define MA_ENABLE_JACK +#endif +#if !defined(MA_NO_COREAUDIO) && defined(MA_SUPPORT_COREAUDIO) + #define MA_ENABLE_COREAUDIO +#endif +#if !defined(MA_NO_SNDIO) && defined(MA_SUPPORT_SNDIO) + #define MA_ENABLE_SNDIO +#endif +#if !defined(MA_NO_AUDIO4) && defined(MA_SUPPORT_AUDIO4) + #define MA_ENABLE_AUDIO4 +#endif +#if !defined(MA_NO_OSS) && defined(MA_SUPPORT_OSS) + #define MA_ENABLE_OSS +#endif +#if !defined(MA_NO_AAUDIO) && defined(MA_SUPPORT_AAUDIO) + #define MA_ENABLE_AAUDIO +#endif +#if !defined(MA_NO_OPENSL) && defined(MA_SUPPORT_OPENSL) + #define MA_ENABLE_OPENSL +#endif +#if !defined(MA_NO_WEBAUDIO) && defined(MA_SUPPORT_WEBAUDIO) + #define MA_ENABLE_WEBAUDIO +#endif +#if !defined(MA_NO_NULL) && defined(MA_SUPPORT_NULL) + #define MA_ENABLE_NULL +#endif + +#ifdef MA_SUPPORT_WASAPI +/* We need a IMMNotificationClient object for WASAPI. */ +typedef struct +{ + void* lpVtbl; + ma_uint32 counter; + ma_device* pDevice; +} ma_IMMNotificationClient; +#endif + +/* Backend enums must be in priority order. */ +typedef enum +{ + ma_backend_wasapi, + ma_backend_dsound, + ma_backend_winmm, + ma_backend_coreaudio, + ma_backend_sndio, + ma_backend_audio4, + ma_backend_oss, + ma_backend_pulseaudio, + ma_backend_alsa, + ma_backend_jack, + ma_backend_aaudio, + ma_backend_opensl, + ma_backend_webaudio, + ma_backend_null /* <-- Must always be the last item. Lowest priority, and used as the terminator for backend enumeration. */ +} ma_backend; + +/* Thread priorties should be ordered such that the default priority of the worker thread is 0. */ +typedef enum +{ + ma_thread_priority_idle = -5, + ma_thread_priority_lowest = -4, + ma_thread_priority_low = -3, + ma_thread_priority_normal = -2, + ma_thread_priority_high = -1, + ma_thread_priority_highest = 0, + ma_thread_priority_realtime = 1, + ma_thread_priority_default = 0 +} ma_thread_priority; + +typedef struct +{ + ma_context* pContext; + + union + { +#ifdef MA_WIN32 + struct + { + /*HANDLE*/ ma_handle hThread; + } win32; +#endif +#ifdef MA_POSIX + struct + { + pthread_t thread; + } posix; +#endif + int _unused; + }; +} ma_thread; + +typedef struct +{ + ma_context* pContext; + + union + { +#ifdef MA_WIN32 + struct + { + /*HANDLE*/ ma_handle hMutex; + } win32; +#endif +#ifdef MA_POSIX + struct + { + pthread_mutex_t mutex; + } posix; +#endif + int _unused; + }; +} ma_mutex; + +typedef struct +{ + ma_context* pContext; + + union + { +#ifdef MA_WIN32 + struct + { + /*HANDLE*/ ma_handle hEvent; + } win32; +#endif +#ifdef MA_POSIX + struct + { + pthread_mutex_t mutex; + pthread_cond_t condition; + ma_uint32 value; + } posix; +#endif + int _unused; + }; +} ma_event; + +typedef struct +{ + ma_context* pContext; + + union + { +#ifdef MA_WIN32 + struct + { + /*HANDLE*/ ma_handle hSemaphore; + } win32; +#endif +#ifdef MA_POSIX + struct + { + sem_t semaphore; + } posix; +#endif + int _unused; + }; +} ma_semaphore; + + +/* +The callback for processing audio data from the device. + +The data callback is fired by miniaudio whenever the device needs to have more data delivered to a playback device, or when a capture device has some data +available. This is called as soon as the backend asks for more data which means it may be called with inconsistent frame counts. You cannot assume the +callback will be fired with a consistent frame count. + + +Parameters +---------- +pDevice (in) + A pointer to the relevant device. + +pOutput (out) + A pointer to the output buffer that will receive audio data that will later be played back through the speakers. This will be non-null for a playback or + full-duplex device and null for a capture and loopback device. + +pInput (in) + A pointer to the buffer containing input data from a recording device. This will be non-null for a capture, full-duplex or loopback device and null for a + playback device. + +frameCount (in) + The number of PCM frames to process. Note that this will not necessarily be equal to what you requested when you initialized the device. The + `periodSizeInFrames` and `periodSizeInMilliseconds` members of the device config are just hints, and are not necessarily exactly what you'll get. You must + not assume this will always be the same value each time the callback is fired. + + +Remarks +------- +You cannot stop and start the device from inside the callback or else you'll get a deadlock. You must also not uninitialize the device from inside the +callback. The following APIs cannot be called from inside the callback: + + ma_device_init() + ma_device_init_ex() + ma_device_uninit() + ma_device_start() + ma_device_stop() + +The proper way to stop the device is to call `ma_device_stop()` from a different thread, normally the main application thread. +*/ +typedef void (* ma_device_callback_proc)(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount); + +/* +The callback for when the device has been stopped. + +This will be called when the device is stopped explicitly with `ma_device_stop()` and also called implicitly when the device is stopped through external forces +such as being unplugged or an internal error occuring. + + +Parameters +---------- +pDevice (in) + A pointer to the device that has just stopped. + + +Remarks +------- +Do not restart or uninitialize the device from the callback. +*/ +typedef void (* ma_stop_proc)(ma_device* pDevice); + +/* +The callback for handling log messages. + + +Parameters +---------- +pContext (in) + A pointer to the context the log message originated from. + +pDevice (in) + A pointer to the device the log message originate from, if any. This can be null, in which case the message came from the context. + +logLevel (in) + The log level. This can be one of the following: + + |----------------------| + | Log Level | + |----------------------| + | MA_LOG_LEVEL_VERBOSE | + | MA_LOG_LEVEL_INFO | + | MA_LOG_LEVEL_WARNING | + | MA_LOG_LEVEL_ERROR | + |----------------------| + +message (in) + The log message. + + +Remarks +------- +Do not modify the state of the device from inside the callback. +*/ +typedef void (* ma_log_proc)(ma_context* pContext, ma_device* pDevice, ma_uint32 logLevel, const char* message); + +typedef enum +{ + ma_device_type_playback = 1, + ma_device_type_capture = 2, + ma_device_type_duplex = ma_device_type_playback | ma_device_type_capture, /* 3 */ + ma_device_type_loopback = 4 +} ma_device_type; + +typedef enum +{ + ma_share_mode_shared = 0, + ma_share_mode_exclusive +} ma_share_mode; + +/* iOS/tvOS/watchOS session categories. */ +typedef enum +{ + ma_ios_session_category_default = 0, /* AVAudioSessionCategoryPlayAndRecord with AVAudioSessionCategoryOptionDefaultToSpeaker. */ + ma_ios_session_category_none, /* Leave the session category unchanged. */ + ma_ios_session_category_ambient, /* AVAudioSessionCategoryAmbient */ + ma_ios_session_category_solo_ambient, /* AVAudioSessionCategorySoloAmbient */ + ma_ios_session_category_playback, /* AVAudioSessionCategoryPlayback */ + ma_ios_session_category_record, /* AVAudioSessionCategoryRecord */ + ma_ios_session_category_play_and_record, /* AVAudioSessionCategoryPlayAndRecord */ + ma_ios_session_category_multi_route /* AVAudioSessionCategoryMultiRoute */ +} ma_ios_session_category; + +/* iOS/tvOS/watchOS session category options */ +typedef enum +{ + ma_ios_session_category_option_mix_with_others = 0x01, /* AVAudioSessionCategoryOptionMixWithOthers */ + ma_ios_session_category_option_duck_others = 0x02, /* AVAudioSessionCategoryOptionDuckOthers */ + ma_ios_session_category_option_allow_bluetooth = 0x04, /* AVAudioSessionCategoryOptionAllowBluetooth */ + ma_ios_session_category_option_default_to_speaker = 0x08, /* AVAudioSessionCategoryOptionDefaultToSpeaker */ + ma_ios_session_category_option_interrupt_spoken_audio_and_mix_with_others = 0x11, /* AVAudioSessionCategoryOptionInterruptSpokenAudioAndMixWithOthers */ + ma_ios_session_category_option_allow_bluetooth_a2dp = 0x20, /* AVAudioSessionCategoryOptionAllowBluetoothA2DP */ + ma_ios_session_category_option_allow_air_play = 0x40, /* AVAudioSessionCategoryOptionAllowAirPlay */ +} ma_ios_session_category_option; + +typedef union +{ + ma_int64 counter; + double counterD; +} ma_timer; + +typedef union +{ + wchar_t wasapi[64]; /* WASAPI uses a wchar_t string for identification. */ + ma_uint8 dsound[16]; /* DirectSound uses a GUID for identification. */ + /*UINT_PTR*/ ma_uint32 winmm; /* When creating a device, WinMM expects a Win32 UINT_PTR for device identification. In practice it's actually just a UINT. */ + char alsa[256]; /* ALSA uses a name string for identification. */ + char pulse[256]; /* PulseAudio uses a name string for identification. */ + int jack; /* JACK always uses default devices. */ + char coreaudio[256]; /* Core Audio uses a string for identification. */ + char sndio[256]; /* "snd/0", etc. */ + char audio4[256]; /* "/dev/audio", etc. */ + char oss[64]; /* "dev/dsp0", etc. "dev/dsp" for the default device. */ + ma_int32 aaudio; /* AAudio uses a 32-bit integer for identification. */ + ma_uint32 opensl; /* OpenSL|ES uses a 32-bit unsigned integer for identification. */ + char webaudio[32]; /* Web Audio always uses default devices for now, but if this changes it'll be a GUID. */ + int nullbackend; /* The null backend uses an integer for device IDs. */ +} ma_device_id; + +typedef struct +{ + /* Basic info. This is the only information guaranteed to be filled in during device enumeration. */ + ma_device_id id; + char name[256]; + + /* + Detailed info. As much of this is filled as possible with ma_context_get_device_info(). Note that you are allowed to initialize + a device with settings outside of this range, but it just means the data will be converted using miniaudio's data conversion + pipeline before sending the data to/from the device. Most programs will need to not worry about these values, but it's provided + here mainly for informational purposes or in the rare case that someone might find it useful. + + These will be set to 0 when returned by ma_context_enumerate_devices() or ma_context_get_devices(). + */ + ma_uint32 formatCount; + ma_format formats[ma_format_count]; + ma_uint32 minChannels; + ma_uint32 maxChannels; + ma_uint32 minSampleRate; + ma_uint32 maxSampleRate; + + struct + { + ma_bool32 isDefault; + } _private; +} ma_device_info; + +typedef struct +{ + ma_device_type deviceType; + ma_uint32 sampleRate; + ma_uint32 periodSizeInFrames; + ma_uint32 periodSizeInMilliseconds; + ma_uint32 periods; + ma_performance_profile performanceProfile; + ma_bool32 noPreZeroedOutputBuffer; /* When set to true, the contents of the output buffer passed into the data callback will be left undefined rather than initialized to zero. */ + ma_bool32 noClip; /* When set to true, the contents of the output buffer passed into the data callback will be clipped after returning. Only applies when the playback sample format is f32. */ + ma_device_callback_proc dataCallback; + ma_stop_proc stopCallback; + void* pUserData; + struct + { + ma_resample_algorithm algorithm; + struct + { + ma_uint32 lpfCount; + } linear; + struct + { + int quality; + } speex; + } resampling; + struct + { + ma_device_id* pDeviceID; + ma_format format; + ma_uint32 channels; + ma_channel channelMap[MA_MAX_CHANNELS]; + ma_share_mode shareMode; + } playback; + struct + { + ma_device_id* pDeviceID; + ma_format format; + ma_uint32 channels; + ma_channel channelMap[MA_MAX_CHANNELS]; + ma_share_mode shareMode; + } capture; + + struct + { + ma_bool32 noAutoConvertSRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM. */ + ma_bool32 noDefaultQualitySRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY. */ + ma_bool32 noAutoStreamRouting; /* Disables automatic stream routing. */ + ma_bool32 noHardwareOffloading; /* Disables WASAPI's hardware offloading feature. */ + } wasapi; + struct + { + ma_bool32 noMMap; /* Disables MMap mode. */ + } alsa; + struct + { + const char* pStreamNamePlayback; + const char* pStreamNameCapture; + } pulse; +} ma_device_config; + +typedef struct +{ + ma_log_proc logCallback; + ma_thread_priority threadPriority; + void* pUserData; + ma_allocation_callbacks allocationCallbacks; + struct + { + ma_bool32 useVerboseDeviceEnumeration; + } alsa; + struct + { + const char* pApplicationName; + const char* pServerName; + ma_bool32 tryAutoSpawn; /* Enables autospawning of the PulseAudio daemon if necessary. */ + } pulse; + struct + { + ma_ios_session_category sessionCategory; + ma_uint32 sessionCategoryOptions; + } coreaudio; + struct + { + const char* pClientName; + ma_bool32 tryStartServer; + } jack; +} ma_context_config; + +/* +The callback for handling device enumeration. This is fired from `ma_context_enumerated_devices()`. + + +Parameters +---------- +pContext (in) + A pointer to the context performing the enumeration. + +deviceType (in) + The type of the device being enumerated. This will always be either `ma_device_type_playback` or `ma_device_type_capture`. + +pInfo (in) + A pointer to a `ma_device_info` containing the ID and name of the enumerated device. Note that this will not include detailed information about the device, + only basic information (ID and name). The reason for this is that it would otherwise require opening the backend device to probe for the information which + is too inefficient. + +pUserData (in) + The user data pointer passed into `ma_context_enumerate_devices()`. +*/ +typedef ma_bool32 (* ma_enum_devices_callback_proc)(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pInfo, void* pUserData); + +struct ma_context +{ + ma_backend backend; /* DirectSound, ALSA, etc. */ + ma_log_proc logCallback; + ma_thread_priority threadPriority; + void* pUserData; + ma_allocation_callbacks allocationCallbacks; + ma_mutex deviceEnumLock; /* Used to make ma_context_get_devices() thread safe. */ + ma_mutex deviceInfoLock; /* Used to make ma_context_get_device_info() thread safe. */ + ma_uint32 deviceInfoCapacity; /* Total capacity of pDeviceInfos. */ + ma_uint32 playbackDeviceInfoCount; + ma_uint32 captureDeviceInfoCount; + ma_device_info* pDeviceInfos; /* Playback devices first, then capture. */ + ma_bool32 isBackendAsynchronous : 1; /* Set when the context is initialized. Set to 1 for asynchronous backends such as Core Audio and JACK. Do not modify. */ + + ma_result (* onUninit )(ma_context* pContext); + ma_bool32 (* onDeviceIDEqual )(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1); + ma_result (* onEnumDevices )(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData); /* Return false from the callback to stop enumeration. */ + ma_result (* onGetDeviceInfo )(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo); + ma_result (* onDeviceInit )(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice); + void (* onDeviceUninit )(ma_device* pDevice); + ma_result (* onDeviceStart )(ma_device* pDevice); + ma_result (* onDeviceStop )(ma_device* pDevice); + ma_result (* onDeviceMainLoop)(ma_device* pDevice); + + union + { +#ifdef MA_SUPPORT_WASAPI + struct + { + int _unused; + } wasapi; +#endif +#ifdef MA_SUPPORT_DSOUND + struct + { + ma_handle hDSoundDLL; + ma_proc DirectSoundCreate; + ma_proc DirectSoundEnumerateA; + ma_proc DirectSoundCaptureCreate; + ma_proc DirectSoundCaptureEnumerateA; + } dsound; +#endif +#ifdef MA_SUPPORT_WINMM + struct + { + ma_handle hWinMM; + ma_proc waveOutGetNumDevs; + ma_proc waveOutGetDevCapsA; + ma_proc waveOutOpen; + ma_proc waveOutClose; + ma_proc waveOutPrepareHeader; + ma_proc waveOutUnprepareHeader; + ma_proc waveOutWrite; + ma_proc waveOutReset; + ma_proc waveInGetNumDevs; + ma_proc waveInGetDevCapsA; + ma_proc waveInOpen; + ma_proc waveInClose; + ma_proc waveInPrepareHeader; + ma_proc waveInUnprepareHeader; + ma_proc waveInAddBuffer; + ma_proc waveInStart; + ma_proc waveInReset; + } winmm; +#endif +#ifdef MA_SUPPORT_ALSA + struct + { + ma_handle asoundSO; + ma_proc snd_pcm_open; + ma_proc snd_pcm_close; + ma_proc snd_pcm_hw_params_sizeof; + ma_proc snd_pcm_hw_params_any; + ma_proc snd_pcm_hw_params_set_format; + ma_proc snd_pcm_hw_params_set_format_first; + ma_proc snd_pcm_hw_params_get_format_mask; + ma_proc snd_pcm_hw_params_set_channels_near; + ma_proc snd_pcm_hw_params_set_rate_resample; + ma_proc snd_pcm_hw_params_set_rate_near; + ma_proc snd_pcm_hw_params_set_buffer_size_near; + ma_proc snd_pcm_hw_params_set_periods_near; + ma_proc snd_pcm_hw_params_set_access; + ma_proc snd_pcm_hw_params_get_format; ma_proc snd_pcm_hw_params_get_channels; ma_proc snd_pcm_hw_params_get_channels_min; ma_proc snd_pcm_hw_params_get_channels_max; @@ -2290,6 +3002,7 @@ struct ma_context ma_proc AAudioStreamBuilder_setBufferCapacityInFrames; ma_proc AAudioStreamBuilder_setFramesPerDataCallback; ma_proc AAudioStreamBuilder_setDataCallback; + ma_proc AAudioStreamBuilder_setErrorCallback; ma_proc AAudioStreamBuilder_setPerformanceMode; ma_proc AAudioStreamBuilder_openStream; ma_proc AAudioStream_close; @@ -2373,14 +3086,14 @@ struct ma_context }; }; -MA_ALIGNED_STRUCT(MA_SIMD_ALIGNMENT) ma_device +struct ma_device { ma_context* pContext; ma_device_type type; ma_uint32 sampleRate; - ma_uint32 state; - ma_device_callback_proc onData; - ma_stop_proc onStop; + volatile ma_uint32 state; /* The state of the device is variable and can change at any time on any thread, so tell the compiler as such with `volatile`. */ + ma_device_callback_proc onData; /* Set once at initialization time and should not be changed after. */ + ma_stop_proc onStop; /* Set once at initialization time and should not be changed after. */ void* pUserData; /* Application defined data. */ ma_mutex lock; ma_event wakeupEvent; @@ -2394,7 +3107,19 @@ MA_ALIGNED_STRUCT(MA_SIMD_ALIGNMENT) ma_device ma_bool32 isOwnerOfContext : 1; /* When set to true, uninitializing the device will also uninitialize the context. Set to true when NULL is passed into ma_device_init(). */ ma_bool32 noPreZeroedOutputBuffer : 1; ma_bool32 noClip : 1; - float masterVolumeFactor; + volatile float masterVolumeFactor; /* Volatile so we can use some thread safety when applying volume to periods. */ + struct + { + ma_resample_algorithm algorithm; + struct + { + ma_uint32 lpfCount; + } linear; + struct + { + int quality; + } speex; + } resampling; struct { char name[256]; /* Maybe temporary. Likely to be replaced with a query API. */ @@ -2409,11 +3134,9 @@ MA_ALIGNED_STRUCT(MA_SIMD_ALIGNMENT) ma_device ma_uint32 internalChannels; ma_uint32 internalSampleRate; ma_channel internalChannelMap[MA_MAX_CHANNELS]; - ma_uint32 internalBufferSizeInFrames; + ma_uint32 internalPeriodSizeInFrames; ma_uint32 internalPeriods; - ma_pcm_converter converter; - ma_uint32 _dspFrameCount; /* Internal use only. Used as the data source when reading from the device. */ - const ma_uint8* _dspFrames; /* ^^^ AS ABOVE ^^^ */ + ma_data_converter converter; } playback; struct { @@ -2429,11 +3152,9 @@ MA_ALIGNED_STRUCT(MA_SIMD_ALIGNMENT) ma_device ma_uint32 internalChannels; ma_uint32 internalSampleRate; ma_channel internalChannelMap[MA_MAX_CHANNELS]; - ma_uint32 internalBufferSizeInFrames; + ma_uint32 internalPeriodSizeInFrames; ma_uint32 internalPeriods; - ma_pcm_converter converter; - ma_uint32 _dspFrameCount; /* Internal use only. Used as the data source when reading from the device. */ - const ma_uint8* _dspFrames; /* ^^^ AS ABOVE ^^^ */ + ma_data_converter converter; } capture; union @@ -2449,19 +3170,22 @@ MA_ALIGNED_STRUCT(MA_SIMD_ALIGNMENT) ma_device ma_IMMNotificationClient notificationClient; /*HANDLE*/ ma_handle hEventPlayback; /* Auto reset. Initialized to signaled. */ /*HANDLE*/ ma_handle hEventCapture; /* Auto reset. Initialized to unsignaled. */ - ma_uint32 actualBufferSizeInFramesPlayback; /* Value from GetBufferSize(). internalBufferSizeInFrames is not set to the _actual_ buffer size when low-latency shared mode is being used due to the way the IAudioClient3 API works. */ - ma_uint32 actualBufferSizeInFramesCapture; - ma_uint32 originalBufferSizeInFrames; - ma_uint32 originalBufferSizeInMilliseconds; + ma_uint32 actualPeriodSizeInFramesPlayback; /* Value from GetBufferSize(). internalPeriodSizeInFrames is not set to the _actual_ buffer size when low-latency shared mode is being used due to the way the IAudioClient3 API works. */ + ma_uint32 actualPeriodSizeInFramesCapture; + ma_uint32 originalPeriodSizeInFrames; + ma_uint32 originalPeriodSizeInMilliseconds; ma_uint32 originalPeriods; ma_bool32 hasDefaultPlaybackDeviceChanged; /* <-- Make sure this is always a whole 32-bits because we use atomic assignments. */ ma_bool32 hasDefaultCaptureDeviceChanged; /* <-- Make sure this is always a whole 32-bits because we use atomic assignments. */ ma_uint32 periodSizeInFramesPlayback; ma_uint32 periodSizeInFramesCapture; - ma_bool32 isStartedCapture; - ma_bool32 isStartedPlayback; - ma_bool32 noAutoConvertSRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM. */ - ma_bool32 noDefaultQualitySRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY. */ + ma_bool32 isStartedCapture; /* <-- Make sure this is always a whole 32-bits because we use atomic assignments. */ + ma_bool32 isStartedPlayback; /* <-- Make sure this is always a whole 32-bits because we use atomic assignments. */ + ma_bool32 noAutoConvertSRC : 1; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM. */ + ma_bool32 noDefaultQualitySRC : 1; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY. */ + ma_bool32 noHardwareOffloading : 1; + ma_bool32 allowCaptureAutoStreamRouting : 1; + ma_bool32 allowPlaybackAutoStreamRouting : 1; } wasapi; #endif #ifdef MA_SUPPORT_DSOUND @@ -2541,14 +3265,15 @@ MA_ALIGNED_STRUCT(MA_SIMD_ALIGNMENT) ma_device /*AudioUnit*/ ma_ptr audioUnitCapture; /*AudioBufferList**/ ma_ptr pAudioBufferList; /* Only used for input devices. */ ma_event stopEvent; - ma_uint32 originalBufferSizeInFrames; - ma_uint32 originalBufferSizeInMilliseconds; + ma_uint32 originalPeriodSizeInFrames; + ma_uint32 originalPeriodSizeInMilliseconds; ma_uint32 originalPeriods; ma_bool32 isDefaultPlaybackDevice; ma_bool32 isDefaultCaptureDevice; ma_bool32 isSwitchingPlaybackDevice; /* <-- Set to true when the default device has changed and miniaudio is in the process of switching. */ ma_bool32 isSwitchingCaptureDevice; /* <-- Set to true when the default device has changed and miniaudio is in the process of switching. */ ma_pcm_rb duplexRB; + void* pRouteChangeHandler; /* Only used on mobile platforms. Obj-C object for handling route changes. */ } coreaudio; #endif #ifdef MA_SUPPORT_SNDIO @@ -2560,8643 +3285,8088 @@ MA_ALIGNED_STRUCT(MA_SIMD_ALIGNMENT) ma_device ma_bool32 isStartedCapture; } sndio; #endif -#ifdef MA_SUPPORT_AUDIO4 - struct - { - int fdPlayback; - int fdCapture; - } audio4; -#endif -#ifdef MA_SUPPORT_OSS - struct - { - int fdPlayback; - int fdCapture; - } oss; -#endif -#ifdef MA_SUPPORT_AAUDIO - struct - { - /*AAudioStream**/ ma_ptr pStreamPlayback; - /*AAudioStream**/ ma_ptr pStreamCapture; - ma_pcm_rb duplexRB; - } aaudio; -#endif -#ifdef MA_SUPPORT_OPENSL - struct - { - /*SLObjectItf*/ ma_ptr pOutputMixObj; - /*SLOutputMixItf*/ ma_ptr pOutputMix; - /*SLObjectItf*/ ma_ptr pAudioPlayerObj; - /*SLPlayItf*/ ma_ptr pAudioPlayer; - /*SLObjectItf*/ ma_ptr pAudioRecorderObj; - /*SLRecordItf*/ ma_ptr pAudioRecorder; - /*SLAndroidSimpleBufferQueueItf*/ ma_ptr pBufferQueuePlayback; - /*SLAndroidSimpleBufferQueueItf*/ ma_ptr pBufferQueueCapture; - ma_uint32 currentBufferIndexPlayback; - ma_uint32 currentBufferIndexCapture; - ma_uint8* pBufferPlayback; /* This is malloc()'d and is used for storing audio data. Typed as ma_uint8 for easy offsetting. */ - ma_uint8* pBufferCapture; - ma_pcm_rb duplexRB; - } opensl; -#endif -#ifdef MA_SUPPORT_WEBAUDIO - struct - { - int indexPlayback; /* We use a factory on the JavaScript side to manage devices and use an index for JS/C interop. */ - int indexCapture; - ma_pcm_rb duplexRB; /* In external capture format. */ - } webaudio; -#endif -#ifdef MA_SUPPORT_NULL - struct - { - ma_thread deviceThread; - ma_event operationEvent; - ma_event operationCompletionEvent; - ma_uint32 operation; - ma_result operationResult; - ma_timer timer; - double priorRunTime; - ma_uint32 currentPeriodFramesRemainingPlayback; - ma_uint32 currentPeriodFramesRemainingCapture; - ma_uint64 lastProcessedFramePlayback; - ma_uint32 lastProcessedFrameCapture; - ma_bool32 isStarted; - } null_device; -#endif - }; -}; -#if defined(_MSC_VER) && !defined(__clang__) - #pragma warning(pop) -#else - #pragma GCC diagnostic pop /* For ISO C99 doesn't support unnamed structs/unions [-Wpedantic] */ -#endif - -/* -Initializes a context. - -The context is used for selecting and initializing the relevant backends. - -Note that the location of the context cannot change throughout it's lifetime. Consider allocating -the ma_context object with malloc() if this is an issue. The reason for this is that a pointer -to the context is stored in the ma_device structure. - - is used to allow the application to prioritize backends depending on it's specific -requirements. This can be null in which case it uses the default priority, which is as follows: - - WASAPI - - DirectSound - - WinMM - - Core Audio (Apple) - - sndio - - audio(4) - - OSS - - PulseAudio - - ALSA - - JACK - - AAudio - - OpenSL|ES - - Web Audio / Emscripten - - Null - - is used to configure the context. Use the logCallback config to set a callback for whenever a -log message is posted. The priority of the worker thread can be set with the threadPriority config. - -It is recommended that only a single context is active at any given time because it's a bulky data -structure which performs run-time linking for the relevant backends every time it's initialized. - -Return Value: - MA_SUCCESS if successful; any other error code otherwise. - -Thread Safety: UNSAFE -*/ -ma_result ma_context_init(const ma_backend backends[], ma_uint32 backendCount, const ma_context_config* pConfig, ma_context* pContext); - -/* -Uninitializes a context. - -Results are undefined if you call this while any device created by this context is still active. - -Return Value: - MA_SUCCESS if successful; any other error code otherwise. - -Thread Safety: UNSAFE -*/ -ma_result ma_context_uninit(ma_context* pContext); - -/* -Enumerates over every device (both playback and capture). - -This is a lower-level enumeration function to the easier to use ma_context_get_devices(). Use -ma_context_enumerate_devices() if you would rather not incur an internal heap allocation, or -it simply suits your code better. - -Do _not_ assume the first enumerated device of a given type is the default device. - -Some backends and platforms may only support default playback and capture devices. - -Note that this only retrieves the ID and name/description of the device. The reason for only -retrieving basic information is that it would otherwise require opening the backend device in -order to probe it for more detailed information which can be inefficient. Consider using -ma_context_get_device_info() for this, but don't call it from within the enumeration callback. - -In general, you should not do anything complicated from within the callback. In particular, do -not try initializing a device from within the callback. - -Consider using ma_context_get_devices() for a simpler and safer API, albeit at the expense of -an internal heap allocation. - -Returning false from the callback will stop enumeration. Returning true will continue enumeration. - -Return Value: - MA_SUCCESS if successful; any other error code otherwise. - -Thread Safety: SAFE - This is guarded using a simple mutex lock. -*/ -ma_result ma_context_enumerate_devices(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData); - -/* -Retrieves basic information about every active playback and/or capture device. - -You can pass in NULL for the playback or capture lists in which case they'll be ignored. - -It is _not_ safe to assume the first device in the list is the default device. - -The returned pointers will become invalid upon the next call this this function, or when the -context is uninitialized. Do not free the returned pointers. - -This function follows the same enumeration rules as ma_context_enumerate_devices(). See -documentation for ma_context_enumerate_devices() for more information. - -Return Value: - MA_SUCCESS if successful; any other error code otherwise. - -Thread Safety: SAFE - Since each call to this function invalidates the pointers from the previous call, you - should not be calling this simultaneously across multiple threads. Instead, you need to - make a copy of the returned data with your own higher level synchronization. -*/ -ma_result ma_context_get_devices(ma_context* pContext, ma_device_info** ppPlaybackDeviceInfos, ma_uint32* pPlaybackDeviceCount, ma_device_info** ppCaptureDeviceInfos, ma_uint32* pCaptureDeviceCount); - -/* -Retrieves information about a device with the given ID. - -Do _not_ call this from within the ma_context_enumerate_devices() callback. - -It's possible for a device to have different information and capabilities depending on whether -or not it's opened in shared or exclusive mode. For example, in shared mode, WASAPI always uses -floating point samples for mixing, but in exclusive mode it can be anything. Therefore, this -function allows you to specify which share mode you want information for. Note that not all -backends and devices support shared or exclusive mode, in which case this function will fail -if the requested share mode is unsupported. - -This leaves pDeviceInfo unmodified in the result of an error. - -Return Value: - MA_SUCCESS if successful; any other error code otherwise. - -Thread Safety: SAFE - This is guarded using a simple mutex lock. -*/ -ma_result ma_context_get_device_info(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo); - -/* -Determines if the given context supports loopback mode. -*/ -ma_bool32 ma_context_is_loopback_supported(ma_context* pContext); - -/* -Initializes a device. - -The context can be null in which case it uses the default. This is equivalent to passing in a -context that was initialized like so: - - ma_context_init(NULL, 0, NULL, &context); - -Do not pass in null for the context if you are needing to open multiple devices. You can, -however, use null when initializing the first device, and then use device.pContext for the -initialization of other devices. - -The device's configuration is controlled with pConfig. This allows you to configure the sample -format, channel count, sample rate, etc. Before calling ma_device_init(), you will need to -initialize a ma_device_config object using ma_device_config_init(). You must set the callback in -the device config. Once initialized, the device's config is immutable. If you need to change the -config you will need to initialize a new device. - -Passing in 0 to any property in pConfig will force the use of a default value. In the case of -sample format, channel count, sample rate and channel map it will default to the values used by -the backend's internal device. For the size of the buffer you can set bufferSizeInFrames or -bufferSizeInMilliseconds (if both are set it will prioritize bufferSizeInFrames). If both are -set to zero, it will default to MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_LOW_LATENCY or -MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_CONSERVATIVE, depending on whether or not performanceProfile -is set to ma_performance_profile_low_latency or ma_performance_profile_conservative. - -If you request exclusive mode and the backend does not support it an error will be returned. For -robustness, you may want to first try initializing the device in exclusive mode, and then fall back -to shared mode if required. Alternatively you can just request shared mode (the default if you -leave it unset in the config) which is the most reliable option. Some backends do not have a -practical way of choosing whether or not the device should be exclusive or not (ALSA, for example) -in which case it just acts as a hint. Unless you have special requirements you should try avoiding -exclusive mode as it's intrusive to the user. Starting with Windows 10, miniaudio will use low-latency -shared mode where possible which may make exclusive mode unnecessary. - -When sending or receiving data to/from a device, miniaudio will internally perform a format -conversion to convert between the format specified by pConfig and the format used internally by -the backend. If you pass in NULL for pConfig or 0 for the sample format, channel count, -sample rate _and_ channel map, data transmission will run on an optimized pass-through fast path. - -The buffer size should be treated as a hint. miniaudio will try it's best to use exactly what you -ask for, but it may differ. You should not assume the number of frames specified in each call to -the data callback is exactly what you originally specified. - -The property controls how frequently the background thread is woken to check for more -data. It's tied to the buffer size, so as an example, if your buffer size is equivalent to 10 -milliseconds and you have 2 periods, the CPU will wake up approximately every 5 milliseconds. - -When compiling for UWP you must ensure you call this function on the main UI thread because the -operating system may need to present the user with a message asking for permissions. Please refer -to the official documentation for ActivateAudioInterfaceAsync() for more information. - -ALSA Specific: When initializing the default device, requesting shared mode will try using the -"dmix" device for playback and the "dsnoop" device for capture. If these fail it will try falling -back to the "hw" device. - -Return Value: - MA_SUCCESS if successful; any other error code otherwise. - -Thread Safety: UNSAFE - It is not safe to call this function simultaneously for different devices because some backends - depend on and mutate global state (such as OpenSL|ES). The same applies to calling this at the - same time as ma_device_uninit(). -*/ -ma_result ma_device_init(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice); - -/* -Initializes a device without a context, with extra parameters for controlling the configuration -of the internal self-managed context. - -See ma_device_init() and ma_context_init(). -*/ -ma_result ma_device_init_ex(const ma_backend backends[], ma_uint32 backendCount, const ma_context_config* pContextConfig, const ma_device_config* pConfig, ma_device* pDevice); - -/* -Uninitializes a device. - -This will explicitly stop the device. You do not need to call ma_device_stop() beforehand, but it's -harmless if you do. - -Do not call this in any callback. - -Return Value: - MA_SUCCESS if successful; any other error code otherwise. - -Thread Safety: UNSAFE - As soon as this API is called the device should be considered undefined. All bets are off if you - try using the device at the same time as uninitializing it. -*/ -void ma_device_uninit(ma_device* pDevice); - -/* -Sets the callback to use when the device has stopped, either explicitly or as a result of an error. - -Thread Safety: SAFE - This API is implemented as a simple atomic assignment. -*/ -void ma_device_set_stop_callback(ma_device* pDevice, ma_stop_proc proc); +#ifdef MA_SUPPORT_AUDIO4 + struct + { + int fdPlayback; + int fdCapture; + } audio4; +#endif +#ifdef MA_SUPPORT_OSS + struct + { + int fdPlayback; + int fdCapture; + } oss; +#endif +#ifdef MA_SUPPORT_AAUDIO + struct + { + /*AAudioStream**/ ma_ptr pStreamPlayback; + /*AAudioStream**/ ma_ptr pStreamCapture; + ma_pcm_rb duplexRB; + } aaudio; +#endif +#ifdef MA_SUPPORT_OPENSL + struct + { + /*SLObjectItf*/ ma_ptr pOutputMixObj; + /*SLOutputMixItf*/ ma_ptr pOutputMix; + /*SLObjectItf*/ ma_ptr pAudioPlayerObj; + /*SLPlayItf*/ ma_ptr pAudioPlayer; + /*SLObjectItf*/ ma_ptr pAudioRecorderObj; + /*SLRecordItf*/ ma_ptr pAudioRecorder; + /*SLAndroidSimpleBufferQueueItf*/ ma_ptr pBufferQueuePlayback; + /*SLAndroidSimpleBufferQueueItf*/ ma_ptr pBufferQueueCapture; + ma_bool32 isDrainingCapture; + ma_bool32 isDrainingPlayback; + ma_uint32 currentBufferIndexPlayback; + ma_uint32 currentBufferIndexCapture; + ma_uint8* pBufferPlayback; /* This is malloc()'d and is used for storing audio data. Typed as ma_uint8 for easy offsetting. */ + ma_uint8* pBufferCapture; + ma_pcm_rb duplexRB; + } opensl; +#endif +#ifdef MA_SUPPORT_WEBAUDIO + struct + { + int indexPlayback; /* We use a factory on the JavaScript side to manage devices and use an index for JS/C interop. */ + int indexCapture; + ma_pcm_rb duplexRB; /* In external capture format. */ + } webaudio; +#endif +#ifdef MA_SUPPORT_NULL + struct + { + ma_thread deviceThread; + ma_event operationEvent; + ma_event operationCompletionEvent; + ma_uint32 operation; + ma_result operationResult; + ma_timer timer; + double priorRunTime; + ma_uint32 currentPeriodFramesRemainingPlayback; + ma_uint32 currentPeriodFramesRemainingCapture; + ma_uint64 lastProcessedFramePlayback; + ma_uint32 lastProcessedFrameCapture; + ma_bool32 isStarted; + } null_device; +#endif + }; +}; +#if defined(_MSC_VER) && !defined(__clang__) + #pragma warning(pop) +#else + #pragma GCC diagnostic pop /* For ISO C99 doesn't support unnamed structs/unions [-Wpedantic] */ +#endif /* -Activates the device. For playback devices this begins playback. For capture devices it begins -recording. +Initializes a `ma_context_config` object. -For a playback device, this will retrieve an initial chunk of audio data from the client before -returning. The reason for this is to ensure there is valid audio data in the buffer, which needs -to be done _before_ the device begins playback. - -This API waits until the backend device has been started for real by the worker thread. It also -waits on a mutex for thread-safety. - -Do not call this in any callback. - -Return Value: - MA_SUCCESS if successful; any other error code otherwise. - -Thread Safety: SAFE -*/ -ma_result ma_device_start(ma_device* pDevice); -/* -Puts the device to sleep, but does not uninitialize it. Use ma_device_start() to start it up again. +Return Value +------------ +A `ma_context_config` initialized to defaults. -This API needs to wait on the worker thread to stop the backend device properly before returning. It -also waits on a mutex for thread-safety. In addition, some backends need to wait for the device to -finish playback/recording of the current fragment which can take some time (usually proportionate to -the buffer size that was specified at initialization time). -This should not drop unprocessed samples. Backends are required to either pause the stream in-place -or drain the buffer if pausing is not possible. The reason for this is that stopping the device and -the resuming it with ma_device_start() (which you might do when your program loses focus) may result -in a situation where those samples are never output to the speakers or received from the microphone -which can in turn result in de-syncs. +Remarks +------- +You must always use this to initialize the default state of the `ma_context_config` object. Not using this will result in your program breaking when miniaudio +is updated and new members are added to `ma_context_config`. It also sets logical defaults. -Do not call this in any callback. +You can override members of the returned object by changing it's members directly. -Return Value: - MA_SUCCESS if successful; any other error code otherwise. -Thread Safety: SAFE +See Also +-------- +ma_context_init() */ -ma_result ma_device_stop(ma_device* pDevice); +ma_context_config ma_context_config_init(void); /* -Determines whether or not the device is started. - -This is implemented as a simple accessor. - -Return Value: - True if the device is started, false otherwise. +Initializes a context. -Thread Safety: SAFE - If another thread calls ma_device_start() or ma_device_stop() at this same time as this function - is called, there's a very small chance the return value will be out of sync. -*/ -ma_bool32 ma_device_is_started(ma_device* pDevice); +The context is used for selecting and initializing an appropriate backend and to represent the backend at a more global level than that of an individual +device. There is one context to many devices, and a device is created from a context. A context is required to enumerate devices. -/* -Sets the master volume factor for the device. -The volume factor must be between 0 (silence) and 1 (full volume). Use ma_device_set_master_gain_db() to -use decibel notation, where 0 is full volume. +Parameters +---------- +backends (in, optional) + A list of backends to try initializing, in priority order. Can be NULL, in which case it uses default priority order. -This applies the volume factor across all channels. +backendCount (in, optional) + The number of items in `backend`. Ignored if `backend` is NULL. -This does not change the operating system's volume. It only affects the volume for the given ma_device -object's audio stream. +pConfig (in, optional) + The context configuration. -Return Value ------------- -MA_SUCCESS if the volume was set successfully. -MA_INVALID_ARGS if pDevice is NULL. -MA_INVALID_ARGS if the volume factor is not within the range of [0, 1]. -*/ -ma_result ma_device_set_master_volume(ma_device* pDevice, float volume); +pContext (in) + A pointer to the context object being initialized. -/* -Retrieves the master volume factor for the device. Return Value ------------ -MA_SUCCESS if successful. -MA_INVALID_ARGS if pDevice is NULL. -MA_INVALID_ARGS if pVolume is NULL. -*/ -ma_result ma_device_get_master_volume(ma_device* pDevice, float* pVolume); - -/* -Sets the master volume for the device as gain in decibels. +MA_SUCCESS if successful; any other error code otherwise. -A gain of 0 is full volume, whereas a gain of < 0 will decrease the volume. -This applies the gain across all channels. +Thread Safety +------------- +Unsafe. Do not call this function across multiple threads as some backends read and write to global state. -This does not change the operating system's volume. It only affects the volume for the given ma_device -object's audio stream. -Return Value ------------- -MA_SUCCESS if the volume was set successfully. -MA_INVALID_ARGS if pDevice is NULL. -MA_INVALID_ARGS if the gain is > 0. -*/ -ma_result ma_device_set_master_gain_db(ma_device* pDevice, float gainDB); +Remarks +------- +When `backends` is NULL, the default priority order will be used. Below is a list of backends in priority order: + + |-------------|-----------------------|--------------------------------------------------------| + | Name | Enum Name | Supported Operating Systems | + |-------------|-----------------------|--------------------------------------------------------| + | WASAPI | ma_backend_wasapi | Windows Vista+ | + | DirectSound | ma_backend_dsound | Windows XP+ | + | WinMM | ma_backend_winmm | Windows XP+ (may work on older versions, but untested) | + | Core Audio | ma_backend_coreaudio | macOS, iOS | + | ALSA | ma_backend_alsa | Linux | + | PulseAudio | ma_backend_pulseaudio | Cross Platform (disabled on Windows, BSD and Android) | + | JACK | ma_backend_jack | Cross Platform (disabled on BSD and Android) | + | sndio | ma_backend_sndio | OpenBSD | + | audio(4) | ma_backend_audio4 | NetBSD, OpenBSD | + | OSS | ma_backend_oss | FreeBSD | + | AAudio | ma_backend_aaudio | Android 8+ | + | OpenSL|ES | ma_backend_opensl | Android (API level 16+) | + | Web Audio | ma_backend_webaudio | Web (via Emscripten) | + | Null | ma_backend_null | Cross Platform (not used on Web) | + |-------------|-----------------------|--------------------------------------------------------| + +The context can be configured via the `pConfig` argument. The config object is initialized with `ma_context_config_init()`. Individual configuration settings +can then be set directly on the structure. Below are the members of the `ma_context_config` object. + + logCallback + Callback for handling log messages from miniaudio. + + threadPriority + The desired priority to use for the audio thread. Allowable values include the following: + + |--------------------------------------| + | Thread Priority | + |--------------------------------------| + | ma_thread_priority_idle | + | ma_thread_priority_lowest | + | ma_thread_priority_low | + | ma_thread_priority_normal | + | ma_thread_priority_high | + | ma_thread_priority_highest (default) | + | ma_thread_priority_realtime | + | ma_thread_priority_default | + |--------------------------------------| + + pUserData + A pointer to application-defined data. This can be accessed from the context object directly such as `context.pUserData`. + + allocationCallbacks + Structure containing custom allocation callbacks. Leaving this at defaults will cause it to use MA_MALLOC, MA_REALLOC and MA_FREE. These allocation + callbacks will be used for anything tied to the context, including devices. + + alsa.useVerboseDeviceEnumeration + ALSA will typically enumerate many different devices which can be intrusive and unuser-friendly. To combat this, miniaudio will enumerate only unique + card/device pairs by default. The problem with this is that you lose a bit of flexibility and control. Setting alsa.useVerboseDeviceEnumeration makes + it so the ALSA backend includes all devices. Defaults to false. + + pulse.pApplicationName + PulseAudio only. The application name to use when initializing the PulseAudio context with `pa_context_new()`. + + pulse.pServerName + PulseAudio only. The name of the server to connect to with `pa_context_connect()`. + + pulse.tryAutoSpawn + PulseAudio only. Whether or not to try automatically starting the PulseAudio daemon. Defaults to false. If you set this to true, keep in mind that + miniaudio uses a trial and error method to find the most appropriate backend, and this will result in the PulseAudio daemon starting which may be + intrusive for the end user. + + coreaudio.sessionCategory + iOS only. The session category to use for the shared AudioSession instance. Below is a list of allowable values and their Core Audio equivalents. + + |-----------------------------------------|-------------------------------------| + | miniaudio Token | Core Audio Token | + |-----------------------------------------|-------------------------------------| + | ma_ios_session_category_ambient | AVAudioSessionCategoryAmbient | + | ma_ios_session_category_solo_ambient | AVAudioSessionCategorySoloAmbient | + | ma_ios_session_category_playback | AVAudioSessionCategoryPlayback | + | ma_ios_session_category_record | AVAudioSessionCategoryRecord | + | ma_ios_session_category_play_and_record | AVAudioSessionCategoryPlayAndRecord | + | ma_ios_session_category_multi_route | AVAudioSessionCategoryMultiRoute | + | ma_ios_session_category_none | AVAudioSessionCategoryAmbient | + | ma_ios_session_category_default | AVAudioSessionCategoryAmbient | + |-----------------------------------------|-------------------------------------| + + coreaudio.sessionCategoryOptions + iOS only. Session category options to use with the shared AudioSession instance. Below is a list of allowable values and their Core Audio equivalents. + + |---------------------------------------------------------------------------|------------------------------------------------------------------| + | miniaudio Token | Core Audio Token | + |---------------------------------------------------------------------------|------------------------------------------------------------------| + | ma_ios_session_category_option_mix_with_others | AVAudioSessionCategoryOptionMixWithOthers | + | ma_ios_session_category_option_duck_others | AVAudioSessionCategoryOptionDuckOthers | + | ma_ios_session_category_option_allow_bluetooth | AVAudioSessionCategoryOptionAllowBluetooth | + | ma_ios_session_category_option_default_to_speaker | AVAudioSessionCategoryOptionDefaultToSpeaker | + | ma_ios_session_category_option_interrupt_spoken_audio_and_mix_with_others | AVAudioSessionCategoryOptionInterruptSpokenAudioAndMixWithOthers | + | ma_ios_session_category_option_allow_bluetooth_a2dp | AVAudioSessionCategoryOptionAllowBluetoothA2DP | + | ma_ios_session_category_option_allow_air_play | AVAudioSessionCategoryOptionAllowAirPlay | + |---------------------------------------------------------------------------|------------------------------------------------------------------| + + jack.pClientName + The name of the client to pass to `jack_client_open()`. + + jack.tryStartServer + Whether or not to try auto-starting the JACK server. Defaults to false. + + +It is recommended that only a single context is active at any given time because it's a bulky data structure which performs run-time linking for the +relevant backends every time it's initialized. + +The location of the context cannot change throughout it's lifetime. Consider allocating the `ma_context` object with `malloc()` if this is an issue. The +reason for this is that a pointer to the context is stored in the `ma_device` structure. + + +Example 1 - Default Initialization +---------------------------------- +The example below shows how to initialize the context using the default configuration. + +```c +ma_context context; +ma_result result = ma_context_init(NULL, 0, NULL, &context); +if (result != MA_SUCCESS) { + // Error. +} +``` + + +Example 2 - Custom Configuration +-------------------------------- +The example below shows how to initialize the context using custom backend priorities and a custom configuration. In this hypothetical example, the program +wants to prioritize ALSA over PulseAudio on Linux. They also want to avoid using the WinMM backend on Windows because it's latency is too high. They also +want an error to be returned if no valid backend is available which they achieve by excluding the Null backend. + +For the configuration, the program wants to capture any log messages so they can, for example, route it to a log file and user interface. + +```c +ma_backend backends[] = { + ma_backend_alsa, + ma_backend_pulseaudio, + ma_backend_wasapi, + ma_backend_dsound +}; -/* -Retrieves the master gain in decibels. +ma_context_config config = ma_context_config_init(); +config.logCallback = my_log_callback; +config.pUserData = pMyUserData; -Return Value ------------- -MA_SUCCESS if successful. -MA_INVALID_ARGS if pDevice is NULL. -MA_INVALID_ARGS if pGainDB is NULL. -*/ -ma_result ma_device_get_master_gain_db(ma_device* pDevice, float* pGainDB); +ma_context context; +ma_result result = ma_context_init(backends, sizeof(backends)/sizeof(backends[0]), &config, &context); +if (result != MA_SUCCESS) { + // Error. + if (result == MA_NO_BACKEND) { + // Couldn't find an appropriate backend. + } +} +``` -/* -Helper function for initializing a ma_context_config object. +See Also +-------- +ma_context_config_init() +ma_context_uninit() */ -ma_context_config ma_context_config_init(void); +ma_result ma_context_init(const ma_backend backends[], ma_uint32 backendCount, const ma_context_config* pConfig, ma_context* pContext); /* -Initializes a device config. - -By default, the device config will use native device settings (format, channels, sample rate, etc.). Using native -settings means you will get an optimized pass-through data transmission pipeline to and from the device, but you will -need to do all format conversions manually. Normally you would want to use a known format that your program can handle -natively, which you can do by specifying it after this function returns, like so: - - ma_device_config config = ma_device_config_init(ma_device_type_playback); - config.callback = my_data_callback; - config.pUserData = pMyUserData; - config.format = ma_format_f32; - config.channels = 2; - config.sampleRate = 44100; - -In this case miniaudio will perform all of the necessary data conversion for you behind the scenes. - -Currently miniaudio only supports asynchronous, callback based data delivery which means you must specify callback. A -pointer to user data can also be specified which is set in the pUserData member of the ma_device object. - -To specify a channel map you can use ma_get_standard_channel_map(): - - ma_get_standard_channel_map(ma_standard_channel_map_default, config.channels, config.channelMap); - -Alternatively you can set the channel map manually if you need something specific or something that isn't one of miniaudio's -stock channel maps. - -By default the system's default device will be used. Set the pDeviceID member to a pointer to a ma_device_id object to -use a specific device. You can enumerate over the devices with ma_context_enumerate_devices() or ma_context_get_devices() -which will give you access to the device ID. Set pDeviceID to NULL to use the default device. - -The device type can be one of the ma_device_type's: - ma_device_type_playback - ma_device_type_capture - ma_device_type_duplex - -Thread Safety: SAFE -*/ -ma_device_config ma_device_config_init(ma_device_type deviceType); - - -/************************************************************************************************************************************************************ - -Utiltities - -************************************************************************************************************************************************************/ +Uninitializes a context. -/* -Creates a mutex. -A mutex must be created from a valid context. A mutex is initially unlocked. -*/ -ma_result ma_mutex_init(ma_context* pContext, ma_mutex* pMutex); +Return Value +------------ +MA_SUCCESS if successful; any other error code otherwise. -/* -Deletes a mutex. -*/ -void ma_mutex_uninit(ma_mutex* pMutex); -/* -Locks a mutex with an infinite timeout. -*/ -void ma_mutex_lock(ma_mutex* pMutex); +Thread Safety +------------- +Unsafe. Do not call this function across multiple threads as some backends read and write to global state. -/* -Unlocks a mutex. -*/ -void ma_mutex_unlock(ma_mutex* pMutex); +Remarks +------- +Results are undefined if you call this while any device created by this context is still active. -/* -Retrieves a friendly name for a backend. -*/ -const char* ma_get_backend_name(ma_backend backend); -/* -Determines whether or not loopback mode is support by a backend. +See Also +-------- +ma_context_init() */ -ma_bool32 ma_is_loopback_supported(ma_backend backend); - +ma_result ma_context_uninit(ma_context* pContext); /* -Adjust buffer size based on a scaling factor. +Enumerates over every device (both playback and capture). -This just multiplies the base size by the scaling factor, making sure it's a size of at least 1. -*/ -ma_uint32 ma_scale_buffer_size(ma_uint32 baseBufferSize, float scale); +This is a lower-level enumeration function to the easier to use `ma_context_get_devices()`. Use `ma_context_enumerate_devices()` if you would rather not incur +an internal heap allocation, or it simply suits your code better. -/* -Calculates a buffer size in milliseconds from the specified number of frames and sample rate. -*/ -ma_uint32 ma_calculate_buffer_size_in_milliseconds_from_frames(ma_uint32 bufferSizeInFrames, ma_uint32 sampleRate); +Note that this only retrieves the ID and name/description of the device. The reason for only retrieving basic information is that it would otherwise require +opening the backend device in order to probe it for more detailed information which can be inefficient. Consider using `ma_context_get_device_info()` for this, +but don't call it from within the enumeration callback. -/* -Calculates a buffer size in frames from the specified number of milliseconds and sample rate. -*/ -ma_uint32 ma_calculate_buffer_size_in_frames_from_milliseconds(ma_uint32 bufferSizeInMilliseconds, ma_uint32 sampleRate); +Returning false from the callback will stop enumeration. Returning true will continue enumeration. -/* -Retrieves the default buffer size in milliseconds based on the specified performance profile. -*/ -ma_uint32 ma_get_default_buffer_size_in_milliseconds(ma_performance_profile performanceProfile); -/* -Calculates a buffer size in frames for the specified performance profile and scale factor. -*/ -ma_uint32 ma_get_default_buffer_size_in_frames(ma_performance_profile performanceProfile, ma_uint32 sampleRate); +Parameters +---------- +pContext (in) + A pointer to the context performing the enumeration. -/* -Copies silent frames into the given buffer. -*/ -void ma_zero_pcm_frames(void* p, ma_uint32 frameCount, ma_format format, ma_uint32 channels); +callback (in) + The callback to fire for each enumerated device. -/* -Clips f32 samples. -*/ -void ma_clip_samples_f32(float* p, ma_uint32 sampleCount); -MA_INLINE void ma_clip_pcm_frames_f32(float* p, ma_uint32 frameCount, ma_uint32 channels) { ma_clip_samples_f32(p, frameCount*channels); } +pUserData (in) + A pointer to application-defined data passed to the callback. -/* -Helper for applying a volume factor to samples. -Note that the source and destination buffers can be the same, in which case it'll perform the operation in-place. -*/ -void ma_copy_and_apply_volume_factor_u8(ma_uint8* pSamplesOut, const ma_uint8* pSamplesIn, ma_uint32 sampleCount, float factor); -void ma_copy_and_apply_volume_factor_s16(ma_int16* pSamplesOut, const ma_int16* pSamplesIn, ma_uint32 sampleCount, float factor); -void ma_copy_and_apply_volume_factor_s24(void* pSamplesOut, const void* pSamplesIn, ma_uint32 sampleCount, float factor); -void ma_copy_and_apply_volume_factor_s32(ma_int32* pSamplesOut, const ma_int32* pSamplesIn, ma_uint32 sampleCount, float factor); -void ma_copy_and_apply_volume_factor_f32(float* pSamplesOut, const float* pSamplesIn, ma_uint32 sampleCount, float factor); +Return Value +------------ +MA_SUCCESS if successful; any other error code otherwise. -void ma_apply_volume_factor_u8(ma_uint8* pSamples, ma_uint32 sampleCount, float factor); -void ma_apply_volume_factor_s16(ma_int16* pSamples, ma_uint32 sampleCount, float factor); -void ma_apply_volume_factor_s24(void* pSamples, ma_uint32 sampleCount, float factor); -void ma_apply_volume_factor_s32(ma_int32* pSamples, ma_uint32 sampleCount, float factor); -void ma_apply_volume_factor_f32(float* pSamples, ma_uint32 sampleCount, float factor); -void ma_copy_and_apply_volume_factor_pcm_frames_u8(ma_uint8* pPCMFramesOut, const ma_uint8* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor); -void ma_copy_and_apply_volume_factor_pcm_frames_s16(ma_int16* pPCMFramesOut, const ma_int16* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor); -void ma_copy_and_apply_volume_factor_pcm_frames_s24(void* pPCMFramesOut, const void* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor); -void ma_copy_and_apply_volume_factor_pcm_frames_s32(ma_int32* pPCMFramesOut, const ma_int32* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor); -void ma_copy_and_apply_volume_factor_pcm_frames_f32(float* pPCMFramesOut, const float* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor); -void ma_copy_and_apply_volume_factor_pcm_frames(void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount, ma_format format, ma_uint32 channels, float factor); +Thread Safety +------------- +Safe. This is guarded using a simple mutex lock. -void ma_apply_volume_factor_pcm_frames_u8(ma_uint8* pFrames, ma_uint32 frameCount, ma_uint32 channels, float factor); -void ma_apply_volume_factor_pcm_frames_s16(ma_int16* pFrames, ma_uint32 frameCount, ma_uint32 channels, float factor); -void ma_apply_volume_factor_pcm_frames_s24(void* pFrames, ma_uint32 frameCount, ma_uint32 channels, float factor); -void ma_apply_volume_factor_pcm_frames_s32(ma_int32* pFrames, ma_uint32 frameCount, ma_uint32 channels, float factor); -void ma_apply_volume_factor_pcm_frames_f32(float* pFrames, ma_uint32 frameCount, ma_uint32 channels, float factor); -void ma_apply_volume_factor_pcm_frames(void* pFrames, ma_uint32 frameCount, ma_format format, ma_uint32 channels, float factor); +Remarks +------- +Do _not_ assume the first enumerated device of a given type is the default device. -/* -Helper for converting a linear factor to gain in decibels. -*/ -float ma_factor_to_gain_db(float factor); +Some backends and platforms may only support default playback and capture devices. -/* -Helper for converting gain in decibels to a linear factor. -*/ -float ma_gain_db_to_factor(float gain); +In general, you should not do anything complicated from within the callback. In particular, do not try initializing a device from within the callback. Also, +do not try to call `ma_context_get_device_info()` from within the callback. -#endif /* MA_NO_DEVICE_IO */ +Consider using `ma_context_get_devices()` for a simpler and safer API, albeit at the expense of an internal heap allocation. +Example 1 - Simple Enumeration +------------------------------ +ma_bool32 ma_device_enum_callback(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pInfo, void* pUserData) +{ + printf("Device Name: %s\n", pInfo->name); + return MA_TRUE; +} +ma_result result = ma_context_enumerate_devices(&context, my_device_enum_callback, pMyUserData); +if (result != MA_SUCCESS) { + // Error. +} -/************************************************************************************************************************************************************ -Decoding +See Also +-------- +ma_context_get_devices() +*/ +ma_result ma_context_enumerate_devices(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData); -************************************************************************************************************************************************************/ -#ifndef MA_NO_DECODING +/* +Retrieves basic information about every active playback and/or capture device. -typedef struct ma_decoder ma_decoder; +This function will allocate memory internally for the device lists and return a pointer to them through the `ppPlaybackDeviceInfos` and `ppCaptureDeviceInfos` +parameters. If you do not want to incur the overhead of these allocations consider using `ma_context_enumerate_devices()` which will instead use a callback. -typedef enum -{ - ma_seek_origin_start, - ma_seek_origin_current -} ma_seek_origin; -typedef size_t (* ma_decoder_read_proc) (ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead); /* Returns the number of bytes read. */ -typedef ma_bool32 (* ma_decoder_seek_proc) (ma_decoder* pDecoder, int byteOffset, ma_seek_origin origin); -typedef ma_result (* ma_decoder_seek_to_pcm_frame_proc) (ma_decoder* pDecoder, ma_uint64 frameIndex); -typedef ma_result (* ma_decoder_uninit_proc) (ma_decoder* pDecoder); -typedef ma_uint64 (* ma_decoder_get_length_in_pcm_frames_proc)(ma_decoder* pDecoder); +Parameters +---------- +pContext (in) + A pointer to the context performing the enumeration. -typedef struct -{ - ma_format format; /* Set to 0 or ma_format_unknown to use the stream's internal format. */ - ma_uint32 channels; /* Set to 0 to use the stream's internal channels. */ - ma_uint32 sampleRate; /* Set to 0 to use the stream's internal sample rate. */ - ma_channel channelMap[MA_MAX_CHANNELS]; - ma_channel_mix_mode channelMixMode; - ma_dither_mode ditherMode; - ma_src_algorithm srcAlgorithm; - union - { - ma_src_config_sinc sinc; - } src; -} ma_decoder_config; +ppPlaybackDeviceInfos (out) + A pointer to a pointer that will receive the address of a buffer containing the list of `ma_device_info` structures for playback devices. -struct ma_decoder -{ - ma_decoder_read_proc onRead; - ma_decoder_seek_proc onSeek; - void* pUserData; - ma_uint64 readPointer; /* Used for returning back to a previous position after analysing the stream or whatnot. */ - ma_format internalFormat; - ma_uint32 internalChannels; - ma_uint32 internalSampleRate; - ma_channel internalChannelMap[MA_MAX_CHANNELS]; - ma_format outputFormat; - ma_uint32 outputChannels; - ma_uint32 outputSampleRate; - ma_channel outputChannelMap[MA_MAX_CHANNELS]; - ma_pcm_converter dsp; /* <-- Format conversion is achieved by running frames through this. */ - ma_decoder_seek_to_pcm_frame_proc onSeekToPCMFrame; - ma_decoder_uninit_proc onUninit; - ma_decoder_get_length_in_pcm_frames_proc onGetLengthInPCMFrames; - void* pInternalDecoder; /* <-- The drwav/drflac/stb_vorbis/etc. objects. */ - struct - { - const ma_uint8* pData; - size_t dataSize; - size_t currentReadPos; - } memory; /* Only used for decoders that were opened against a block of memory. */ -}; +pPlaybackDeviceCount (out) + A pointer to an unsigned integer that will receive the number of playback devices. -ma_decoder_config ma_decoder_config_init(ma_format outputFormat, ma_uint32 outputChannels, ma_uint32 outputSampleRate); +ppCaptureDeviceInfos (out) + A pointer to a pointer that will receive the address of a buffer containing the list of `ma_device_info` structures for capture devices. -ma_result ma_decoder_init(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -ma_result ma_decoder_init_wav(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -ma_result ma_decoder_init_flac(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -ma_result ma_decoder_init_vorbis(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -ma_result ma_decoder_init_mp3(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -ma_result ma_decoder_init_raw(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfigIn, const ma_decoder_config* pConfigOut, ma_decoder* pDecoder); +pCaptureDeviceCount (out) + A pointer to an unsigned integer that will receive the number of capture devices. -ma_result ma_decoder_init_memory(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -ma_result ma_decoder_init_memory_wav(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -ma_result ma_decoder_init_memory_flac(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -ma_result ma_decoder_init_memory_vorbis(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -ma_result ma_decoder_init_memory_mp3(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -ma_result ma_decoder_init_memory_raw(const void* pData, size_t dataSize, const ma_decoder_config* pConfigIn, const ma_decoder_config* pConfigOut, ma_decoder* pDecoder); -#ifndef MA_NO_STDIO -ma_result ma_decoder_init_file(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -ma_result ma_decoder_init_file_wav(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -ma_result ma_decoder_init_file_flac(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -ma_result ma_decoder_init_file_vorbis(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -ma_result ma_decoder_init_file_mp3(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +Return Value +------------ +MA_SUCCESS if successful; any other error code otherwise. -ma_result ma_decoder_init_file_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -ma_result ma_decoder_init_file_wav_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -ma_result ma_decoder_init_file_flac_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -ma_result ma_decoder_init_file_vorbis_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -ma_result ma_decoder_init_file_mp3_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -#endif -ma_result ma_decoder_uninit(ma_decoder* pDecoder); +Thread Safety +------------- +Unsafe. Since each call to this function invalidates the pointers from the previous call, you should not be calling this simultaneously across multiple +threads. Instead, you need to make a copy of the returned data with your own higher level synchronization. -/* -Retrieves the length of the decoder in PCM frames. -Do not call this on streams of an undefined length, such as internet radio. +Remarks +------- +It is _not_ safe to assume the first device in the list is the default device. -If the length is unknown or an error occurs, 0 will be returned. +You can pass in NULL for the playback or capture lists in which case they'll be ignored. -This will always return 0 for Vorbis decoders. This is due to a limitation with stb_vorbis in push mode which is what miniaudio -uses internally. +The returned pointers will become invalid upon the next call this this function, or when the context is uninitialized. Do not free the returned pointers. -This will run in linear time for MP3 decoders. Do not call this in time critical scenarios. -*/ -ma_uint64 ma_decoder_get_length_in_pcm_frames(ma_decoder* pDecoder); -ma_uint64 ma_decoder_read_pcm_frames(ma_decoder* pDecoder, void* pFramesOut, ma_uint64 frameCount); -ma_result ma_decoder_seek_to_pcm_frame(ma_decoder* pDecoder, ma_uint64 frameIndex); +See Also +-------- +ma_context_get_devices() +*/ +ma_result ma_context_get_devices(ma_context* pContext, ma_device_info** ppPlaybackDeviceInfos, ma_uint32* pPlaybackDeviceCount, ma_device_info** ppCaptureDeviceInfos, ma_uint32* pCaptureDeviceCount); /* -Helper for opening and decoding a file into a heap allocated block of memory. Free the returned pointer with ma_free(). On input, -pConfig should be set to what you want. On output it will be set to what you got. -*/ -#ifndef MA_NO_STDIO -ma_result ma_decode_file(const char* pFilePath, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppDataOut); -#endif -ma_result ma_decode_memory(const void* pData, size_t dataSize, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppDataOut); +Retrieves information about a device of the given type, with the specified ID and share mode. -#endif /* MA_NO_DECODING */ +Parameters +---------- +pContext (in) + A pointer to the context performing the query. -/************************************************************************************************************************************************************ +deviceType (in) + The type of the device being queried. Must be either `ma_device_type_playback` or `ma_device_type_capture`. -Generation +pDeviceID (in) + The ID of the device being queried. -************************************************************************************************************************************************************/ -typedef struct -{ - double amplitude; - double periodsPerSecond; - double delta; - double time; -} ma_sine_wave; +shareMode (in) + The share mode to query for device capabilities. This should be set to whatever you're intending on using when initializing the device. If you're unsure, + set this to `ma_share_mode_shared`. -ma_result ma_sine_wave_init(double amplitude, double period, ma_uint32 sampleRate, ma_sine_wave* pSineWave); -ma_uint64 ma_sine_wave_read_f32(ma_sine_wave* pSineWave, ma_uint64 count, float* pSamples); -ma_uint64 ma_sine_wave_read_f32_ex(ma_sine_wave* pSineWave, ma_uint64 frameCount, ma_uint32 channels, ma_stream_layout layout, float** ppFrames); +pDeviceInfo (out) + A pointer to the `ma_device_info` structure that will receive the device information. -#ifdef __cplusplus -} -#endif -#endif /* miniaudio_h */ +Return Value +------------ +MA_SUCCESS if successful; any other error code otherwise. -/************************************************************************************************************************************************************ -************************************************************************************************************************************************************* +Thread Safety +------------- +Safe. This is guarded using a simple mutex lock. -IMPLEMENTATION -************************************************************************************************************************************************************* -************************************************************************************************************************************************************/ -#if defined(MINIAUDIO_IMPLEMENTATION) || defined(MA_IMPLEMENTATION) -#include -#include /* For INT_MAX */ -#include /* sin(), etc. */ +Remarks +------- +Do _not_ call this from within the `ma_context_enumerate_devices()` callback. -#if defined(MA_DEBUG_OUTPUT) -#include /* for printf() for debug output */ -#endif +It's possible for a device to have different information and capabilities depending on whether or not it's opened in shared or exclusive mode. For example, in +shared mode, WASAPI always uses floating point samples for mixing, but in exclusive mode it can be anything. Therefore, this function allows you to specify +which share mode you want information for. Note that not all backends and devices support shared or exclusive mode, in which case this function will fail if +the requested share mode is unsupported. -#ifdef MA_WIN32 -// @raysan5: To avoid conflicting windows.h symbols with raylib, so flags are defined -// WARNING: Those flags avoid inclusion of some Win32 headers that could be required -// by user at some point and won't be included... -//------------------------------------------------------------------------------------- - -// If defined, the following flags inhibit definition of the indicated items. -#define NOGDICAPMASKS // CC_*, LC_*, PC_*, CP_*, TC_*, RC_ -#define NOVIRTUALKEYCODES // VK_* -#define NOWINMESSAGES // WM_*, EM_*, LB_*, CB_* -#define NOWINSTYLES // WS_*, CS_*, ES_*, LBS_*, SBS_*, CBS_* -#define NOSYSMETRICS // SM_* -#define NOMENUS // MF_* -#define NOICONS // IDI_* -#define NOKEYSTATES // MK_* -#define NOSYSCOMMANDS // SC_* -#define NORASTEROPS // Binary and Tertiary raster ops -#define NOSHOWWINDOW // SW_* -#define OEMRESOURCE // OEM Resource values -#define NOATOM // Atom Manager routines -#define NOCLIPBOARD // Clipboard routines -#define NOCOLOR // Screen colors -#define NOCTLMGR // Control and Dialog routines -#define NODRAWTEXT // DrawText() and DT_* -#define NOGDI // All GDI defines and routines -#define NOKERNEL // All KERNEL defines and routines -#define NOUSER // All USER defines and routines -//#define NONLS // All NLS defines and routines -#define NOMB // MB_* and MessageBox() -#define NOMEMMGR // GMEM_*, LMEM_*, GHND, LHND, associated routines -#define NOMETAFILE // typedef METAFILEPICT -#define NOMINMAX // Macros min(a,b) and max(a,b) -#define NOMSG // typedef MSG and associated routines -#define NOOPENFILE // OpenFile(), OemToAnsi, AnsiToOem, and OF_* -#define NOSCROLL // SB_* and scrolling routines -#define NOSERVICE // All Service Controller routines, SERVICE_ equates, etc. -#define NOSOUND // Sound driver routines -#define NOTEXTMETRIC // typedef TEXTMETRIC and associated routines -#define NOWH // SetWindowsHook and WH_* -#define NOWINOFFSETS // GWL_*, GCL_*, associated routines -#define NOCOMM // COMM driver routines -#define NOKANJI // Kanji support stuff. -#define NOHELP // Help engine interface. -#define NOPROFILER // Profiler interface. -#define NODEFERWINDOWPOS // DeferWindowPos routines -#define NOMCX // Modem Configuration Extensions - -// Type required before windows.h inclusion -typedef struct tagMSG *LPMSG; +This leaves pDeviceInfo unmodified in the result of an error. +*/ +ma_result ma_context_get_device_info(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo); -#include +/* +Determines if the given context supports loopback mode. -// Type required by some unused function... -typedef struct tagBITMAPINFOHEADER { - DWORD biSize; - LONG biWidth; - LONG biHeight; - WORD biPlanes; - WORD biBitCount; - DWORD biCompression; - DWORD biSizeImage; - LONG biXPelsPerMeter; - LONG biYPelsPerMeter; - DWORD biClrUsed; - DWORD biClrImportant; -} BITMAPINFOHEADER, *PBITMAPINFOHEADER; -#include -#include -#include +Parameters +---------- +pContext (in) + A pointer to the context getting queried. -// @raysan5: Some required types defined for MSVC/TinyC compiler -#if defined(_MSC_VER) || defined(__TINYC__) - #include "propidl.h" -#endif -//---------------------------------------------------------------------------------- -#else -#include /* For malloc(), free(), wcstombs(). */ -#include /* For memset() */ -#endif -#if defined(MA_APPLE) && (__MAC_OS_X_VERSION_MIN_REQUIRED < 101200) -#include /* For mach_absolute_time() */ -#endif +Return Value +------------ +MA_TRUE if the context supports loopback mode; MA_FALSE otherwise. +*/ +ma_bool32 ma_context_is_loopback_supported(ma_context* pContext); -#ifdef MA_POSIX -#include -#include -#include -#include -#endif -#ifdef MA_EMSCRIPTEN -#include -#endif -#if !defined(MA_64BIT) && !defined(MA_32BIT) -#ifdef _WIN32 -#ifdef _WIN64 -#define MA_64BIT -#else -#define MA_32BIT -#endif -#endif -#endif +/* +Initializes a device config with default settings. -#if !defined(MA_64BIT) && !defined(MA_32BIT) -#ifdef __GNUC__ -#ifdef __LP64__ -#define MA_64BIT -#else -#define MA_32BIT -#endif -#endif -#endif -#if !defined(MA_64BIT) && !defined(MA_32BIT) -#include -#if INTPTR_MAX == INT64_MAX -#define MA_64BIT -#else -#define MA_32BIT -#endif -#endif +Parameters +---------- +deviceType (in) + The type of the device this config is being initialized for. This must set to one of the following: -/* Architecture Detection */ -#if defined(__x86_64__) || defined(_M_X64) -#define MA_X64 -#elif defined(__i386) || defined(_M_IX86) -#define MA_X86 -#elif defined(__arm__) || defined(_M_ARM) -#define MA_ARM -#endif + |-------------------------| + | Device Type | + |-------------------------| + | ma_device_type_playback | + | ma_device_type_capture | + | ma_device_type_duplex | + | ma_device_type_loopback | + |-------------------------| -/* Cannot currently support AVX-512 if AVX is disabled. */ -#if !defined(MA_NO_AVX512) && defined(MA_NO_AVX2) -#define MA_NO_AVX512 -#endif -/* Intrinsics Support */ -#if defined(MA_X64) || defined(MA_X86) - #if defined(_MSC_VER) && !defined(__clang__) - /* MSVC. */ - #if _MSC_VER >= 1400 && !defined(MA_NO_SSE2) /* 2005 */ - #define MA_SUPPORT_SSE2 - #endif - /*#if _MSC_VER >= 1600 && !defined(MA_NO_AVX)*/ /* 2010 */ - /* #define MA_SUPPORT_AVX*/ - /*#endif*/ - #if _MSC_VER >= 1700 && !defined(MA_NO_AVX2) /* 2012 */ - #define MA_SUPPORT_AVX2 - #endif - #if _MSC_VER >= 1910 && !defined(MA_NO_AVX512) /* 2017 */ - #define MA_SUPPORT_AVX512 - #endif - #else - /* Assume GNUC-style. */ - #if defined(__SSE2__) && !defined(MA_NO_SSE2) - #define MA_SUPPORT_SSE2 - #endif - /*#if defined(__AVX__) && !defined(MA_NO_AVX)*/ - /* #define MA_SUPPORT_AVX*/ - /*#endif*/ - #if defined(__AVX2__) && !defined(MA_NO_AVX2) - #define MA_SUPPORT_AVX2 - #endif - #if defined(__AVX512F__) && !defined(MA_NO_AVX512) - #define MA_SUPPORT_AVX512 - #endif - #endif +Return Value +------------ +A new device config object with default settings. You will typically want to adjust the config after this function returns. See remarks. + - /* If at this point we still haven't determined compiler support for the intrinsics just fall back to __has_include. */ - #if !defined(__GNUC__) && !defined(__clang__) && defined(__has_include) - #if !defined(MA_SUPPORT_SSE2) && !defined(MA_NO_SSE2) && __has_include() - #define MA_SUPPORT_SSE2 - #endif - /*#if !defined(MA_SUPPORT_AVX) && !defined(MA_NO_AVX) && __has_include()*/ - /* #define MA_SUPPORT_AVX*/ - /*#endif*/ - #if !defined(MA_SUPPORT_AVX2) && !defined(MA_NO_AVX2) && __has_include() - #define MA_SUPPORT_AVX2 - #endif - #if !defined(MA_SUPPORT_AVX512) && !defined(MA_NO_AVX512) && __has_include() - #define MA_SUPPORT_AVX512 - #endif - #endif +Thread Safety +------------- +Safe. - #if defined(MA_SUPPORT_AVX512) - #include /* Not a mistake. Intentionally including instead of because otherwise the compiler will complain. */ - #elif defined(MA_SUPPORT_AVX2) || defined(MA_SUPPORT_AVX) - #include - #elif defined(MA_SUPPORT_SSE2) - #include - #endif -#endif -#if defined(MA_ARM) - #if !defined(MA_NO_NEON) && (defined(__ARM_NEON) || defined(__aarch64__) || defined(_M_ARM64)) - #define MA_SUPPORT_NEON - #endif +Callback Safety +--------------- +Safe, but don't try initializing a device in a callback. - /* Fall back to looking for the #include file. */ - #if !defined(__GNUC__) && !defined(__clang__) && defined(__has_include) - #if !defined(MA_SUPPORT_NEON) && !defined(MA_NO_NEON) && __has_include() - #define MA_SUPPORT_NEON - #endif - #endif - #if defined(MA_SUPPORT_NEON) - #include - #endif -#endif +Remarks +------- +The returned config will be initialized to defaults. You will normally want to customize a few variables before initializing the device. See Example 1 for a +typical configuration which sets the sample format, channel count, sample rate, data callback and user data. These are usually things you will want to change +before initializing the device. -#if defined(_MSC_VER) - #pragma warning(push) - #pragma warning(disable:4752) /* found Intel(R) Advanced Vector Extensions; consider using /arch:AVX */ -#endif +See `ma_device_init()` for details on specific configuration options. -#if defined(MA_X64) || defined(MA_X86) - #if defined(_MSC_VER) && !defined(__clang__) - #if _MSC_VER >= 1400 - #include - static MA_INLINE void ma_cpuid(int info[4], int fid) - { - __cpuid(info, fid); - } - #else - #define MA_NO_CPUID - #endif - #if _MSC_VER >= 1600 && (defined(_MSC_FULL_VER) && _MSC_FULL_VER >= 160040219) - static MA_INLINE unsigned __int64 ma_xgetbv(int reg) - { - return _xgetbv(reg); - } - #else - #define MA_NO_XGETBV - #endif - #elif (defined(__GNUC__) || defined(__clang__)) && !defined(MA_ANDROID) - static MA_INLINE void ma_cpuid(int info[4], int fid) - { - /* - It looks like the -fPIC option uses the ebx register which GCC complains about. We can work around this by just using a different register, the - specific register of which I'm letting the compiler decide on. The "k" prefix is used to specify a 32-bit register. The {...} syntax is for - supporting different assembly dialects. - - What's basically happening is that we're saving and restoring the ebx register manually. - */ - #if defined(DRFLAC_X86) && defined(__PIC__) - __asm__ __volatile__ ( - "xchg{l} {%%}ebx, %k1;" - "cpuid;" - "xchg{l} {%%}ebx, %k1;" - : "=a"(info[0]), "=&r"(info[1]), "=c"(info[2]), "=d"(info[3]) : "a"(fid), "c"(0) - ); - #else - __asm__ __volatile__ ( - "cpuid" : "=a"(info[0]), "=b"(info[1]), "=c"(info[2]), "=d"(info[3]) : "a"(fid), "c"(0) - ); - #endif - } +Example 1 - Simple Configuration +-------------------------------- +The example below is what a program will typically want to configure for each device at a minimum. Notice how `ma_device_config_init()` is called first, and +then the returned object is modified directly. This is important because it ensures that your program continues to work as new configuration options are added +to the `ma_device_config` structure. - static MA_INLINE ma_uint64 ma_xgetbv(int reg) - { - unsigned int hi; - unsigned int lo; +```c +ma_device_config config = ma_device_config_init(ma_device_type_playback); +config.playback.format = ma_format_f32; +config.playback.channels = 2; +config.sampleRate = 48000; +config.dataCallback = ma_data_callback; +config.pUserData = pMyUserData; +``` - __asm__ __volatile__ ( - "xgetbv" : "=a"(lo), "=d"(hi) : "c"(reg) - ); - return ((ma_uint64)hi << 32) | (ma_uint64)lo; - } - #else - #define MA_NO_CPUID - #define MA_NO_XGETBV - #endif -#else - #define MA_NO_CPUID - #define MA_NO_XGETBV -#endif +See Also +-------- +ma_device_init() +ma_device_init_ex() +*/ +ma_device_config ma_device_config_init(ma_device_type deviceType); -static MA_INLINE ma_bool32 ma_has_sse2() -{ -#if defined(MA_SUPPORT_SSE2) - #if (defined(MA_X64) || defined(MA_X86)) && !defined(MA_NO_SSE2) - #if defined(MA_X64) - return MA_TRUE; /* 64-bit targets always support SSE2. */ - #elif (defined(_M_IX86_FP) && _M_IX86_FP == 2) || defined(__SSE2__) - return MA_TRUE; /* If the compiler is allowed to freely generate SSE2 code we can assume support. */ - #else - #if defined(MA_NO_CPUID) - return MA_FALSE; - #else - int info[4]; - ma_cpuid(info, 1); - return (info[3] & (1 << 26)) != 0; - #endif - #endif - #else - return MA_FALSE; /* SSE2 is only supported on x86 and x64 architectures. */ - #endif -#else - return MA_FALSE; /* No compiler support. */ -#endif -} -#if 0 -static MA_INLINE ma_bool32 ma_has_avx() -{ -#if defined(MA_SUPPORT_AVX) - #if (defined(MA_X64) || defined(MA_X86)) && !defined(MA_NO_AVX) - #if defined(_AVX_) || defined(__AVX__) - return MA_TRUE; /* If the compiler is allowed to freely generate AVX code we can assume support. */ - #else - /* AVX requires both CPU and OS support. */ - #if defined(MA_NO_CPUID) || defined(MA_NO_XGETBV) - return MA_FALSE; - #else - int info[4]; - ma_cpuid(info, 1); - if (((info[2] & (1 << 27)) != 0) && ((info[2] & (1 << 28)) != 0)) { - ma_uint64 xrc = ma_xgetbv(0); - if ((xrc & 0x06) == 0x06) { - return MA_TRUE; - } else { - return MA_FALSE; - } - } else { - return MA_FALSE; - } - #endif - #endif - #else - return MA_FALSE; /* AVX is only supported on x86 and x64 architectures. */ - #endif -#else - return MA_FALSE; /* No compiler support. */ -#endif -} -#endif +/* +Initializes a device. -static MA_INLINE ma_bool32 ma_has_avx2() -{ -#if defined(MA_SUPPORT_AVX2) - #if (defined(MA_X64) || defined(MA_X86)) && !defined(MA_NO_AVX2) - #if defined(_AVX2_) || defined(__AVX2__) - return MA_TRUE; /* If the compiler is allowed to freely generate AVX2 code we can assume support. */ - #else - /* AVX2 requires both CPU and OS support. */ - #if defined(MA_NO_CPUID) || defined(MA_NO_XGETBV) - return MA_FALSE; - #else - int info1[4]; - int info7[4]; - ma_cpuid(info1, 1); - ma_cpuid(info7, 7); - if (((info1[2] & (1 << 27)) != 0) && ((info7[1] & (1 << 5)) != 0)) { - ma_uint64 xrc = ma_xgetbv(0); - if ((xrc & 0x06) == 0x06) { - return MA_TRUE; - } else { - return MA_FALSE; - } - } else { - return MA_FALSE; - } - #endif - #endif - #else - return MA_FALSE; /* AVX2 is only supported on x86 and x64 architectures. */ - #endif -#else - return MA_FALSE; /* No compiler support. */ -#endif -} +A device represents a physical audio device. The idea is you send or receive audio data from the device to either play it back through a speaker, or capture it +from a microphone. Whether or not you should send or receive data from the device (or both) depends on the type of device you are initializing which can be +playback, capture, full-duplex or loopback. (Note that loopback mode is only supported on select backends.) Sending and receiving audio data to and from the +device is done via a callback which is fired by miniaudio at periodic time intervals. -static MA_INLINE ma_bool32 ma_has_avx512f() -{ -#if defined(MA_SUPPORT_AVX512) - #if (defined(MA_X64) || defined(MA_X86)) && !defined(MA_NO_AVX512) - #if defined(__AVX512F__) - return MA_TRUE; /* If the compiler is allowed to freely generate AVX-512F code we can assume support. */ - #else - /* AVX-512 requires both CPU and OS support. */ - #if defined(MA_NO_CPUID) || defined(MA_NO_XGETBV) - return MA_FALSE; - #else - int info1[4]; - int info7[4]; - ma_cpuid(info1, 1); - ma_cpuid(info7, 7); - if (((info1[2] & (1 << 27)) != 0) && ((info7[1] & (1 << 16)) != 0)) { - ma_uint64 xrc = ma_xgetbv(0); - if ((xrc & 0xE6) == 0xE6) { - return MA_TRUE; - } else { - return MA_FALSE; - } - } else { - return MA_FALSE; - } - #endif - #endif - #else - return MA_FALSE; /* AVX-512F is only supported on x86 and x64 architectures. */ - #endif -#else - return MA_FALSE; /* No compiler support. */ -#endif -} +The frequency at which data is deilvered to and from a device depends on the size of it's period which is defined by a buffer size and a period count. The size +of the buffer can be defined in terms of PCM frames or milliseconds, whichever is more convenient. The size of a period is the size of this buffer, divided by +the period count. Generally speaking, the smaller the period, the lower the latency at the expense of higher CPU usage and increased risk of glitching due to +the more frequent and granular data deliver intervals. The size of a period will depend on your requirements, but miniaudio's defaults should work fine for +most scenarios. If you're building a game you should leave this fairly small, whereas if you're building a simple media player you can make it larger. Note +that the period size you request is actually just a hint - miniaudio will tell the backend what you want, but the backend is ultimately responsible for what it +gives you. You cannot assume you will get exactly what you ask for. -static MA_INLINE ma_bool32 ma_has_neon() -{ -#if defined(MA_SUPPORT_NEON) - #if defined(MA_ARM) && !defined(MA_NO_NEON) - #if (defined(__ARM_NEON) || defined(__aarch64__) || defined(_M_ARM64)) - return MA_TRUE; /* If the compiler is allowed to freely generate NEON code we can assume support. */ - #else - /* TODO: Runtime check. */ - return MA_FALSE; - #endif - #else - return MA_FALSE; /* NEON is only supported on ARM architectures. */ - #endif -#else - return MA_FALSE; /* No compiler support. */ -#endif -} +When delivering data to and from a device you need to make sure it's in the correct format which you can set through the device configuration. You just set the +format that you want to use and miniaudio will perform all of the necessary conversion for you internally. When delivering data to and from the callback you +can assume the format is the same as what you requested when you initialized the device. See Remarks for more details on miniaudio's data conversion pipeline. + + +Parameters +---------- +pContext (in, optional) + A pointer to the context that owns the device. This can be null, in which case it creates a default context internally. + +pConfig (in) + A pointer to the device configuration. Cannot be null. See remarks for details. + +pDevice (out) + A pointer to the device object being initialized. + + +Return Value +------------ +MA_SUCCESS if successful; any other error code otherwise. -static MA_INLINE ma_bool32 ma_is_little_endian() -{ -#if defined(MA_X86) || defined(MA_X64) - return MA_TRUE; -#else - int n = 1; - return (*(char*)&n) == 1; -#endif -} +Thread Safety +------------- +Unsafe. It is not safe to call this function simultaneously for different devices because some backends depend on and mutate global state. The same applies to +calling this at the same time as `ma_device_uninit()`. -static MA_INLINE ma_bool32 ma_is_big_endian() -{ - return !ma_is_little_endian(); -} +Callback Safety +--------------- +Unsafe. It is not safe to call this inside any callback. -#ifndef MA_COINIT_VALUE -#define MA_COINIT_VALUE 0 /* 0 = COINIT_MULTITHREADED*/ -#endif +Remarks +------- +Setting `pContext` to NULL will result in miniaudio creating a default context internally and is equivalent to passing in a context initialized like so: + ```c + ma_context_init(NULL, 0, NULL, &context); + ``` -#ifndef MA_PI -#define MA_PI 3.14159265358979323846264f -#endif -#ifndef MA_PI_D -#define MA_PI_D 3.14159265358979323846264 -#endif -#ifndef MA_TAU -#define MA_TAU 6.28318530717958647693f -#endif -#ifndef MA_TAU_D -#define MA_TAU_D 6.28318530717958647693 -#endif +Do not set `pContext` to NULL if you are needing to open multiple devices. You can, however, use NULL when initializing the first device, and then use +device.pContext for the initialization of other devices. +The device can be configured via the `pConfig` argument. The config object is initialized with `ma_device_config_init()`. Individual configuration settings can +then be set directly on the structure. Below are the members of the `ma_device_config` object. -/* The default format when ma_format_unknown (0) is requested when initializing a device. */ -#ifndef MA_DEFAULT_FORMAT -#define MA_DEFAULT_FORMAT ma_format_f32 -#endif + deviceType + Must be `ma_device_type_playback`, `ma_device_type_capture`, `ma_device_type_duplex` of `ma_device_type_loopback`. -/* The default channel count to use when 0 is used when initializing a device. */ -#ifndef MA_DEFAULT_CHANNELS -#define MA_DEFAULT_CHANNELS 2 -#endif + sampleRate + The sample rate, in hertz. The most common sample rates are 48000 and 44100. Setting this to 0 will use the device's native sample rate. -/* The default sample rate to use when 0 is used when initializing a device. */ -#ifndef MA_DEFAULT_SAMPLE_RATE -#define MA_DEFAULT_SAMPLE_RATE 48000 -#endif + periodSizeInFrames + The desired size of a period in PCM frames. If this is 0, `periodSizeInMilliseconds` will be used instead. If both are 0 the default buffer size will + be used depending on the selected performance profile. This value affects latency. See below for details. -/* Default periods when none is specified in ma_device_init(). More periods means more work on the CPU. */ -#ifndef MA_DEFAULT_PERIODS -#define MA_DEFAULT_PERIODS 3 -#endif + periodSizeInMilliseconds + The desired size of a period in milliseconds. If this is 0, `periodSizeInFrames` will be used instead. If both are 0 the default buffer size will be + used depending on the selected performance profile. The value affects latency. See below for details. -/* The base buffer size in milliseconds for low latency mode. */ -#ifndef MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_LOW_LATENCY -#define MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_LOW_LATENCY (10*MA_DEFAULT_PERIODS) -#endif + periods + The number of periods making up the device's entire buffer. The total buffer size is `periodSizeInFrames` or `periodSizeInMilliseconds` multiplied by + this value. This is just a hint as backends will be the ones who ultimately decide how your periods will be configured. -/* The base buffer size in milliseconds for conservative mode. */ -#ifndef MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_CONSERVATIVE -#define MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_CONSERVATIVE (100*MA_DEFAULT_PERIODS) -#endif + performanceProfile + A hint to miniaudio as to the performance requirements of your program. Can be either `ma_performance_profile_low_latency` (default) or + `ma_performance_profile_conservative`. This mainly affects the size of default buffers and can usually be left at it's default value. + noPreZeroedOutputBuffer + When set to true, the contents of the output buffer passed into the data callback will be left undefined. When set to false (default), the contents of + the output buffer will be cleared the zero. You can use this to avoid the overhead of zeroing out the buffer if you know can guarantee that your data + callback will write to every sample in the output buffer, or if you are doing your own clearing. -/* Standard sample rates, in order of priority. */ -ma_uint32 g_maStandardSampleRatePriorities[] = { - MA_SAMPLE_RATE_48000, /* Most common */ - MA_SAMPLE_RATE_44100, + noClip + When set to true, the contents of the output buffer passed into the data callback will be clipped after returning. When set to false (default), the + contents of the output buffer are left alone after returning and it will be left up to the backend itself to decide whether or not the clip. This only + applies when the playback sample format is f32. - MA_SAMPLE_RATE_32000, /* Lows */ - MA_SAMPLE_RATE_24000, - MA_SAMPLE_RATE_22050, + dataCallback + The callback to fire whenever data is ready to be delivered to or from the device. - MA_SAMPLE_RATE_88200, /* Highs */ - MA_SAMPLE_RATE_96000, - MA_SAMPLE_RATE_176400, - MA_SAMPLE_RATE_192000, + stopCallback + The callback to fire whenever the device has stopped, either explicitly via `ma_device_stop()`, or implicitly due to things like the device being + disconnected. - MA_SAMPLE_RATE_16000, /* Extreme lows */ - MA_SAMPLE_RATE_11025, - MA_SAMPLE_RATE_8000, + pUserData + The user data pointer to use with the device. You can access this directly from the device object like `device.pUserData`. - MA_SAMPLE_RATE_352800, /* Extreme highs */ - MA_SAMPLE_RATE_384000 -}; + resampling.algorithm + The resampling algorithm to use when miniaudio needs to perform resampling between the rate specified by `sampleRate` and the device's native rate. The + default value is `ma_resample_algorithm_linear`, and the quality can be configured with `resampling.linear.lpfCount`. -ma_format g_maFormatPriorities[] = { - ma_format_s16, /* Most common */ - ma_format_f32, - - /*ma_format_s24_32,*/ /* Clean alignment */ - ma_format_s32, - - ma_format_s24, /* Unclean alignment */ - - ma_format_u8 /* Low quality */ -}; + resampling.linear.lpfCount + The linear resampler applies a low-pass filter as part of it's procesing for anti-aliasing. This setting controls the quality of the filter. The higher + the value, the better the quality. Setting this to 0 will disable low-pass filtering altogether. The maximum value is `MA_MAX_RESAMPLER_LPF_FILTERS`. + The default value is `min(2, MA_MAX_RESAMPLER_LPF_FILTERS)`. + playback.pDeviceID + A pointer to a `ma_device_id` structure containing the ID of the playback device to initialize. Setting this NULL (default) will use the system's + default playback device. Retrieve the device ID from the `ma_device_info` structure, which can be retrieved using device enumeration. + playback.format + The sample format to use for playback. When set to `ma_format_unknown` the device's native format will be used. This can be retrieved after + initialization from the device object directly with `device.playback.format`. -/****************************************************************************** + playback.channels + The number of channels to use for playback. When set to 0 the device's native channel count will be used. This can be retrieved after initialization + from the device object directly with `device.playback.channels`. -Standard Library Stuff + playback.channelMap + The channel map to use for playback. When left empty, the device's native channel map will be used. This can be retrieved after initialization from the + device object direct with `device.playback.channelMap`. -******************************************************************************/ -#ifndef MA_MALLOC -#ifdef MA_WIN32 -#define MA_MALLOC(sz) HeapAlloc(GetProcessHeap(), 0, (sz)) -#else -#define MA_MALLOC(sz) malloc((sz)) -#endif -#endif + playback.shareMode + The preferred share mode to use for playback. Can be either `ma_share_mode_shared` (default) or `ma_share_mode_exclusive`. Note that if you specify + exclusive mode, but it's not supported by the backend, initialization will fail. You can then fall back to shared mode if desired. -#ifndef MA_REALLOC -#ifdef MA_WIN32 -#define MA_REALLOC(p, sz) (((sz) > 0) ? ((p) ? HeapReAlloc(GetProcessHeap(), 0, (p), (sz)) : HeapAlloc(GetProcessHeap(), 0, (sz))) : ((VOID*)(size_t)(HeapFree(GetProcessHeap(), 0, (p)) & 0))) -#else -#define MA_REALLOC(p, sz) realloc((p), (sz)) -#endif -#endif + playback.pDeviceID + A pointer to a `ma_device_id` structure containing the ID of the playback device to initialize. Setting this NULL (default) will use the system's + default playback device. Retrieve the device ID from the `ma_device_info` structure, which can be retrieved using device enumeration. -#ifndef MA_FREE -#ifdef MA_WIN32 -#define MA_FREE(p) HeapFree(GetProcessHeap(), 0, (p)) -#else -#define MA_FREE(p) free((p)) -#endif -#endif + capture.format + The sample format to use for capture. When set to `ma_format_unknown` the device's native format will be used. This can be retrieved after + initialization from the device object directly with `device.capture.format`. -#ifndef MA_ZERO_MEMORY -#ifdef MA_WIN32 -#define MA_ZERO_MEMORY(p, sz) ZeroMemory((p), (sz)) -#else -#define MA_ZERO_MEMORY(p, sz) memset((p), 0, (sz)) -#endif -#endif + capture.channels + The number of channels to use for capture. When set to 0 the device's native channel count will be used. This can be retrieved after initialization + from the device object directly with `device.capture.channels`. -#ifndef MA_COPY_MEMORY -#ifdef MA_WIN32 -#define MA_COPY_MEMORY(dst, src, sz) CopyMemory((dst), (src), (sz)) -#else -#define MA_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz)) -#endif -#endif + capture.channelMap + The channel map to use for capture. When left empty, the device's native channel map will be used. This can be retrieved after initialization from the + device object direct with `device.capture.channelMap`. -#ifndef MA_ASSERT -#ifdef MA_WIN32 -#define MA_ASSERT(condition) assert(condition) -#else -#define MA_ASSERT(condition) assert(condition) -#endif -#endif + capture.shareMode + The preferred share mode to use for capture. Can be either `ma_share_mode_shared` (default) or `ma_share_mode_exclusive`. Note that if you specify + exclusive mode, but it's not supported by the backend, initialization will fail. You can then fall back to shared mode if desired. -#define ma_zero_memory MA_ZERO_MEMORY -#define ma_copy_memory MA_COPY_MEMORY -#define ma_assert MA_ASSERT + wasapi.noAutoConvertSRC + WASAPI only. When set to true, disables WASAPI's automatic resampling and forces the use of miniaudio's resampler. Defaults to false. -#define ma_zero_object(p) ma_zero_memory((p), sizeof(*(p))) -#define ma_countof(x) (sizeof(x) / sizeof(x[0])) -#define ma_max(x, y) (((x) > (y)) ? (x) : (y)) -#define ma_min(x, y) (((x) < (y)) ? (x) : (y)) -#define ma_clamp(x, lo, hi) (ma_max(lo, ma_min(x, hi))) -#define ma_offset_ptr(p, offset) (((ma_uint8*)(p)) + (offset)) + wasapi.noDefaultQualitySRC + WASAPI only. Only used when `wasapi.noAutoConvertSRC` is set to false. When set to true, disables the use of `AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY`. + You should usually leave this set to false, which is the default. -#define ma_buffer_frame_capacity(buffer, channels, format) (sizeof(buffer) / ma_get_bytes_per_sample(format) / (channels)) + wasapi.noAutoStreamRouting + WASAPI only. When set to true, disables automatic stream routing on the WASAPI backend. Defaults to false. -/* -Return Values: - 0: Success - 22: EINVAL - 34: ERANGE + wasapi.noHardwareOffloading + WASAPI only. When set to true, disables the use of WASAPI's hardware offloading feature. Defaults to false. -Not using symbolic constants for errors because I want to avoid #including errno.h -*/ -int ma_strcpy_s(char* dst, size_t dstSizeInBytes, const char* src) -{ - size_t i; + alsa.noMMap + ALSA only. When set to true, disables MMap mode. Defaults to false. - if (dst == 0) { - return 22; - } - if (dstSizeInBytes == 0) { - return 34; - } - if (src == 0) { - dst[0] = '\0'; - return 22; - } + pulse.pStreamNamePlayback + PulseAudio only. Sets the stream name for playback. - for (i = 0; i < dstSizeInBytes && src[i] != '\0'; ++i) { - dst[i] = src[i]; - } + pulse.pStreamNameCapture + PulseAudio only. Sets the stream name for capture. - if (i < dstSizeInBytes) { - dst[i] = '\0'; - return 0; - } - dst[0] = '\0'; - return 34; +Once initialized, the device's config is immutable. If you need to change the config you will need to initialize a new device. + +If both `periodSizeInFrames` and `periodSizeInMilliseconds` are set to zero, it will default to `MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY` or +`MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE`, depending on whether or not `performanceProfile` is set to `ma_performance_profile_low_latency` or +`ma_performance_profile_conservative`. + +If you request exclusive mode and the backend does not support it an error will be returned. For robustness, you may want to first try initializing the device +in exclusive mode, and then fall back to shared mode if required. Alternatively you can just request shared mode (the default if you leave it unset in the +config) which is the most reliable option. Some backends do not have a practical way of choosing whether or not the device should be exclusive or not (ALSA, +for example) in which case it just acts as a hint. Unless you have special requirements you should try avoiding exclusive mode as it's intrusive to the user. +Starting with Windows 10, miniaudio will use low-latency shared mode where possible which may make exclusive mode unnecessary. + +After initializing the device it will be in a stopped state. To start it, use `ma_device_start()`. + +When sending or receiving data to/from a device, miniaudio will internally perform a format conversion to convert between the format specified by pConfig and +the format used internally by the backend. If you pass in 0 for the sample format, channel count, sample rate _and_ channel map, data transmission will run on +an optimized pass-through fast path. You can retrieve the format, channel count and sample rate by inspecting the `playback/capture.format`, +`playback/capture.channels` and `sampleRate` members of the device object. + +When compiling for UWP you must ensure you call this function on the main UI thread because the operating system may need to present the user with a message +asking for permissions. Please refer to the official documentation for ActivateAudioInterfaceAsync() for more information. + +ALSA Specific: When initializing the default device, requesting shared mode will try using the "dmix" device for playback and the "dsnoop" device for capture. +If these fail it will try falling back to the "hw" device. + + +Example 1 - Simple Initialization +--------------------------------- +This example shows how to initialize a simple playback default using a standard configuration. If you are just needing to do simple playback from the default +playback device this is usually all you need. + +```c +ma_device_config config = ma_device_config_init(ma_device_type_playback); +config.playback.format = ma_format_f32; +config.playback.channels = 2; +config.sampleRate = 48000; +config.dataCallback = ma_data_callback; +config.pMyUserData = pMyUserData; + +ma_device device; +ma_result result = ma_device_init(NULL, &config, &device); +if (result != MA_SUCCESS) { + // Error } +``` -int ma_strncpy_s(char* dst, size_t dstSizeInBytes, const char* src, size_t count) -{ - size_t maxcount; - size_t i; - if (dst == 0) { - return 22; - } - if (dstSizeInBytes == 0) { - return 34; - } - if (src == 0) { - dst[0] = '\0'; - return 22; - } +Example 2 - Advanced Initialization +----------------------------------- +This example show how you might do some more advanced initialization. In this hypothetical example we want to control the latency by setting the buffer size +and period count. We also want to allow the user to be able to choose which device to output from which means we need a context so we can perform device +enumeration. - maxcount = count; - if (count == ((size_t)-1) || count >= dstSizeInBytes) { /* -1 = _TRUNCATE */ - maxcount = dstSizeInBytes - 1; - } +```c +ma_context context; +ma_result result = ma_context_init(NULL, 0, NULL, &context); +if (result != MA_SUCCESS) { + // Error +} - for (i = 0; i < maxcount && src[i] != '\0'; ++i) { - dst[i] = src[i]; - } +ma_device_info* pPlaybackDeviceInfos; +ma_uint32 playbackDeviceCount; +result = ma_context_get_devices(&context, &pPlaybackDeviceInfos, &playbackDeviceCount, NULL, NULL); +if (result != MA_SUCCESS) { + // Error +} - if (src[i] == '\0' || i == count || count == ((size_t)-1)) { - dst[i] = '\0'; - return 0; - } +// ... choose a device from pPlaybackDeviceInfos ... - dst[0] = '\0'; - return 34; +ma_device_config config = ma_device_config_init(ma_device_type_playback); +config.playback.pDeviceID = pMyChosenDeviceID; // <-- Get this from the `id` member of one of the `ma_device_info` objects returned by ma_context_get_devices(). +config.playback.format = ma_format_f32; +config.playback.channels = 2; +config.sampleRate = 48000; +config.dataCallback = ma_data_callback; +config.pUserData = pMyUserData; +config.periodSizeInMilliseconds = 10; +config.periods = 3; + +ma_device device; +result = ma_device_init(&context, &config, &device); +if (result != MA_SUCCESS) { + // Error } +``` -int ma_strcat_s(char* dst, size_t dstSizeInBytes, const char* src) -{ - char* dstorig; - if (dst == 0) { - return 22; - } - if (dstSizeInBytes == 0) { - return 34; - } - if (src == 0) { - dst[0] = '\0'; - return 22; - } +See Also +-------- +ma_device_config_init() +ma_device_uninit() +ma_device_start() +ma_context_init() +ma_context_get_devices() +ma_context_enumerate_devices() +*/ +ma_result ma_device_init(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice); - dstorig = dst; +/* +Initializes a device without a context, with extra parameters for controlling the configuration of the internal self-managed context. - while (dstSizeInBytes > 0 && dst[0] != '\0') { - dst += 1; - dstSizeInBytes -= 1; - } +This is the same as `ma_device_init()`, only instead of a context being passed in, the parameters from `ma_context_init()` are passed in instead. This function +allows you to configure the internally created context. - if (dstSizeInBytes == 0) { - return 22; /* Unterminated. */ - } +Parameters +---------- +backends (in, optional) + A list of backends to try initializing, in priority order. Can be NULL, in which case it uses default priority order. - while (dstSizeInBytes > 0 && src[0] != '\0') { - *dst++ = *src++; - dstSizeInBytes -= 1; - } +backendCount (in, optional) + The number of items in `backend`. Ignored if `backend` is NULL. - if (dstSizeInBytes > 0) { - dst[0] = '\0'; - } else { - dstorig[0] = '\0'; - return 34; - } +pContextConfig (in, optional) + The context configuration. - return 0; -} +pConfig (in) + A pointer to the device configuration. Cannot be null. See remarks for details. -int ma_strncat_s(char* dst, size_t dstSizeInBytes, const char* src, size_t count) -{ - char* dstorig; +pDevice (out) + A pointer to the device object being initialized. - if (dst == 0) { - return 22; - } - if (dstSizeInBytes == 0) { - return 34; - } - if (src == 0) { - return 22; - } - dstorig = dst; +Return Value +------------ +MA_SUCCESS if successful; any other error code otherwise. + + +Thread Safety +------------- +Unsafe. It is not safe to call this function simultaneously for different devices because some backends depend on and mutate global state. The same applies to +calling this at the same time as `ma_device_uninit()`. + + +Callback Safety +--------------- +Unsafe. It is not safe to call this inside any callback. + + +Remarks +------- +You only need to use this function if you want to configure the context differently to it's defaults. You should never use this function if you want to manage +your own context. + +See the documentation for `ma_context_init()` for information on the different context configuration options. + + +See Also +-------- +ma_device_init() +ma_device_uninit() +ma_device_config_init() +ma_context_init() +*/ +ma_result ma_device_init_ex(const ma_backend backends[], ma_uint32 backendCount, const ma_context_config* pContextConfig, const ma_device_config* pConfig, ma_device* pDevice); + +/* +Uninitializes a device. + +This will explicitly stop the device. You do not need to call `ma_device_stop()` beforehand, but it's harmless if you do. - while (dstSizeInBytes > 0 && dst[0] != '\0') { - dst += 1; - dstSizeInBytes -= 1; - } - if (dstSizeInBytes == 0) { - return 22; /* Unterminated. */ - } +Parameters +---------- +pDevice (in) + A pointer to the device to stop. - if (count == ((size_t)-1)) { /* _TRUNCATE */ - count = dstSizeInBytes - 1; - } +Return Value +------------ +MA_SUCCESS if successful; any other error code otherwise. - while (dstSizeInBytes > 0 && src[0] != '\0' && count > 0) { - *dst++ = *src++; - dstSizeInBytes -= 1; - count -= 1; - } - if (dstSizeInBytes > 0) { - dst[0] = '\0'; - } else { - dstorig[0] = '\0'; - return 34; - } +Thread Safety +------------- +Unsafe. As soon as this API is called the device should be considered undefined. - return 0; -} -int ma_itoa_s(int value, char* dst, size_t dstSizeInBytes, int radix) -{ - int sign; - unsigned int valueU; - char* dstEnd; +Callback Safety +--------------- +Unsafe. It is not safe to call this inside any callback. Doing this will result in a deadlock. - if (dst == NULL || dstSizeInBytes == 0) { - return 22; - } - if (radix < 2 || radix > 36) { - dst[0] = '\0'; - return 22; - } - sign = (value < 0 && radix == 10) ? -1 : 1; /* The negative sign is only used when the base is 10. */ +See Also +-------- +ma_device_init() +ma_device_stop() +*/ +void ma_device_uninit(ma_device* pDevice); - if (value < 0) { - valueU = -value; - } else { - valueU = value; - } +/* +Starts the device. For playback devices this begins playback. For capture devices it begins recording. - dstEnd = dst; - do - { - int remainder = valueU % radix; - if (remainder > 9) { - *dstEnd = (char)((remainder - 10) + 'a'); - } else { - *dstEnd = (char)(remainder + '0'); - } +Use `ma_device_stop()` to stop the device. - dstEnd += 1; - dstSizeInBytes -= 1; - valueU /= radix; - } while (dstSizeInBytes > 0 && valueU > 0); - if (dstSizeInBytes == 0) { - dst[0] = '\0'; - return 22; /* Ran out of room in the output buffer. */ - } +Parameters +---------- +pDevice (in) + A pointer to the device to start. - if (sign < 0) { - *dstEnd++ = '-'; - dstSizeInBytes -= 1; - } - if (dstSizeInBytes == 0) { - dst[0] = '\0'; - return 22; /* Ran out of room in the output buffer. */ - } +Return Value +------------ +MA_SUCCESS if successful; any other error code otherwise. - *dstEnd = '\0'; +Thread Safety +------------- +Safe. It's safe to call this from any thread with the exception of the callback thread. - /* At this point the string will be reversed. */ - dstEnd -= 1; - while (dst < dstEnd) { - char temp = *dst; - *dst = *dstEnd; - *dstEnd = temp; - dst += 1; - dstEnd -= 1; - } +Callback Safety +--------------- +Unsafe. It is not safe to call this inside any callback. - return 0; -} -int ma_strcmp(const char* str1, const char* str2) -{ - if (str1 == str2) return 0; +Remarks +------- +For a playback device, this will retrieve an initial chunk of audio data from the client before returning. The reason for this is to ensure there is valid +audio data in the buffer, which needs to be done before the device begins playback. - /* These checks differ from the standard implementation. It's not important, but I prefer it just for sanity. */ - if (str1 == NULL) return -1; - if (str2 == NULL) return 1; +This API waits until the backend device has been started for real by the worker thread. It also waits on a mutex for thread-safety. - for (;;) { - if (str1[0] == '\0') { - break; - } - if (str1[0] != str2[0]) { - break; - } +Do not call this in any callback. - str1 += 1; - str2 += 1; - } - return ((unsigned char*)str1)[0] - ((unsigned char*)str2)[0]; -} +See Also +-------- +ma_device_stop() +*/ +ma_result ma_device_start(ma_device* pDevice); -int ma_strappend(char* dst, size_t dstSize, const char* srcA, const char* srcB) -{ - int result; +/* +Stops the device. For playback devices this stops playback. For capture devices it stops recording. - result = ma_strncpy_s(dst, dstSize, srcA, (size_t)-1); - if (result != 0) { - return result; - } +Use `ma_device_start()` to start the device again. - result = ma_strncat_s(dst, dstSize, srcB, (size_t)-1); - if (result != 0) { - return result; - } - return result; -} +Parameters +---------- +pDevice (in) + A pointer to the device to stop. -char* ma_copy_string(const char* src) -{ - size_t sz = strlen(src)+1; - char* dst = (char*)ma_malloc(sz); - if (dst == NULL) { - return NULL; - } - ma_strcpy_s(dst, sz, src); +Return Value +------------ +MA_SUCCESS if successful; any other error code otherwise. - return dst; -} +Thread Safety +------------- +Safe. It's safe to call this from any thread with the exception of the callback thread. -/* Thanks to good old Bit Twiddling Hacks for this one: http://graphics.stanford.edu/~seander/bithacks.html#RoundUpPowerOf2 */ -static MA_INLINE unsigned int ma_next_power_of_2(unsigned int x) -{ - x--; - x |= x >> 1; - x |= x >> 2; - x |= x >> 4; - x |= x >> 8; - x |= x >> 16; - x++; - return x; -} +Callback Safety +--------------- +Unsafe. It is not safe to call this inside any callback. Doing this will result in a deadlock. -static MA_INLINE unsigned int ma_prev_power_of_2(unsigned int x) -{ - return ma_next_power_of_2(x) >> 1; -} -static MA_INLINE unsigned int ma_round_to_power_of_2(unsigned int x) -{ - unsigned int prev = ma_prev_power_of_2(x); - unsigned int next = ma_next_power_of_2(x); - if ((next - x) > (x - prev)) { - return prev; - } else { - return next; - } -} +Remarks +------- +This API needs to wait on the worker thread to stop the backend device properly before returning. It also waits on a mutex for thread-safety. In addition, some +backends need to wait for the device to finish playback/recording of the current fragment which can take some time (usually proportionate to the buffer size +that was specified at initialization time). -static MA_INLINE unsigned int ma_count_set_bits(unsigned int x) -{ - unsigned int count = 0; - while (x != 0) { - if (x & 1) { - count += 1; - } - - x = x >> 1; - } - - return count; -} +Backends are required to either pause the stream in-place or drain the buffer if pausing is not possible. The reason for this is that stopping the device and +the resuming it with ma_device_start() (which you might do when your program loses focus) may result in a situation where those samples are never output to the +speakers or received from the microphone which can in turn result in de-syncs. +Do not call this in any callback. +This will be called implicitly by `ma_device_uninit()`. -/* Clamps an f32 sample to -1..1 */ -static MA_INLINE float ma_clip_f32(float x) -{ - if (x < -1) return -1; - if (x > +1) return +1; - return x; -} -static MA_INLINE float ma_mix_f32(float x, float y, float a) -{ - return x*(1-a) + y*a; -} -static MA_INLINE float ma_mix_f32_fast(float x, float y, float a) -{ - float r0 = (y - x); - float r1 = r0*a; - return x + r1; - /*return x + (y - x)*a;*/ -} +See Also +-------- +ma_device_start() +*/ +ma_result ma_device_stop(ma_device* pDevice); -#if defined(MA_SUPPORT_SSE2) -static MA_INLINE __m128 ma_mix_f32_fast__sse2(__m128 x, __m128 y, __m128 a) -{ - return _mm_add_ps(x, _mm_mul_ps(_mm_sub_ps(y, x), a)); -} -#endif -#if defined(MA_SUPPORT_AVX2) -static MA_INLINE __m256 ma_mix_f32_fast__avx2(__m256 x, __m256 y, __m256 a) -{ - return _mm256_add_ps(x, _mm256_mul_ps(_mm256_sub_ps(y, x), a)); -} -#endif -#if defined(MA_SUPPORT_AVX512) -static MA_INLINE __m512 ma_mix_f32_fast__avx512(__m512 x, __m512 y, __m512 a) -{ - return _mm512_add_ps(x, _mm512_mul_ps(_mm512_sub_ps(y, x), a)); -} -#endif -#if defined(MA_SUPPORT_NEON) -static MA_INLINE float32x4_t ma_mix_f32_fast__neon(float32x4_t x, float32x4_t y, float32x4_t a) -{ - return vaddq_f32(x, vmulq_f32(vsubq_f32(y, x), a)); -} -#endif +/* +Determines whether or not the device is started. -static MA_INLINE double ma_mix_f64(double x, double y, double a) -{ - return x*(1-a) + y*a; -} -static MA_INLINE double ma_mix_f64_fast(double x, double y, double a) -{ - return x + (y - x)*a; -} +Parameters +---------- +pDevice (in) + A pointer to the device whose start state is being retrieved. -static MA_INLINE float ma_scale_to_range_f32(float x, float lo, float hi) -{ - return lo + x*(hi-lo); -} + +Return Value +------------ +True if the device is started, false otherwise. -/* -Random Number Generation +Thread Safety +------------- +Safe. If another thread calls `ma_device_start()` or `ma_device_stop()` at this same time as this function is called, there's a very small chance the return +value will be out of sync. -miniaudio uses the LCG random number generation algorithm. This is good enough for audio. -Note that miniaudio's LCG implementation uses global state which is _not_ thread-local. When this is called across -multiple threads, results will be unpredictable. However, it won't crash and results will still be random enough -for miniaudio's purposes. +Callback Safety +--------------- +Safe. This is implemented as a simple accessor. + + +See Also +-------- +ma_device_start() +ma_device_stop() */ -#define MA_LCG_M 2147483647 -#define MA_LCG_A 48271 -#define MA_LCG_C 0 -static ma_int32 g_maLCG = 4321; /* Non-zero initial seed. Use ma_seed() to use an explicit seed. */ +ma_bool32 ma_device_is_started(ma_device* pDevice); -void ma_seed(ma_int32 seed) -{ - g_maLCG = seed; -} +/* +Sets the master volume factor for the device. -ma_int32 ma_rand_s32() -{ - ma_int32 lcg = g_maLCG; - ma_int32 r = (MA_LCG_A * lcg + MA_LCG_C) % MA_LCG_M; - g_maLCG = r; - return r; -} +The volume factor must be between 0 (silence) and 1 (full volume). Use `ma_device_set_master_gain_db()` to use decibel notation, where 0 is full volume and +values less than 0 decreases the volume. -ma_uint32 ma_rand_u32() -{ - return (ma_uint32)ma_rand_s32(); -} -double ma_rand_f64() -{ - return ma_rand_s32() / (double)0x7FFFFFFF; -} +Parameters +---------- +pDevice (in) + A pointer to the device whose volume is being set. -float ma_rand_f32() -{ - return (float)ma_rand_f64(); -} +volume (in) + The new volume factor. Must be within the range of [0, 1]. -float ma_rand_range_f32(float lo, float hi) -{ - return ma_scale_to_range_f32(ma_rand_f32(), lo, hi); -} -ma_int32 ma_rand_range_s32(ma_int32 lo, ma_int32 hi) -{ - if (lo == hi) { - return lo; - } +Return Value +------------ +MA_SUCCESS if the volume was set successfully. +MA_INVALID_ARGS if pDevice is NULL. +MA_INVALID_ARGS if the volume factor is not within the range of [0, 1]. - return lo + ma_rand_u32() / (0xFFFFFFFF / (hi - lo + 1) + 1); -} +Thread Safety +------------- +Safe. This just sets a local member of the device object. -static MA_INLINE float ma_dither_f32_rectangle(float ditherMin, float ditherMax) -{ - return ma_rand_range_f32(ditherMin, ditherMax); -} -static MA_INLINE float ma_dither_f32_triangle(float ditherMin, float ditherMax) -{ - float a = ma_rand_range_f32(ditherMin, 0); - float b = ma_rand_range_f32(0, ditherMax); - return a + b; -} +Callback Safety +--------------- +Safe. If you set the volume in the data callback, that data written to the output buffer will have the new volume applied. -static MA_INLINE float ma_dither_f32(ma_dither_mode ditherMode, float ditherMin, float ditherMax) -{ - if (ditherMode == ma_dither_mode_rectangle) { - return ma_dither_f32_rectangle(ditherMin, ditherMax); - } - if (ditherMode == ma_dither_mode_triangle) { - return ma_dither_f32_triangle(ditherMin, ditherMax); - } - return 0; -} +Remarks +------- +This applies the volume factor across all channels. -static MA_INLINE ma_int32 ma_dither_s32(ma_dither_mode ditherMode, ma_int32 ditherMin, ma_int32 ditherMax) -{ - if (ditherMode == ma_dither_mode_rectangle) { - ma_int32 a = ma_rand_range_s32(ditherMin, ditherMax); - return a; - } - if (ditherMode == ma_dither_mode_triangle) { - ma_int32 a = ma_rand_range_s32(ditherMin, 0); - ma_int32 b = ma_rand_range_s32(0, ditherMax); - return a + b; - } +This does not change the operating system's volume. It only affects the volume for the given `ma_device` object's audio stream. - return 0; -} +See Also +-------- +ma_device_get_master_volume() +ma_device_set_master_volume_gain_db() +ma_device_get_master_volume_gain_db() +*/ +ma_result ma_device_set_master_volume(ma_device* pDevice, float volume); /* -Splits a buffer into parts of equal length and of the given alignment. The returned size of the split buffers will be a -multiple of the alignment. The alignment must be a power of 2. -*/ -void ma_split_buffer(void* pBuffer, size_t bufferSize, size_t splitCount, size_t alignment, void** ppBuffersOut, size_t* pSplitSizeOut) -{ - ma_uintptr pBufferUnaligned; - ma_uintptr pBufferAligned; - size_t unalignedBytes; - size_t splitSize; +Retrieves the master volume factor for the device. - if (pSplitSizeOut) { - *pSplitSizeOut = 0; - } - if (pBuffer == NULL || bufferSize == 0 || splitCount == 0) { - return; - } +Parameters +---------- +pDevice (in) + A pointer to the device whose volume factor is being retrieved. - if (alignment == 0) { - alignment = 1; - } +pVolume (in) + A pointer to the variable that will receive the volume factor. The returned value will be in the range of [0, 1]. - pBufferUnaligned = (ma_uintptr)pBuffer; - pBufferAligned = (pBufferUnaligned + (alignment-1)) & ~(alignment-1); - unalignedBytes = (size_t)(pBufferAligned - pBufferUnaligned); - splitSize = 0; - if (bufferSize >= unalignedBytes) { - splitSize = (bufferSize - unalignedBytes) / splitCount; - splitSize = splitSize & ~(alignment-1); - } +Return Value +------------ +MA_SUCCESS if successful. +MA_INVALID_ARGS if pDevice is NULL. +MA_INVALID_ARGS if pVolume is NULL. - if (ppBuffersOut != NULL) { - size_t i; - for (i = 0; i < splitCount; ++i) { - ppBuffersOut[i] = (ma_uint8*)(pBufferAligned + (splitSize*i)); - } - } - if (pSplitSizeOut) { - *pSplitSizeOut = splitSize; - } -} +Thread Safety +------------- +Safe. This just a simple member retrieval. -/****************************************************************************** +Callback Safety +--------------- +Safe. -Atomics -******************************************************************************/ -#if defined(__clang__) - #if defined(__has_builtin) - #if __has_builtin(__sync_swap) - #define MA_HAS_SYNC_SWAP - #endif - #endif -#elif defined(__GNUC__) - #if __GNUC__ > 4 || (__GNUC__ == 4 && __GNUC__ >= 7) - #define MA_HAS_GNUC_ATOMICS - #endif -#endif +Remarks +------- +If an error occurs, `*pVolume` will be set to 0. -#if defined(_WIN32) && !defined(__GNUC__) && !defined(__clang__) -#define ma_memory_barrier() MemoryBarrier() -#define ma_atomic_exchange_32(a, b) InterlockedExchange((LONG*)a, (LONG)b) -#define ma_atomic_exchange_64(a, b) InterlockedExchange64((LONGLONG*)a, (LONGLONG)b) -#define ma_atomic_increment_32(a) InterlockedIncrement((LONG*)a) -#define ma_atomic_decrement_32(a) InterlockedDecrement((LONG*)a) -#else -#define ma_memory_barrier() __sync_synchronize() -#if defined(MA_HAS_SYNC_SWAP) - #define ma_atomic_exchange_32(a, b) __sync_swap(a, b) - #define ma_atomic_exchange_64(a, b) __sync_swap(a, b) -#elif defined(MA_HAS_GNUC_ATOMICS) - #define ma_atomic_exchange_32(a, b) (void)__atomic_exchange_n(a, b, __ATOMIC_ACQ_REL) - #define ma_atomic_exchange_64(a, b) (void)__atomic_exchange_n(a, b, __ATOMIC_ACQ_REL) -#else - #define ma_atomic_exchange_32(a, b) __sync_synchronize(); (void)__sync_lock_test_and_set(a, b) - #define ma_atomic_exchange_64(a, b) __sync_synchronize(); (void)__sync_lock_test_and_set(a, b) -#endif -#define ma_atomic_increment_32(a) __sync_add_and_fetch(a, 1) -#define ma_atomic_decrement_32(a) __sync_sub_and_fetch(a, 1) -#endif -#ifdef MA_64BIT -#define ma_atomic_exchange_ptr ma_atomic_exchange_64 -#endif -#ifdef MA_32BIT -#define ma_atomic_exchange_ptr ma_atomic_exchange_32 -#endif +See Also +-------- +ma_device_set_master_volume() +ma_device_set_master_volume_gain_db() +ma_device_get_master_volume_gain_db() +*/ +ma_result ma_device_get_master_volume(ma_device* pDevice, float* pVolume); +/* +Sets the master volume for the device as gain in decibels. -ma_uint32 ma_get_standard_sample_rate_priority_index(ma_uint32 sampleRate) /* Lower = higher priority */ -{ - ma_uint32 i; - for (i = 0; i < ma_countof(g_maStandardSampleRatePriorities); ++i) { - if (g_maStandardSampleRatePriorities[i] == sampleRate) { - return i; - } - } +A gain of 0 is full volume, whereas a gain of < 0 will decrease the volume. - return (ma_uint32)-1; -} -ma_uint64 ma_calculate_frame_count_after_src(ma_uint32 sampleRateOut, ma_uint32 sampleRateIn, ma_uint64 frameCountIn) -{ - double srcRatio = (double)sampleRateOut / sampleRateIn; - double frameCountOutF = (ma_int64)frameCountIn * srcRatio; /* Cast to int64 required for VC6. */ - ma_uint64 frameCountOut = (ma_uint64)frameCountOutF; +Parameters +---------- +pDevice (in) + A pointer to the device whose gain is being set. - /* If the output frame count is fractional, make sure we add an extra frame to ensure there's enough room for that last sample. */ - if ((frameCountOutF - (ma_int64)frameCountOut) > 0.0) { - frameCountOut += 1; - } +gainDB (in) + The new volume as gain in decibels. Must be less than or equal to 0, where 0 is full volume and anything less than 0 decreases the volume. - return frameCountOut; -} + +Return Value +------------ +MA_SUCCESS if the volume was set successfully. +MA_INVALID_ARGS if pDevice is NULL. +MA_INVALID_ARGS if the gain is > 0. -/************************************************************************************************************************************************************ -************************************************************************************************************************************************************* +Thread Safety +------------- +Safe. This just sets a local member of the device object. -DEVICE I/O -========== -************************************************************************************************************************************************************* -************************************************************************************************************************************************************/ -#ifndef MA_NO_DEVICE_IO -/* -Unfortunately using runtime linking for pthreads causes problems. This has occurred for me when testing on FreeBSD. When -using runtime linking, deadlocks can occur (for me it happens when loading data from fread()). It turns out that doing -compile-time linking fixes this. I'm not sure why this happens, but the safest way I can think of to fix this is to simply -disable runtime linking by default. To enable runtime linking, #define this before the implementation of this file. I am -not officially supporting this, but I'm leaving it here in case it's useful for somebody, somewhere. -*/ -/*#define MA_USE_RUNTIME_LINKING_FOR_PTHREAD*/ +Callback Safety +--------------- +Safe. If you set the volume in the data callback, that data written to the output buffer will have the new volume applied. -/* Disable run-time linking on certain backends. */ -#ifndef MA_NO_RUNTIME_LINKING - #if defined(MA_ANDROID) || defined(MA_EMSCRIPTEN) - #define MA_NO_RUNTIME_LINKING - #endif -#endif -/* -Check if we have the necessary development packages for each backend at the top so we can use this to determine whether or not -certain unused functions and variables can be excluded from the build to avoid warnings. +Remarks +------- +This applies the gain across all channels. + +This does not change the operating system's volume. It only affects the volume for the given `ma_device` object's audio stream. + + +See Also +-------- +ma_device_get_master_volume_gain_db() +ma_device_set_master_volume() +ma_device_get_master_volume() */ -#ifdef MA_ENABLE_WASAPI - #define MA_HAS_WASAPI /* Every compiler should support WASAPI */ -#endif -#ifdef MA_ENABLE_DSOUND - #define MA_HAS_DSOUND /* Every compiler should support DirectSound. */ -#endif -#ifdef MA_ENABLE_WINMM - #define MA_HAS_WINMM /* Every compiler I'm aware of supports WinMM. */ -#endif -#ifdef MA_ENABLE_ALSA - #define MA_HAS_ALSA - #ifdef MA_NO_RUNTIME_LINKING - #ifdef __has_include - #if !__has_include() - #undef MA_HAS_ALSA - #endif - #endif - #endif -#endif -#ifdef MA_ENABLE_PULSEAUDIO - #define MA_HAS_PULSEAUDIO - #ifdef MA_NO_RUNTIME_LINKING - #ifdef __has_include - #if !__has_include() - #undef MA_HAS_PULSEAUDIO - #endif - #endif - #endif -#endif -#ifdef MA_ENABLE_JACK - #define MA_HAS_JACK - #ifdef MA_NO_RUNTIME_LINKING - #ifdef __has_include - #if !__has_include() - #undef MA_HAS_JACK - #endif - #endif - #endif -#endif -#ifdef MA_ENABLE_COREAUDIO - #define MA_HAS_COREAUDIO -#endif -#ifdef MA_ENABLE_SNDIO - #define MA_HAS_SNDIO -#endif -#ifdef MA_ENABLE_AUDIO4 - #define MA_HAS_AUDIO4 -#endif -#ifdef MA_ENABLE_OSS - #define MA_HAS_OSS -#endif -#ifdef MA_ENABLE_AAUDIO - #define MA_HAS_AAUDIO -#endif -#ifdef MA_ENABLE_OPENSL - #define MA_HAS_OPENSL -#endif -#ifdef MA_ENABLE_WEBAUDIO - #define MA_HAS_WEBAUDIO -#endif -#ifdef MA_ENABLE_NULL - #define MA_HAS_NULL /* Everything supports the null backend. */ -#endif +ma_result ma_device_set_master_gain_db(ma_device* pDevice, float gainDB); + +/* +Retrieves the master gain in decibels. -const char* ma_get_backend_name(ma_backend backend) -{ - switch (backend) - { - case ma_backend_wasapi: return "WASAPI"; - case ma_backend_dsound: return "DirectSound"; - case ma_backend_winmm: return "WinMM"; - case ma_backend_coreaudio: return "Core Audio"; - case ma_backend_sndio: return "sndio"; - case ma_backend_audio4: return "audio(4)"; - case ma_backend_oss: return "OSS"; - case ma_backend_pulseaudio: return "PulseAudio"; - case ma_backend_alsa: return "ALSA"; - case ma_backend_jack: return "JACK"; - case ma_backend_aaudio: return "AAudio"; - case ma_backend_opensl: return "OpenSL|ES"; - case ma_backend_webaudio: return "Web Audio"; - case ma_backend_null: return "Null"; - default: return "Unknown"; - } -} -ma_bool32 ma_is_loopback_supported(ma_backend backend) -{ - switch (backend) - { - case ma_backend_wasapi: return MA_TRUE; - case ma_backend_dsound: return MA_FALSE; - case ma_backend_winmm: return MA_FALSE; - case ma_backend_coreaudio: return MA_FALSE; - case ma_backend_sndio: return MA_FALSE; - case ma_backend_audio4: return MA_FALSE; - case ma_backend_oss: return MA_FALSE; - case ma_backend_pulseaudio: return MA_FALSE; - case ma_backend_alsa: return MA_FALSE; - case ma_backend_jack: return MA_FALSE; - case ma_backend_aaudio: return MA_FALSE; - case ma_backend_opensl: return MA_FALSE; - case ma_backend_webaudio: return MA_FALSE; - case ma_backend_null: return MA_FALSE; - default: return MA_FALSE; - } -} +Parameters +---------- +pDevice (in) + A pointer to the device whose gain is being retrieved. +pGainDB (in) + A pointer to the variable that will receive the gain in decibels. The returned value will be <= 0. -#ifdef MA_WIN32 - #define MA_THREADCALL WINAPI - typedef unsigned long ma_thread_result; -#else - #define MA_THREADCALL - typedef void* ma_thread_result; -#endif -typedef ma_thread_result (MA_THREADCALL * ma_thread_entry_proc)(void* pData); +Return Value +------------ +MA_SUCCESS if successful. +MA_INVALID_ARGS if pDevice is NULL. +MA_INVALID_ARGS if pGainDB is NULL. -#ifdef MA_WIN32 -typedef HRESULT (WINAPI * MA_PFN_CoInitializeEx)(LPVOID pvReserved, DWORD dwCoInit); -typedef void (WINAPI * MA_PFN_CoUninitialize)(); -typedef HRESULT (WINAPI * MA_PFN_CoCreateInstance)(REFCLSID rclsid, LPUNKNOWN pUnkOuter, DWORD dwClsContext, REFIID riid, LPVOID *ppv); -typedef void (WINAPI * MA_PFN_CoTaskMemFree)(LPVOID pv); -typedef HRESULT (WINAPI * MA_PFN_PropVariantClear)(PROPVARIANT *pvar); -typedef int (WINAPI * MA_PFN_StringFromGUID2)(const GUID* const rguid, LPOLESTR lpsz, int cchMax); -typedef HWND (WINAPI * MA_PFN_GetForegroundWindow)(); -typedef HWND (WINAPI * MA_PFN_GetDesktopWindow)(); +Thread Safety +------------- +Safe. This just a simple member retrieval. -/* Microsoft documents these APIs as returning LSTATUS, but the Win32 API shipping with some compilers do not define it. It's just a LONG. */ -typedef LONG (WINAPI * MA_PFN_RegOpenKeyExA)(HKEY hKey, LPCSTR lpSubKey, DWORD ulOptions, REGSAM samDesired, PHKEY phkResult); -typedef LONG (WINAPI * MA_PFN_RegCloseKey)(HKEY hKey); -typedef LONG (WINAPI * MA_PFN_RegQueryValueExA)(HKEY hKey, LPCSTR lpValueName, LPDWORD lpReserved, LPDWORD lpType, LPBYTE lpData, LPDWORD lpcbData); -#endif +Callback Safety +--------------- +Safe. -#define MA_STATE_UNINITIALIZED 0 -#define MA_STATE_STOPPED 1 /* The device's default state after initialization. */ -#define MA_STATE_STARTED 2 /* The worker thread is in it's main loop waiting for the driver to request or deliver audio data. */ -#define MA_STATE_STARTING 3 /* Transitioning from a stopped state to started. */ -#define MA_STATE_STOPPING 4 /* Transitioning from a started state to stopped. */ -#define MA_DEFAULT_PLAYBACK_DEVICE_NAME "Default Playback Device" -#define MA_DEFAULT_CAPTURE_DEVICE_NAME "Default Capture Device" +Remarks +------- +If an error occurs, `*pGainDB` will be set to 0. -const char* ma_log_level_to_string(ma_uint32 logLevel) -{ - switch (logLevel) - { - case MA_LOG_LEVEL_VERBOSE: return ""; - case MA_LOG_LEVEL_INFO: return "INFO"; - case MA_LOG_LEVEL_WARNING: return "WARNING"; - case MA_LOG_LEVEL_ERROR: return "ERROR"; - default: return "ERROR"; - } -} +See Also +-------- +ma_device_set_master_volume_gain_db() +ma_device_set_master_volume() +ma_device_get_master_volume() +*/ +ma_result ma_device_get_master_gain_db(ma_device* pDevice, float* pGainDB); -/* Posts a log message. */ -void ma_log(ma_context* pContext, ma_device* pDevice, ma_uint32 logLevel, const char* message) -{ - if (pContext == NULL) { - return; - } - -#if defined(MA_LOG_LEVEL) - if (logLevel <= MA_LOG_LEVEL) { - ma_log_proc onLog; - #if defined(MA_DEBUG_OUTPUT) - if (logLevel <= MA_LOG_LEVEL) { - printf("%s: %s\n", ma_log_level_to_string(logLevel), message); - } - #endif - - onLog = pContext->logCallback; - if (onLog) { - onLog(pContext, pDevice, logLevel, message); - } - } -#endif -} -/* Posts an log message. Throw a breakpoint in here if you're needing to debug. The return value is always "resultCode". */ -ma_result ma_context_post_error(ma_context* pContext, ma_device* pDevice, ma_uint32 logLevel, const char* message, ma_result resultCode) -{ - /* Derive the context from the device if necessary. */ - if (pContext == NULL) { - if (pDevice != NULL) { - pContext = pDevice->pContext; - } - } +/************************************************************************************************************************************************************ - ma_log(pContext, pDevice, logLevel, message); - return resultCode; -} +Utiltities -ma_result ma_post_error(ma_device* pDevice, ma_uint32 logLevel, const char* message, ma_result resultCode) -{ - return ma_context_post_error(NULL, pDevice, logLevel, message, resultCode); -} +************************************************************************************************************************************************************/ +/* +Creates a mutex. -/******************************************************************************* +A mutex must be created from a valid context. A mutex is initially unlocked. +*/ +ma_result ma_mutex_init(ma_context* pContext, ma_mutex* pMutex); -Timing +/* +Deletes a mutex. +*/ +void ma_mutex_uninit(ma_mutex* pMutex); -*******************************************************************************/ -#ifdef MA_WIN32 -LARGE_INTEGER g_ma_TimerFrequency = {{0}}; -void ma_timer_init(ma_timer* pTimer) -{ - LARGE_INTEGER counter; +/* +Locks a mutex with an infinite timeout. +*/ +void ma_mutex_lock(ma_mutex* pMutex); - if (g_ma_TimerFrequency.QuadPart == 0) { - QueryPerformanceFrequency(&g_ma_TimerFrequency); - } +/* +Unlocks a mutex. +*/ +void ma_mutex_unlock(ma_mutex* pMutex); - QueryPerformanceCounter(&counter); - pTimer->counter = counter.QuadPart; -} -double ma_timer_get_time_in_seconds(ma_timer* pTimer) -{ - LARGE_INTEGER counter; - if (!QueryPerformanceCounter(&counter)) { - return 0; - } +/* +Retrieves a friendly name for a backend. +*/ +const char* ma_get_backend_name(ma_backend backend); - return (double)(counter.QuadPart - pTimer->counter) / g_ma_TimerFrequency.QuadPart; -} -#elif defined(MA_APPLE) && (__MAC_OS_X_VERSION_MIN_REQUIRED < 101200) -ma_uint64 g_ma_TimerFrequency = 0; -void ma_timer_init(ma_timer* pTimer) -{ - mach_timebase_info_data_t baseTime; - mach_timebase_info(&baseTime); - g_ma_TimerFrequency = (baseTime.denom * 1e9) / baseTime.numer; +/* +Determines whether or not loopback mode is support by a backend. +*/ +ma_bool32 ma_is_loopback_supported(ma_backend backend); - pTimer->counter = mach_absolute_time(); -} -double ma_timer_get_time_in_seconds(ma_timer* pTimer) -{ - ma_uint64 newTimeCounter = mach_absolute_time(); - ma_uint64 oldTimeCounter = pTimer->counter; +/* +Adjust buffer size based on a scaling factor. - return (newTimeCounter - oldTimeCounter) / g_ma_TimerFrequency; -} -#elif defined(MA_EMSCRIPTEN) -void ma_timer_init(ma_timer* pTimer) -{ - pTimer->counterD = emscripten_get_now(); -} +This just multiplies the base size by the scaling factor, making sure it's a size of at least 1. +*/ +ma_uint32 ma_scale_buffer_size(ma_uint32 baseBufferSize, float scale); -double ma_timer_get_time_in_seconds(ma_timer* pTimer) -{ - return (emscripten_get_now() - pTimer->counterD) / 1000; /* Emscripten is in milliseconds. */ -} -#else -#if _POSIX_C_SOURCE >= 199309L -#if defined(CLOCK_MONOTONIC) - #define MA_CLOCK_ID CLOCK_MONOTONIC -#else - #define MA_CLOCK_ID CLOCK_REALTIME -#endif +/* +Calculates a buffer size in milliseconds from the specified number of frames and sample rate. +*/ +ma_uint32 ma_calculate_buffer_size_in_milliseconds_from_frames(ma_uint32 bufferSizeInFrames, ma_uint32 sampleRate); -void ma_timer_init(ma_timer* pTimer) -{ - struct timespec newTime; - clock_gettime(MA_CLOCK_ID, &newTime); +/* +Calculates a buffer size in frames from the specified number of milliseconds and sample rate. +*/ +ma_uint32 ma_calculate_buffer_size_in_frames_from_milliseconds(ma_uint32 bufferSizeInMilliseconds, ma_uint32 sampleRate); - pTimer->counter = (newTime.tv_sec * 1000000000) + newTime.tv_nsec; -} +/* +Copies silent frames into the given buffer. +*/ +void ma_zero_pcm_frames(void* p, ma_uint32 frameCount, ma_format format, ma_uint32 channels); -double ma_timer_get_time_in_seconds(ma_timer* pTimer) -{ - ma_uint64 newTimeCounter; - ma_uint64 oldTimeCounter; +/* +Clips f32 samples. +*/ +void ma_clip_samples_f32(float* p, ma_uint32 sampleCount); +MA_INLINE void ma_clip_pcm_frames_f32(float* p, ma_uint32 frameCount, ma_uint32 channels) { ma_clip_samples_f32(p, frameCount*channels); } + +/* +Helper for applying a volume factor to samples. + +Note that the source and destination buffers can be the same, in which case it'll perform the operation in-place. +*/ +void ma_copy_and_apply_volume_factor_u8(ma_uint8* pSamplesOut, const ma_uint8* pSamplesIn, ma_uint32 sampleCount, float factor); +void ma_copy_and_apply_volume_factor_s16(ma_int16* pSamplesOut, const ma_int16* pSamplesIn, ma_uint32 sampleCount, float factor); +void ma_copy_and_apply_volume_factor_s24(void* pSamplesOut, const void* pSamplesIn, ma_uint32 sampleCount, float factor); +void ma_copy_and_apply_volume_factor_s32(ma_int32* pSamplesOut, const ma_int32* pSamplesIn, ma_uint32 sampleCount, float factor); +void ma_copy_and_apply_volume_factor_f32(float* pSamplesOut, const float* pSamplesIn, ma_uint32 sampleCount, float factor); + +void ma_apply_volume_factor_u8(ma_uint8* pSamples, ma_uint32 sampleCount, float factor); +void ma_apply_volume_factor_s16(ma_int16* pSamples, ma_uint32 sampleCount, float factor); +void ma_apply_volume_factor_s24(void* pSamples, ma_uint32 sampleCount, float factor); +void ma_apply_volume_factor_s32(ma_int32* pSamples, ma_uint32 sampleCount, float factor); +void ma_apply_volume_factor_f32(float* pSamples, ma_uint32 sampleCount, float factor); + +void ma_copy_and_apply_volume_factor_pcm_frames_u8(ma_uint8* pPCMFramesOut, const ma_uint8* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor); +void ma_copy_and_apply_volume_factor_pcm_frames_s16(ma_int16* pPCMFramesOut, const ma_int16* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor); +void ma_copy_and_apply_volume_factor_pcm_frames_s24(void* pPCMFramesOut, const void* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor); +void ma_copy_and_apply_volume_factor_pcm_frames_s32(ma_int32* pPCMFramesOut, const ma_int32* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor); +void ma_copy_and_apply_volume_factor_pcm_frames_f32(float* pPCMFramesOut, const float* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor); +void ma_copy_and_apply_volume_factor_pcm_frames(void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount, ma_format format, ma_uint32 channels, float factor); - struct timespec newTime; - clock_gettime(MA_CLOCK_ID, &newTime); +void ma_apply_volume_factor_pcm_frames_u8(ma_uint8* pFrames, ma_uint32 frameCount, ma_uint32 channels, float factor); +void ma_apply_volume_factor_pcm_frames_s16(ma_int16* pFrames, ma_uint32 frameCount, ma_uint32 channels, float factor); +void ma_apply_volume_factor_pcm_frames_s24(void* pFrames, ma_uint32 frameCount, ma_uint32 channels, float factor); +void ma_apply_volume_factor_pcm_frames_s32(ma_int32* pFrames, ma_uint32 frameCount, ma_uint32 channels, float factor); +void ma_apply_volume_factor_pcm_frames_f32(float* pFrames, ma_uint32 frameCount, ma_uint32 channels, float factor); +void ma_apply_volume_factor_pcm_frames(void* pFrames, ma_uint32 frameCount, ma_format format, ma_uint32 channels, float factor); - newTimeCounter = (newTime.tv_sec * 1000000000) + newTime.tv_nsec; - oldTimeCounter = pTimer->counter; - return (newTimeCounter - oldTimeCounter) / 1000000000.0; -} -#else -void ma_timer_init(ma_timer* pTimer) -{ - struct timeval newTime; - gettimeofday(&newTime, NULL); +/* +Helper for converting a linear factor to gain in decibels. +*/ +float ma_factor_to_gain_db(float factor); - pTimer->counter = (newTime.tv_sec * 1000000) + newTime.tv_usec; -} +/* +Helper for converting gain in decibels to a linear factor. +*/ +float ma_gain_db_to_factor(float gain); -double ma_timer_get_time_in_seconds(ma_timer* pTimer) -{ - ma_uint64 newTimeCounter; - ma_uint64 oldTimeCounter; +#endif /* MA_NO_DEVICE_IO */ - struct timeval newTime; - gettimeofday(&newTime, NULL); - newTimeCounter = (newTime.tv_sec * 1000000) + newTime.tv_usec; - oldTimeCounter = pTimer->counter; - return (newTimeCounter - oldTimeCounter) / 1000000.0; -} -#endif -#endif +/************************************************************************************************************************************************************ -/******************************************************************************* +Decoding +======== -Dynamic Linking +Decoders are independent of the main device API. Decoding APIs can be called freely inside the device's data callback, but they are not thread safe unless +you do your own synchronization. -*******************************************************************************/ -ma_handle ma_dlopen(ma_context* pContext, const char* filename) -{ - ma_handle handle; +************************************************************************************************************************************************************/ +#ifndef MA_NO_DECODING -#if MA_LOG_LEVEL >= MA_LOG_LEVEL_VERBOSE - if (pContext != NULL) { - char message[256]; - ma_strappend(message, sizeof(message), "Loading library: ", filename); - ma_log(pContext, NULL, MA_LOG_LEVEL_VERBOSE, message); - } -#endif +typedef struct ma_decoder ma_decoder; -#ifdef _WIN32 -#ifdef MA_WIN32_DESKTOP - handle = (ma_handle)LoadLibraryA(filename); -#else - /* *sigh* It appears there is no ANSI version of LoadPackagedLibrary()... */ - WCHAR filenameW[4096]; - if (MultiByteToWideChar(CP_UTF8, 0, filename, -1, filenameW, sizeof(filenameW)) == 0) { - handle = NULL; - } else { - handle = (ma_handle)LoadPackagedLibrary(filenameW, 0); - } -#endif -#else - handle = (ma_handle)dlopen(filename, RTLD_NOW); -#endif +typedef enum +{ + ma_seek_origin_start, + ma_seek_origin_current +} ma_seek_origin; - /* - I'm not considering failure to load a library an error nor a warning because seamlessly falling through to a lower-priority - backend is a deliberate design choice. Instead I'm logging it as an informational message. - */ -#if MA_LOG_LEVEL >= MA_LOG_LEVEL_INFO - if (handle == NULL) { - char message[256]; - ma_strappend(message, sizeof(message), "Failed to load library: ", filename); - ma_log(pContext, NULL, MA_LOG_LEVEL_INFO, message); - } -#endif +typedef size_t (* ma_decoder_read_proc) (ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead); /* Returns the number of bytes read. */ +typedef ma_bool32 (* ma_decoder_seek_proc) (ma_decoder* pDecoder, int byteOffset, ma_seek_origin origin); +typedef ma_uint64 (* ma_decoder_read_pcm_frames_proc) (ma_decoder* pDecoder, void* pFramesOut, ma_uint64 frameCount); /* Returns the number of frames read. Output data is in internal format. */ +typedef ma_result (* ma_decoder_seek_to_pcm_frame_proc) (ma_decoder* pDecoder, ma_uint64 frameIndex); +typedef ma_result (* ma_decoder_uninit_proc) (ma_decoder* pDecoder); +typedef ma_uint64 (* ma_decoder_get_length_in_pcm_frames_proc)(ma_decoder* pDecoder); - (void)pContext; /* It's possible for pContext to be unused. */ - return handle; -} +typedef struct +{ + ma_format format; /* Set to 0 or ma_format_unknown to use the stream's internal format. */ + ma_uint32 channels; /* Set to 0 to use the stream's internal channels. */ + ma_uint32 sampleRate; /* Set to 0 to use the stream's internal sample rate. */ + ma_channel channelMap[MA_MAX_CHANNELS]; + ma_channel_mix_mode channelMixMode; + ma_dither_mode ditherMode; + struct + { + ma_resample_algorithm algorithm; + struct + { + ma_uint32 lpfCount; + } linear; + struct + { + int quality; + } speex; + } resampling; + ma_allocation_callbacks allocationCallbacks; +} ma_decoder_config; -void ma_dlclose(ma_context* pContext, ma_handle handle) +struct ma_decoder { -#ifdef _WIN32 - FreeLibrary((HMODULE)handle); -#else - dlclose((void*)handle); -#endif + ma_decoder_read_proc onRead; + ma_decoder_seek_proc onSeek; + void* pUserData; + ma_uint64 readPointer; /* Used for returning back to a previous position after analysing the stream or whatnot. */ + ma_format internalFormat; + ma_uint32 internalChannels; + ma_uint32 internalSampleRate; + ma_channel internalChannelMap[MA_MAX_CHANNELS]; + ma_format outputFormat; + ma_uint32 outputChannels; + ma_uint32 outputSampleRate; + ma_channel outputChannelMap[MA_MAX_CHANNELS]; + ma_data_converter converter; /* <-- Data conversion is achieved by running frames through this. */ + ma_allocation_callbacks allocationCallbacks; + ma_decoder_read_pcm_frames_proc onReadPCMFrames; + ma_decoder_seek_to_pcm_frame_proc onSeekToPCMFrame; + ma_decoder_uninit_proc onUninit; + ma_decoder_get_length_in_pcm_frames_proc onGetLengthInPCMFrames; + void* pInternalDecoder; /* <-- The drwav/drflac/stb_vorbis/etc. objects. */ + struct + { + const ma_uint8* pData; + size_t dataSize; + size_t currentReadPos; + } memory; /* Only used for decoders that were opened against a block of memory. */ +}; - (void)pContext; -} +ma_decoder_config ma_decoder_config_init(ma_format outputFormat, ma_uint32 outputChannels, ma_uint32 outputSampleRate); -ma_proc ma_dlsym(ma_context* pContext, ma_handle handle, const char* symbol) -{ - ma_proc proc; +ma_result ma_decoder_init(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +ma_result ma_decoder_init_wav(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +ma_result ma_decoder_init_flac(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +ma_result ma_decoder_init_vorbis(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +ma_result ma_decoder_init_mp3(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +ma_result ma_decoder_init_raw(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfigIn, const ma_decoder_config* pConfigOut, ma_decoder* pDecoder); -#if MA_LOG_LEVEL >= MA_LOG_LEVEL_VERBOSE - if (pContext != NULL) { - char message[256]; - ma_strappend(message, sizeof(message), "Loading symbol: ", symbol); - ma_log(pContext, NULL, MA_LOG_LEVEL_VERBOSE, message); - } -#endif +ma_result ma_decoder_init_memory(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +ma_result ma_decoder_init_memory_wav(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +ma_result ma_decoder_init_memory_flac(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +ma_result ma_decoder_init_memory_vorbis(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +ma_result ma_decoder_init_memory_mp3(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +ma_result ma_decoder_init_memory_raw(const void* pData, size_t dataSize, const ma_decoder_config* pConfigIn, const ma_decoder_config* pConfigOut, ma_decoder* pDecoder); -#ifdef _WIN32 - proc = (ma_proc)GetProcAddress((HMODULE)handle, symbol); -#else -#if defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)) - #pragma GCC diagnostic push - #pragma GCC diagnostic ignored "-Wpedantic" -#endif - proc = (ma_proc)dlsym((void*)handle, symbol); -#if defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)) - #pragma GCC diagnostic pop -#endif -#endif +#ifndef MA_NO_STDIO +ma_result ma_decoder_init_file(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +ma_result ma_decoder_init_file_wav(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +ma_result ma_decoder_init_file_flac(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +ma_result ma_decoder_init_file_vorbis(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +ma_result ma_decoder_init_file_mp3(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); -#if MA_LOG_LEVEL >= MA_LOG_LEVEL_WARNING - if (handle == NULL) { - char message[256]; - ma_strappend(message, sizeof(message), "Failed to load symbol: ", symbol); - ma_log(pContext, NULL, MA_LOG_LEVEL_WARNING, message); - } +ma_result ma_decoder_init_file_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +ma_result ma_decoder_init_file_wav_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +ma_result ma_decoder_init_file_flac_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +ma_result ma_decoder_init_file_vorbis_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); +ma_result ma_decoder_init_file_mp3_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder); #endif - (void)pContext; /* It's possible for pContext to be unused. */ - return proc; -} +ma_result ma_decoder_uninit(ma_decoder* pDecoder); +/* +Retrieves the length of the decoder in PCM frames. -/******************************************************************************* +Do not call this on streams of an undefined length, such as internet radio. -Threading +If the length is unknown or an error occurs, 0 will be returned. -*******************************************************************************/ -#ifdef MA_WIN32 -int ma_thread_priority_to_win32(ma_thread_priority priority) -{ - switch (priority) { - case ma_thread_priority_idle: return THREAD_PRIORITY_IDLE; - case ma_thread_priority_lowest: return THREAD_PRIORITY_LOWEST; - case ma_thread_priority_low: return THREAD_PRIORITY_BELOW_NORMAL; - case ma_thread_priority_normal: return THREAD_PRIORITY_NORMAL; - case ma_thread_priority_high: return THREAD_PRIORITY_ABOVE_NORMAL; - case ma_thread_priority_highest: return THREAD_PRIORITY_HIGHEST; - case ma_thread_priority_realtime: return THREAD_PRIORITY_TIME_CRITICAL; - default: return THREAD_PRIORITY_NORMAL; - } -} +This will always return 0 for Vorbis decoders. This is due to a limitation with stb_vorbis in push mode which is what miniaudio +uses internally. -ma_result ma_thread_create__win32(ma_context* pContext, ma_thread* pThread, ma_thread_entry_proc entryProc, void* pData) -{ - pThread->win32.hThread = CreateThread(NULL, 0, entryProc, pData, 0, NULL); - if (pThread->win32.hThread == NULL) { - return MA_FAILED_TO_CREATE_THREAD; - } +For MP3's, this will decode the entire file. Do not call this in time critical scenarios. - SetThreadPriority((HANDLE)pThread->win32.hThread, ma_thread_priority_to_win32(pContext->threadPriority)); +This function is not thread safe without your own synchronization. +*/ +ma_uint64 ma_decoder_get_length_in_pcm_frames(ma_decoder* pDecoder); - return MA_SUCCESS; -} +/* +Reads PCM frames from the given decoder. -void ma_thread_wait__win32(ma_thread* pThread) -{ - WaitForSingleObject(pThread->win32.hThread, INFINITE); -} +This is not thread safe without your own synchronization. +*/ +ma_uint64 ma_decoder_read_pcm_frames(ma_decoder* pDecoder, void* pFramesOut, ma_uint64 frameCount); -void ma_sleep__win32(ma_uint32 milliseconds) -{ - Sleep((DWORD)milliseconds); -} +/* +Seeks to a PCM frame based on it's absolute index. +This is not thread safe without your own synchronization. +*/ +ma_result ma_decoder_seek_to_pcm_frame(ma_decoder* pDecoder, ma_uint64 frameIndex); -ma_result ma_mutex_init__win32(ma_context* pContext, ma_mutex* pMutex) -{ - (void)pContext; +/* +Helper for opening and decoding a file into a heap allocated block of memory. Free the returned pointer with ma_free(). On input, +pConfig should be set to what you want. On output it will be set to what you got. +*/ +#ifndef MA_NO_STDIO +ma_result ma_decode_file(const char* pFilePath, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppDataOut); +#endif +ma_result ma_decode_memory(const void* pData, size_t dataSize, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppDataOut); - pMutex->win32.hMutex = CreateEventA(NULL, FALSE, TRUE, NULL); - if (pMutex->win32.hMutex == NULL) { - return MA_FAILED_TO_CREATE_MUTEX; - } +#endif /* MA_NO_DECODING */ - return MA_SUCCESS; -} -void ma_mutex_uninit__win32(ma_mutex* pMutex) -{ - CloseHandle(pMutex->win32.hMutex); -} +/************************************************************************************************************************************************************ + +Generation -void ma_mutex_lock__win32(ma_mutex* pMutex) +************************************************************************************************************************************************************/ +typedef enum { - WaitForSingleObject(pMutex->win32.hMutex, INFINITE); -} + ma_waveform_type_sine, + ma_waveform_type_square, + ma_waveform_type_triangle, + ma_waveform_type_sawtooth +} ma_waveform_type; -void ma_mutex_unlock__win32(ma_mutex* pMutex) +typedef struct { - SetEvent(pMutex->win32.hMutex); + ma_waveform_type type; + double amplitude; + double frequency; + double deltaTime; + double time; +} ma_waveform; + +ma_result ma_waveform_init(ma_waveform_type type, double amplitude, double frequency, ma_uint32 sampleRate, ma_waveform* pWaveform); +ma_uint64 ma_waveform_read_pcm_frames(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount, ma_format format, ma_uint32 channels); +ma_result ma_waveform_set_amplitude(ma_waveform* pWaveform, double amplitude); +ma_result ma_waveform_set_frequency(ma_waveform* pWaveform, double frequency); +ma_result ma_waveform_set_sample_rate(ma_waveform* pWaveform, ma_uint32 sampleRate); + +#ifdef __cplusplus } +#endif +#endif /* miniaudio_h */ -ma_result ma_event_init__win32(ma_context* pContext, ma_event* pEvent) -{ - (void)pContext; - pEvent->win32.hEvent = CreateEventW(NULL, FALSE, FALSE, NULL); - if (pEvent->win32.hEvent == NULL) { - return MA_FAILED_TO_CREATE_EVENT; - } +/************************************************************************************************************************************************************ +************************************************************************************************************************************************************* - return MA_SUCCESS; -} +IMPLEMENTATION -void ma_event_uninit__win32(ma_event* pEvent) -{ - CloseHandle(pEvent->win32.hEvent); -} +************************************************************************************************************************************************************* +************************************************************************************************************************************************************/ +#if defined(MINIAUDIO_IMPLEMENTATION) || defined(MA_IMPLEMENTATION) +#include +#include /* For INT_MAX */ +#include /* sin(), etc. */ -ma_bool32 ma_event_wait__win32(ma_event* pEvent) -{ - return WaitForSingleObject(pEvent->win32.hEvent, INFINITE) == WAIT_OBJECT_0; -} +#if !defined(MA_NO_STDIO) || defined(MA_DEBUG_OUTPUT) + #include + #if !defined(_MSC_VER) && !defined(__DMC__) + #include /* For strcasecmp(). */ + #include /* For wcslen(), wcsrtombs() */ + #endif +#endif -ma_bool32 ma_event_signal__win32(ma_event* pEvent) -{ - return SetEvent(pEvent->win32.hEvent); -} +#ifdef MA_WIN32 +#include +#include +#include +#include +#else +#include /* For malloc(), free(), wcstombs(). */ +#include /* For memset() */ #endif +#if defined(MA_APPLE) && (__MAC_OS_X_VERSION_MIN_REQUIRED < 101200) +#include /* For mach_absolute_time() */ +#endif #ifdef MA_POSIX -#include - -typedef int (* ma_pthread_create_proc)(pthread_t *thread, const pthread_attr_t *attr, void *(*start_routine) (void *), void *arg); -typedef int (* ma_pthread_join_proc)(pthread_t thread, void **retval); -typedef int (* ma_pthread_mutex_init_proc)(pthread_mutex_t *__mutex, const pthread_mutexattr_t *__mutexattr); -typedef int (* ma_pthread_mutex_destroy_proc)(pthread_mutex_t *__mutex); -typedef int (* ma_pthread_mutex_lock_proc)(pthread_mutex_t *__mutex); -typedef int (* ma_pthread_mutex_unlock_proc)(pthread_mutex_t *__mutex); -typedef int (* ma_pthread_cond_init_proc)(pthread_cond_t *__restrict __cond, const pthread_condattr_t *__restrict __cond_attr); -typedef int (* ma_pthread_cond_destroy_proc)(pthread_cond_t *__cond); -typedef int (* ma_pthread_cond_signal_proc)(pthread_cond_t *__cond); -typedef int (* ma_pthread_cond_wait_proc)(pthread_cond_t *__restrict __cond, pthread_mutex_t *__restrict __mutex); -typedef int (* ma_pthread_attr_init_proc)(pthread_attr_t *attr); -typedef int (* ma_pthread_attr_destroy_proc)(pthread_attr_t *attr); -typedef int (* ma_pthread_attr_setschedpolicy_proc)(pthread_attr_t *attr, int policy); -typedef int (* ma_pthread_attr_getschedparam_proc)(const pthread_attr_t *attr, struct sched_param *param); -typedef int (* ma_pthread_attr_setschedparam_proc)(pthread_attr_t *attr, const struct sched_param *param); +#include +#include +#include +#include +#endif -ma_result ma_thread_create__posix(ma_context* pContext, ma_thread* pThread, ma_thread_entry_proc entryProc, void* pData) -{ - int result; - pthread_attr_t* pAttr = NULL; +#ifdef MA_EMSCRIPTEN +#include +#endif -#if !defined(__EMSCRIPTEN__) - /* Try setting the thread priority. It's not critical if anything fails here. */ - pthread_attr_t attr; - if (((ma_pthread_attr_init_proc)pContext->posix.pthread_attr_init)(&attr) == 0) { - int scheduler = -1; - if (pContext->threadPriority == ma_thread_priority_idle) { -#ifdef SCHED_IDLE - if (((ma_pthread_attr_setschedpolicy_proc)pContext->posix.pthread_attr_setschedpolicy)(&attr, SCHED_IDLE) == 0) { - scheduler = SCHED_IDLE; - } +#if !defined(MA_64BIT) && !defined(MA_32BIT) +#ifdef _WIN32 +#ifdef _WIN64 +#define MA_64BIT +#else +#define MA_32BIT #endif - } else if (pContext->threadPriority == ma_thread_priority_realtime) { -#ifdef SCHED_FIFO - if (((ma_pthread_attr_setschedpolicy_proc)pContext->posix.pthread_attr_setschedpolicy)(&attr, SCHED_FIFO) == 0) { - scheduler = SCHED_FIFO; - } #endif -#ifdef MA_LINUX - } else { - scheduler = sched_getscheduler(0); #endif - } - if (scheduler != -1) { - int priorityMin = sched_get_priority_min(scheduler); - int priorityMax = sched_get_priority_max(scheduler); - int priorityStep = (priorityMax - priorityMin) / 7; /* 7 = number of priorities supported by miniaudio. */ +#if !defined(MA_64BIT) && !defined(MA_32BIT) +#ifdef __GNUC__ +#ifdef __LP64__ +#define MA_64BIT +#else +#define MA_32BIT +#endif +#endif +#endif - struct sched_param sched; - if (((ma_pthread_attr_getschedparam_proc)pContext->posix.pthread_attr_getschedparam)(&attr, &sched) == 0) { - if (pContext->threadPriority == ma_thread_priority_idle) { - sched.sched_priority = priorityMin; - } else if (pContext->threadPriority == ma_thread_priority_realtime) { - sched.sched_priority = priorityMax; - } else { - sched.sched_priority += ((int)pContext->threadPriority + 5) * priorityStep; /* +5 because the lowest priority is -5. */ - if (sched.sched_priority < priorityMin) { - sched.sched_priority = priorityMin; - } - if (sched.sched_priority > priorityMax) { - sched.sched_priority = priorityMax; - } - } +#if !defined(MA_64BIT) && !defined(MA_32BIT) +#include +#if INTPTR_MAX == INT64_MAX +#define MA_64BIT +#else +#define MA_32BIT +#endif +#endif - if (((ma_pthread_attr_setschedparam_proc)pContext->posix.pthread_attr_setschedparam)(&attr, &sched) == 0) { - pAttr = &attr; - } - } - } +/* Architecture Detection */ +#if defined(__x86_64__) || defined(_M_X64) +#define MA_X64 +#elif defined(__i386) || defined(_M_IX86) +#define MA_X86 +#elif defined(__arm__) || defined(_M_ARM) +#define MA_ARM +#endif - ((ma_pthread_attr_destroy_proc)pContext->posix.pthread_attr_destroy)(&attr); - } +/* Cannot currently support AVX-512 if AVX is disabled. */ +#if !defined(MA_NO_AVX512) && defined(MA_NO_AVX2) +#define MA_NO_AVX512 #endif - result = ((ma_pthread_create_proc)pContext->posix.pthread_create)(&pThread->posix.thread, pAttr, entryProc, pData); - if (result != 0) { - return MA_FAILED_TO_CREATE_THREAD; - } +/* Intrinsics Support */ +#if defined(MA_X64) || defined(MA_X86) + #if defined(_MSC_VER) && !defined(__clang__) + /* MSVC. */ + #if _MSC_VER >= 1400 && !defined(MA_NO_SSE2) /* 2005 */ + #define MA_SUPPORT_SSE2 + #endif + /*#if _MSC_VER >= 1600 && !defined(MA_NO_AVX)*/ /* 2010 */ + /* #define MA_SUPPORT_AVX*/ + /*#endif*/ + #if _MSC_VER >= 1700 && !defined(MA_NO_AVX2) /* 2012 */ + #define MA_SUPPORT_AVX2 + #endif + #if _MSC_VER >= 1910 && !defined(MA_NO_AVX512) /* 2017 */ + #define MA_SUPPORT_AVX512 + #endif + #else + /* Assume GNUC-style. */ + #if defined(__SSE2__) && !defined(MA_NO_SSE2) + #define MA_SUPPORT_SSE2 + #endif + /*#if defined(__AVX__) && !defined(MA_NO_AVX)*/ + /* #define MA_SUPPORT_AVX*/ + /*#endif*/ + #if defined(__AVX2__) && !defined(MA_NO_AVX2) + #define MA_SUPPORT_AVX2 + #endif + #if defined(__AVX512F__) && !defined(MA_NO_AVX512) + #define MA_SUPPORT_AVX512 + #endif + #endif - return MA_SUCCESS; -} + /* If at this point we still haven't determined compiler support for the intrinsics just fall back to __has_include. */ + #if !defined(__GNUC__) && !defined(__clang__) && defined(__has_include) + #if !defined(MA_SUPPORT_SSE2) && !defined(MA_NO_SSE2) && __has_include() + #define MA_SUPPORT_SSE2 + #endif + /*#if !defined(MA_SUPPORT_AVX) && !defined(MA_NO_AVX) && __has_include()*/ + /* #define MA_SUPPORT_AVX*/ + /*#endif*/ + #if !defined(MA_SUPPORT_AVX2) && !defined(MA_NO_AVX2) && __has_include() + #define MA_SUPPORT_AVX2 + #endif + #if !defined(MA_SUPPORT_AVX512) && !defined(MA_NO_AVX512) && __has_include() + #define MA_SUPPORT_AVX512 + #endif + #endif -void ma_thread_wait__posix(ma_thread* pThread) -{ - ((ma_pthread_join_proc)pThread->pContext->posix.pthread_join)(pThread->posix.thread, NULL); -} + #if defined(MA_SUPPORT_AVX512) + #include /* Not a mistake. Intentionally including instead of because otherwise the compiler will complain. */ + #elif defined(MA_SUPPORT_AVX2) || defined(MA_SUPPORT_AVX) + #include + #elif defined(MA_SUPPORT_SSE2) + #include + #endif +#endif -void ma_sleep__posix(ma_uint32 milliseconds) -{ -#ifdef MA_EMSCRIPTEN - (void)milliseconds; - ma_assert(MA_FALSE); /* The Emscripten build should never sleep. */ -#else - #if _POSIX_C_SOURCE >= 199309L - struct timespec ts; - ts.tv_sec = milliseconds / 1000000; - ts.tv_nsec = milliseconds % 1000000 * 1000000; - nanosleep(&ts, NULL); - #else - struct timeval tv; - tv.tv_sec = milliseconds / 1000; - tv.tv_usec = milliseconds % 1000 * 1000; - select(0, NULL, NULL, NULL, &tv); +#if defined(MA_ARM) + #if !defined(MA_NO_NEON) && (defined(__ARM_NEON) || defined(__aarch64__) || defined(_M_ARM64)) + #define MA_SUPPORT_NEON + #endif + + /* Fall back to looking for the #include file. */ + #if !defined(__GNUC__) && !defined(__clang__) && defined(__has_include) + #if !defined(MA_SUPPORT_NEON) && !defined(MA_NO_NEON) && __has_include() + #define MA_SUPPORT_NEON + #endif + #endif + + #if defined(MA_SUPPORT_NEON) + #include #endif #endif -} - -ma_result ma_mutex_init__posix(ma_context* pContext, ma_mutex* pMutex) -{ - int result = ((ma_pthread_mutex_init_proc)pContext->posix.pthread_mutex_init)(&pMutex->posix.mutex, NULL); - if (result != 0) { - return MA_FAILED_TO_CREATE_MUTEX; - } +/* Begin globally disabled warnings. */ +#if defined(_MSC_VER) + #pragma warning(push) + #pragma warning(disable:4752) /* found Intel(R) Advanced Vector Extensions; consider using /arch:AVX */ +#endif - return MA_SUCCESS; -} +#if defined(MA_X64) || defined(MA_X86) + #if defined(_MSC_VER) && !defined(__clang__) + #if _MSC_VER >= 1400 + #include + static MA_INLINE void ma_cpuid(int info[4], int fid) + { + __cpuid(info, fid); + } + #else + #define MA_NO_CPUID + #endif -void ma_mutex_uninit__posix(ma_mutex* pMutex) -{ - ((ma_pthread_mutex_destroy_proc)pMutex->pContext->posix.pthread_mutex_destroy)(&pMutex->posix.mutex); -} + #if _MSC_VER >= 1600 && (defined(_MSC_FULL_VER) && _MSC_FULL_VER >= 160040219) + static MA_INLINE unsigned __int64 ma_xgetbv(int reg) + { + return _xgetbv(reg); + } + #else + #define MA_NO_XGETBV + #endif + #elif (defined(__GNUC__) || defined(__clang__)) && !defined(MA_ANDROID) + static MA_INLINE void ma_cpuid(int info[4], int fid) + { + /* + It looks like the -fPIC option uses the ebx register which GCC complains about. We can work around this by just using a different register, the + specific register of which I'm letting the compiler decide on. The "k" prefix is used to specify a 32-bit register. The {...} syntax is for + supporting different assembly dialects. + + What's basically happening is that we're saving and restoring the ebx register manually. + */ + #if defined(DRFLAC_X86) && defined(__PIC__) + __asm__ __volatile__ ( + "xchg{l} {%%}ebx, %k1;" + "cpuid;" + "xchg{l} {%%}ebx, %k1;" + : "=a"(info[0]), "=&r"(info[1]), "=c"(info[2]), "=d"(info[3]) : "a"(fid), "c"(0) + ); + #else + __asm__ __volatile__ ( + "cpuid" : "=a"(info[0]), "=b"(info[1]), "=c"(info[2]), "=d"(info[3]) : "a"(fid), "c"(0) + ); + #endif + } -void ma_mutex_lock__posix(ma_mutex* pMutex) -{ - ((ma_pthread_mutex_lock_proc)pMutex->pContext->posix.pthread_mutex_lock)(&pMutex->posix.mutex); -} + static MA_INLINE ma_uint64 ma_xgetbv(int reg) + { + unsigned int hi; + unsigned int lo; -void ma_mutex_unlock__posix(ma_mutex* pMutex) -{ - ((ma_pthread_mutex_unlock_proc)pMutex->pContext->posix.pthread_mutex_unlock)(&pMutex->posix.mutex); -} + __asm__ __volatile__ ( + "xgetbv" : "=a"(lo), "=d"(hi) : "c"(reg) + ); + return ((ma_uint64)hi << 32) | (ma_uint64)lo; + } + #else + #define MA_NO_CPUID + #define MA_NO_XGETBV + #endif +#else + #define MA_NO_CPUID + #define MA_NO_XGETBV +#endif -ma_result ma_event_init__posix(ma_context* pContext, ma_event* pEvent) +static MA_INLINE ma_bool32 ma_has_sse2() { - if (((ma_pthread_mutex_init_proc)pContext->posix.pthread_mutex_init)(&pEvent->posix.mutex, NULL) != 0) { - return MA_FAILED_TO_CREATE_MUTEX; - } - - if (((ma_pthread_cond_init_proc)pContext->posix.pthread_cond_init)(&pEvent->posix.condition, NULL) != 0) { - return MA_FAILED_TO_CREATE_EVENT; - } - - pEvent->posix.value = 0; - return MA_SUCCESS; +#if defined(MA_SUPPORT_SSE2) + #if (defined(MA_X64) || defined(MA_X86)) && !defined(MA_NO_SSE2) + #if defined(MA_X64) + return MA_TRUE; /* 64-bit targets always support SSE2. */ + #elif (defined(_M_IX86_FP) && _M_IX86_FP == 2) || defined(__SSE2__) + return MA_TRUE; /* If the compiler is allowed to freely generate SSE2 code we can assume support. */ + #else + #if defined(MA_NO_CPUID) + return MA_FALSE; + #else + int info[4]; + ma_cpuid(info, 1); + return (info[3] & (1 << 26)) != 0; + #endif + #endif + #else + return MA_FALSE; /* SSE2 is only supported on x86 and x64 architectures. */ + #endif +#else + return MA_FALSE; /* No compiler support. */ +#endif } -void ma_event_uninit__posix(ma_event* pEvent) +#if 0 +static MA_INLINE ma_bool32 ma_has_avx() { - ((ma_pthread_cond_destroy_proc)pEvent->pContext->posix.pthread_cond_destroy)(&pEvent->posix.condition); - ((ma_pthread_mutex_destroy_proc)pEvent->pContext->posix.pthread_mutex_destroy)(&pEvent->posix.mutex); +#if defined(MA_SUPPORT_AVX) + #if (defined(MA_X64) || defined(MA_X86)) && !defined(MA_NO_AVX) + #if defined(_AVX_) || defined(__AVX__) + return MA_TRUE; /* If the compiler is allowed to freely generate AVX code we can assume support. */ + #else + /* AVX requires both CPU and OS support. */ + #if defined(MA_NO_CPUID) || defined(MA_NO_XGETBV) + return MA_FALSE; + #else + int info[4]; + ma_cpuid(info, 1); + if (((info[2] & (1 << 27)) != 0) && ((info[2] & (1 << 28)) != 0)) { + ma_uint64 xrc = ma_xgetbv(0); + if ((xrc & 0x06) == 0x06) { + return MA_TRUE; + } else { + return MA_FALSE; + } + } else { + return MA_FALSE; + } + #endif + #endif + #else + return MA_FALSE; /* AVX is only supported on x86 and x64 architectures. */ + #endif +#else + return MA_FALSE; /* No compiler support. */ +#endif } +#endif -ma_bool32 ma_event_wait__posix(ma_event* pEvent) +static MA_INLINE ma_bool32 ma_has_avx2() { - ((ma_pthread_mutex_lock_proc)pEvent->pContext->posix.pthread_mutex_lock)(&pEvent->posix.mutex); - { - while (pEvent->posix.value == 0) { - ((ma_pthread_cond_wait_proc)pEvent->pContext->posix.pthread_cond_wait)(&pEvent->posix.condition, &pEvent->posix.mutex); - } - pEvent->posix.value = 0; /* Auto-reset. */ - } - ((ma_pthread_mutex_unlock_proc)pEvent->pContext->posix.pthread_mutex_unlock)(&pEvent->posix.mutex); - - return MA_TRUE; +#if defined(MA_SUPPORT_AVX2) + #if (defined(MA_X64) || defined(MA_X86)) && !defined(MA_NO_AVX2) + #if defined(_AVX2_) || defined(__AVX2__) + return MA_TRUE; /* If the compiler is allowed to freely generate AVX2 code we can assume support. */ + #else + /* AVX2 requires both CPU and OS support. */ + #if defined(MA_NO_CPUID) || defined(MA_NO_XGETBV) + return MA_FALSE; + #else + int info1[4]; + int info7[4]; + ma_cpuid(info1, 1); + ma_cpuid(info7, 7); + if (((info1[2] & (1 << 27)) != 0) && ((info7[1] & (1 << 5)) != 0)) { + ma_uint64 xrc = ma_xgetbv(0); + if ((xrc & 0x06) == 0x06) { + return MA_TRUE; + } else { + return MA_FALSE; + } + } else { + return MA_FALSE; + } + #endif + #endif + #else + return MA_FALSE; /* AVX2 is only supported on x86 and x64 architectures. */ + #endif +#else + return MA_FALSE; /* No compiler support. */ +#endif } -ma_bool32 ma_event_signal__posix(ma_event* pEvent) +static MA_INLINE ma_bool32 ma_has_avx512f() { - ((ma_pthread_mutex_lock_proc)pEvent->pContext->posix.pthread_mutex_lock)(&pEvent->posix.mutex); - { - pEvent->posix.value = 1; - ((ma_pthread_cond_signal_proc)pEvent->pContext->posix.pthread_cond_signal)(&pEvent->posix.condition); - } - ((ma_pthread_mutex_unlock_proc)pEvent->pContext->posix.pthread_mutex_unlock)(&pEvent->posix.mutex); - - return MA_TRUE; -} +#if defined(MA_SUPPORT_AVX512) + #if (defined(MA_X64) || defined(MA_X86)) && !defined(MA_NO_AVX512) + #if defined(__AVX512F__) + return MA_TRUE; /* If the compiler is allowed to freely generate AVX-512F code we can assume support. */ + #else + /* AVX-512 requires both CPU and OS support. */ + #if defined(MA_NO_CPUID) || defined(MA_NO_XGETBV) + return MA_FALSE; + #else + int info1[4]; + int info7[4]; + ma_cpuid(info1, 1); + ma_cpuid(info7, 7); + if (((info1[2] & (1 << 27)) != 0) && ((info7[1] & (1 << 16)) != 0)) { + ma_uint64 xrc = ma_xgetbv(0); + if ((xrc & 0xE6) == 0xE6) { + return MA_TRUE; + } else { + return MA_FALSE; + } + } else { + return MA_FALSE; + } + #endif + #endif + #else + return MA_FALSE; /* AVX-512F is only supported on x86 and x64 architectures. */ + #endif +#else + return MA_FALSE; /* No compiler support. */ #endif +} -ma_result ma_thread_create(ma_context* pContext, ma_thread* pThread, ma_thread_entry_proc entryProc, void* pData) +static MA_INLINE ma_bool32 ma_has_neon() { - if (pContext == NULL || pThread == NULL || entryProc == NULL) { - return MA_FALSE; - } - - pThread->pContext = pContext; - -#ifdef MA_WIN32 - return ma_thread_create__win32(pContext, pThread, entryProc, pData); -#endif -#ifdef MA_POSIX - return ma_thread_create__posix(pContext, pThread, entryProc, pData); +#if defined(MA_SUPPORT_NEON) + #if defined(MA_ARM) && !defined(MA_NO_NEON) + #if (defined(__ARM_NEON) || defined(__aarch64__) || defined(_M_ARM64)) + return MA_TRUE; /* If the compiler is allowed to freely generate NEON code we can assume support. */ + #else + /* TODO: Runtime check. */ + return MA_FALSE; + #endif + #else + return MA_FALSE; /* NEON is only supported on ARM architectures. */ + #endif +#else + return MA_FALSE; /* No compiler support. */ #endif } -void ma_thread_wait(ma_thread* pThread) -{ - if (pThread == NULL) { - return; - } +#define MA_SIMD_NONE 0 +#define MA_SIMD_SSE2 1 +#define MA_SIMD_AVX2 2 +#define MA_SIMD_NEON 3 -#ifdef MA_WIN32 - ma_thread_wait__win32(pThread); +#ifndef MA_PREFERRED_SIMD + # if defined(MA_SUPPORT_SSE2) && defined(MA_PREFER_SSE2) + #define MA_PREFERRED_SIMD MA_SIMD_SSE2 + #elif defined(MA_SUPPORT_AVX2) && defined(MA_PREFER_AVX2) + #define MA_PREFERRED_SIMD MA_SIMD_AVX2 + #elif defined(MA_SUPPORT_NEON) && defined(MA_PREFER_NEON) + #define MA_PREFERRED_SIMD MA_SIMD_NEON + #else + #define MA_PREFERRED_SIMD MA_SIMD_NONE + #endif #endif -#ifdef MA_POSIX - ma_thread_wait__posix(pThread); + + +static MA_INLINE ma_bool32 ma_is_little_endian() +{ +#if defined(MA_X86) || defined(MA_X64) + return MA_TRUE; +#else + int n = 1; + return (*(char*)&n) == 1; #endif } -void ma_sleep(ma_uint32 milliseconds) +static MA_INLINE ma_bool32 ma_is_big_endian() { -#ifdef MA_WIN32 - ma_sleep__win32(milliseconds); -#endif -#ifdef MA_POSIX - ma_sleep__posix(milliseconds); -#endif + return !ma_is_little_endian(); } -ma_result ma_mutex_init(ma_context* pContext, ma_mutex* pMutex) -{ - if (pContext == NULL || pMutex == NULL) { - return MA_INVALID_ARGS; - } +#ifndef MA_COINIT_VALUE +#define MA_COINIT_VALUE 0 /* 0 = COINIT_MULTITHREADED */ +#endif - pMutex->pContext = pContext; -#ifdef MA_WIN32 - return ma_mutex_init__win32(pContext, pMutex); + +#ifndef MA_PI +#define MA_PI 3.14159265358979323846264f #endif -#ifdef MA_POSIX - return ma_mutex_init__posix(pContext, pMutex); +#ifndef MA_PI_D +#define MA_PI_D 3.14159265358979323846264 +#endif +#ifndef MA_TAU +#define MA_TAU 6.28318530717958647693f +#endif +#ifndef MA_TAU_D +#define MA_TAU_D 6.28318530717958647693 #endif -} -void ma_mutex_uninit(ma_mutex* pMutex) -{ - if (pMutex == NULL || pMutex->pContext == NULL) { - return; - } -#ifdef MA_WIN32 - ma_mutex_uninit__win32(pMutex); -#endif -#ifdef MA_POSIX - ma_mutex_uninit__posix(pMutex); +/* The default format when ma_format_unknown (0) is requested when initializing a device. */ +#ifndef MA_DEFAULT_FORMAT +#define MA_DEFAULT_FORMAT ma_format_f32 #endif -} -void ma_mutex_lock(ma_mutex* pMutex) -{ - if (pMutex == NULL || pMutex->pContext == NULL) { - return; - } +/* The default channel count to use when 0 is used when initializing a device. */ +#ifndef MA_DEFAULT_CHANNELS +#define MA_DEFAULT_CHANNELS 2 +#endif -#ifdef MA_WIN32 - ma_mutex_lock__win32(pMutex); +/* The default sample rate to use when 0 is used when initializing a device. */ +#ifndef MA_DEFAULT_SAMPLE_RATE +#define MA_DEFAULT_SAMPLE_RATE 48000 #endif -#ifdef MA_POSIX - ma_mutex_lock__posix(pMutex); + +/* Default periods when none is specified in ma_device_init(). More periods means more work on the CPU. */ +#ifndef MA_DEFAULT_PERIODS +#define MA_DEFAULT_PERIODS 3 #endif -} -void ma_mutex_unlock(ma_mutex* pMutex) -{ - if (pMutex == NULL || pMutex->pContext == NULL) { - return; -} +/* The default period size in milliseconds for low latency mode. */ +#ifndef MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY +#define MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY 10 +#endif -#ifdef MA_WIN32 - ma_mutex_unlock__win32(pMutex); +/* The default buffer size in milliseconds for conservative mode. */ +#ifndef MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE +#define MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE 100 #endif -#ifdef MA_POSIX - ma_mutex_unlock__posix(pMutex); + +/* The default LPF count for linear resampling. Note that this is clamped to MA_MAX_RESAMPLER_LPF_FILTERS. */ +#ifndef MA_DEFAULT_RESAMPLER_LPF_FILTERS + #if MA_MAX_RESAMPLER_LPF_FILTERS >= 2 + #define MA_DEFAULT_RESAMPLER_LPF_FILTERS 2 + #else + #define MA_DEFAULT_RESAMPLER_LPF_FILTERS MA_MAX_RESAMPLER_LPF_FILTERS + #endif #endif -} -ma_result ma_event_init(ma_context* pContext, ma_event* pEvent) -{ - if (pContext == NULL || pEvent == NULL) { - return MA_FALSE; - } +/* Standard sample rates, in order of priority. */ +ma_uint32 g_maStandardSampleRatePriorities[] = { + MA_SAMPLE_RATE_48000, /* Most common */ + MA_SAMPLE_RATE_44100, + + MA_SAMPLE_RATE_32000, /* Lows */ + MA_SAMPLE_RATE_24000, + MA_SAMPLE_RATE_22050, + + MA_SAMPLE_RATE_88200, /* Highs */ + MA_SAMPLE_RATE_96000, + MA_SAMPLE_RATE_176400, + MA_SAMPLE_RATE_192000, + + MA_SAMPLE_RATE_16000, /* Extreme lows */ + MA_SAMPLE_RATE_11025, + MA_SAMPLE_RATE_8000, + + MA_SAMPLE_RATE_352800, /* Extreme highs */ + MA_SAMPLE_RATE_384000 +}; + +ma_format g_maFormatPriorities[] = { + ma_format_s16, /* Most common */ + ma_format_f32, + + /*ma_format_s24_32,*/ /* Clean alignment */ + ma_format_s32, + + ma_format_s24, /* Unclean alignment */ + + ma_format_u8 /* Low quality */ +}; - pEvent->pContext = pContext; + +/****************************************************************************** + +Standard Library Stuff + +******************************************************************************/ +#ifndef MA_MALLOC #ifdef MA_WIN32 - return ma_event_init__win32(pContext, pEvent); +#define MA_MALLOC(sz) HeapAlloc(GetProcessHeap(), 0, (sz)) +#else +#define MA_MALLOC(sz) malloc((sz)) #endif -#ifdef MA_POSIX - return ma_event_init__posix(pContext, pEvent); #endif -} - -void ma_event_uninit(ma_event* pEvent) -{ - if (pEvent == NULL || pEvent->pContext == NULL) { - return; - } +#ifndef MA_REALLOC #ifdef MA_WIN32 - ma_event_uninit__win32(pEvent); +#define MA_REALLOC(p, sz) (((sz) > 0) ? ((p) ? HeapReAlloc(GetProcessHeap(), 0, (p), (sz)) : HeapAlloc(GetProcessHeap(), 0, (sz))) : ((VOID*)(size_t)(HeapFree(GetProcessHeap(), 0, (p)) & 0))) +#else +#define MA_REALLOC(p, sz) realloc((p), (sz)) #endif -#ifdef MA_POSIX - ma_event_uninit__posix(pEvent); #endif -} - -ma_bool32 ma_event_wait(ma_event* pEvent) -{ - if (pEvent == NULL || pEvent->pContext == NULL) { - return MA_FALSE; - } +#ifndef MA_FREE #ifdef MA_WIN32 - return ma_event_wait__win32(pEvent); +#define MA_FREE(p) HeapFree(GetProcessHeap(), 0, (p)) +#else +#define MA_FREE(p) free((p)) #endif -#ifdef MA_POSIX - return ma_event_wait__posix(pEvent); #endif -} - -ma_bool32 ma_event_signal(ma_event* pEvent) -{ - if (pEvent == NULL || pEvent->pContext == NULL) { - return MA_FALSE; - } +#ifndef MA_ZERO_MEMORY #ifdef MA_WIN32 - return ma_event_signal__win32(pEvent); +#define MA_ZERO_MEMORY(p, sz) ZeroMemory((p), (sz)) +#else +#define MA_ZERO_MEMORY(p, sz) memset((p), 0, (sz)) #endif -#ifdef MA_POSIX - return ma_event_signal__posix(pEvent); #endif -} +#ifndef MA_COPY_MEMORY +#ifdef MA_WIN32 +#define MA_COPY_MEMORY(dst, src, sz) CopyMemory((dst), (src), (sz)) +#else +#define MA_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz)) +#endif +#endif -ma_uint32 ma_get_best_sample_rate_within_range(ma_uint32 sampleRateMin, ma_uint32 sampleRateMax) -{ - /* Normalize the range in case we were given something stupid. */ - if (sampleRateMin < MA_MIN_SAMPLE_RATE) { - sampleRateMin = MA_MIN_SAMPLE_RATE; - } - if (sampleRateMax > MA_MAX_SAMPLE_RATE) { - sampleRateMax = MA_MAX_SAMPLE_RATE; - } - if (sampleRateMin > sampleRateMax) { - sampleRateMin = sampleRateMax; - } - - if (sampleRateMin == sampleRateMax) { - return sampleRateMax; - } else { - size_t iStandardRate; - for (iStandardRate = 0; iStandardRate < ma_countof(g_maStandardSampleRatePriorities); ++iStandardRate) { - ma_uint32 standardRate = g_maStandardSampleRatePriorities[iStandardRate]; - if (standardRate >= sampleRateMin && standardRate <= sampleRateMax) { - return standardRate; - } - } - } - - /* Should never get here. */ - ma_assert(MA_FALSE); - return 0; -} - -ma_uint32 ma_get_closest_standard_sample_rate(ma_uint32 sampleRateIn) -{ - ma_uint32 closestRate = 0; - ma_uint32 closestDiff = 0xFFFFFFFF; - size_t iStandardRate; - - for (iStandardRate = 0; iStandardRate < ma_countof(g_maStandardSampleRatePriorities); ++iStandardRate) { - ma_uint32 standardRate = g_maStandardSampleRatePriorities[iStandardRate]; - ma_uint32 diff; +#ifndef MA_ASSERT +#ifdef MA_WIN32 +#define MA_ASSERT(condition) assert(condition) +#else +#define MA_ASSERT(condition) assert(condition) +#endif +#endif - if (sampleRateIn > standardRate) { - diff = sampleRateIn - standardRate; - } else { - diff = standardRate - sampleRateIn; - } +#define MA_ZERO_OBJECT(p) MA_ZERO_MEMORY((p), sizeof(*(p))) - if (diff == 0) { - return standardRate; /* The input sample rate is a standard rate. */ - } +#define ma_countof(x) (sizeof(x) / sizeof(x[0])) +#define ma_max(x, y) (((x) > (y)) ? (x) : (y)) +#define ma_min(x, y) (((x) < (y)) ? (x) : (y)) +#define ma_abs(x) (((x) > 0) ? (x) : -(x)) +#define ma_clamp(x, lo, hi) (ma_max(lo, ma_min(x, hi))) +#define ma_offset_ptr(p, offset) (((ma_uint8*)(p)) + (offset)) - if (closestDiff > diff) { - closestDiff = diff; - closestRate = standardRate; - } - } +#define ma_buffer_frame_capacity(buffer, channels, format) (sizeof(buffer) / ma_get_bytes_per_sample(format) / (channels)) - return closestRate; +static MA_INLINE double ma_sin(double x) +{ + /* TODO: Implement custom sin(x). */ + return sin(x); } +static MA_INLINE double ma_cos(double x) +{ + return ma_sin((MA_PI*0.5) - x); +} -ma_uint32 ma_scale_buffer_size(ma_uint32 baseBufferSize, float scale) +static MA_INLINE double ma_log(double x) { - return ma_max(1, (ma_uint32)(baseBufferSize*scale)); + /* TODO: Implement custom log(x). */ + return log(x); } -ma_uint32 ma_calculate_buffer_size_in_milliseconds_from_frames(ma_uint32 bufferSizeInFrames, ma_uint32 sampleRate) +static MA_INLINE double ma_pow(double x, double y) { - return bufferSizeInFrames / (sampleRate/1000); + /* TODO: Implement custom pow(x, y). */ + return pow(x, y); } -ma_uint32 ma_calculate_buffer_size_in_frames_from_milliseconds(ma_uint32 bufferSizeInMilliseconds, ma_uint32 sampleRate) +static MA_INLINE double ma_log10(double x) { - return bufferSizeInMilliseconds * (sampleRate/1000); + return ma_log(x) * 0.43429448190325182765; } -ma_uint32 ma_get_default_buffer_size_in_milliseconds(ma_performance_profile performanceProfile) +static MA_INLINE float ma_powf(float x, float y) { - if (performanceProfile == ma_performance_profile_low_latency) { - return MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_LOW_LATENCY; - } else { - return MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_CONSERVATIVE; - } + return (float)ma_pow((double)x, (double)y); +} + +static MA_INLINE float ma_log10f(float x) +{ + return (float)ma_log10((double)x); } -ma_uint32 ma_get_default_buffer_size_in_frames(ma_performance_profile performanceProfile, ma_uint32 sampleRate) + +/* +Return Values: + 0: Success + 22: EINVAL + 34: ERANGE + +Not using symbolic constants for errors because I want to avoid #including errno.h +*/ +int ma_strcpy_s(char* dst, size_t dstSizeInBytes, const char* src) { - ma_uint32 bufferSizeInMilliseconds; - ma_uint32 sampleRateMS; + size_t i; - bufferSizeInMilliseconds = ma_get_default_buffer_size_in_milliseconds(performanceProfile); - if (bufferSizeInMilliseconds == 0) { - bufferSizeInMilliseconds = 1; + if (dst == 0) { + return 22; + } + if (dstSizeInBytes == 0) { + return 34; + } + if (src == 0) { + dst[0] = '\0'; + return 22; } - sampleRateMS = (sampleRate/1000); - if (sampleRateMS == 0) { - sampleRateMS = 1; + for (i = 0; i < dstSizeInBytes && src[i] != '\0'; ++i) { + dst[i] = src[i]; } - return bufferSizeInMilliseconds * sampleRateMS; -} + if (i < dstSizeInBytes) { + dst[i] = '\0'; + return 0; + } -ma_uint32 ma_get_fragment_size_in_bytes(ma_uint32 bufferSizeInFrames, ma_uint32 periods, ma_format format, ma_uint32 channels) -{ - ma_uint32 fragmentSizeInFrames = bufferSizeInFrames / periods; - return fragmentSizeInFrames * ma_get_bytes_per_frame(format, channels); + dst[0] = '\0'; + return 34; } -void ma_zero_pcm_frames(void* p, ma_uint32 frameCount, ma_format format, ma_uint32 channels) +int ma_strncpy_s(char* dst, size_t dstSizeInBytes, const char* src, size_t count) { - ma_zero_memory(p, frameCount * ma_get_bytes_per_frame(format, channels)); -} + size_t maxcount; + size_t i; -void ma_clip_samples_f32(float* p, ma_uint32 sampleCount) -{ - ma_uint32 iSample; + if (dst == 0) { + return 22; + } + if (dstSizeInBytes == 0) { + return 34; + } + if (src == 0) { + dst[0] = '\0'; + return 22; + } - /* TODO: Research a branchless SSE implementation. */ - for (iSample = 0; iSample < sampleCount; iSample += 1) { - p[iSample] = ma_clip_f32(p[iSample]); + maxcount = count; + if (count == ((size_t)-1) || count >= dstSizeInBytes) { /* -1 = _TRUNCATE */ + maxcount = dstSizeInBytes - 1; + } + + for (i = 0; i < maxcount && src[i] != '\0'; ++i) { + dst[i] = src[i]; } -} + if (src[i] == '\0' || i == count || count == ((size_t)-1)) { + dst[i] = '\0'; + return 0; + } -void ma_copy_and_apply_volume_factor_u8(ma_uint8* pSamplesOut, const ma_uint8* pSamplesIn, ma_uint32 sampleCount, float factor) + dst[0] = '\0'; + return 34; +} + +int ma_strcat_s(char* dst, size_t dstSizeInBytes, const char* src) { - ma_uint32 iSample; + char* dstorig; - if (pSamplesOut == NULL || pSamplesIn == NULL) { - return; + if (dst == 0) { + return 22; + } + if (dstSizeInBytes == 0) { + return 34; + } + if (src == 0) { + dst[0] = '\0'; + return 22; } - for (iSample = 0; iSample < sampleCount; iSample += 1) { - pSamplesOut[iSample] = (ma_uint8)(pSamplesIn[iSample] * factor); + dstorig = dst; + + while (dstSizeInBytes > 0 && dst[0] != '\0') { + dst += 1; + dstSizeInBytes -= 1; } -} -void ma_copy_and_apply_volume_factor_s16(ma_int16* pSamplesOut, const ma_int16* pSamplesIn, ma_uint32 sampleCount, float factor) -{ - ma_uint32 iSample; + if (dstSizeInBytes == 0) { + return 22; /* Unterminated. */ + } - if (pSamplesOut == NULL || pSamplesIn == NULL) { - return; + + while (dstSizeInBytes > 0 && src[0] != '\0') { + *dst++ = *src++; + dstSizeInBytes -= 1; } - for (iSample = 0; iSample < sampleCount; iSample += 1) { - pSamplesOut[iSample] = (ma_int16)(pSamplesIn[iSample] * factor); + if (dstSizeInBytes > 0) { + dst[0] = '\0'; + } else { + dstorig[0] = '\0'; + return 34; } + + return 0; } -void ma_copy_and_apply_volume_factor_s24(void* pSamplesOut, const void* pSamplesIn, ma_uint32 sampleCount, float factor) +int ma_strncat_s(char* dst, size_t dstSizeInBytes, const char* src, size_t count) { - ma_uint32 iSample; - ma_uint8* pSamplesOut8; - ma_uint8* pSamplesIn8; + char* dstorig; - if (pSamplesOut == NULL || pSamplesIn == NULL) { - return; + if (dst == 0) { + return 22; + } + if (dstSizeInBytes == 0) { + return 34; + } + if (src == 0) { + return 22; } - pSamplesOut8 = (ma_uint8*)pSamplesOut; - pSamplesIn8 = (ma_uint8*)pSamplesIn; - - for (iSample = 0; iSample < sampleCount; iSample += 1) { - ma_int32 sampleS32; + dstorig = dst; - sampleS32 = (ma_int32)(((ma_uint32)(pSamplesIn8[iSample*3+0]) << 8) | ((ma_uint32)(pSamplesIn8[iSample*3+1]) << 16) | ((ma_uint32)(pSamplesIn8[iSample*3+2])) << 24); - sampleS32 = (ma_int32)(sampleS32 * factor); + while (dstSizeInBytes > 0 && dst[0] != '\0') { + dst += 1; + dstSizeInBytes -= 1; + } - pSamplesOut8[iSample*3+0] = (ma_uint8)(((ma_uint32)sampleS32 & 0x0000FF00) >> 8); - pSamplesOut8[iSample*3+1] = (ma_uint8)(((ma_uint32)sampleS32 & 0x00FF0000) >> 16); - pSamplesOut8[iSample*3+2] = (ma_uint8)(((ma_uint32)sampleS32 & 0xFF000000) >> 24); + if (dstSizeInBytes == 0) { + return 22; /* Unterminated. */ } -} -void ma_copy_and_apply_volume_factor_s32(ma_int32* pSamplesOut, const ma_int32* pSamplesIn, ma_uint32 sampleCount, float factor) -{ - ma_uint32 iSample; - if (pSamplesOut == NULL || pSamplesIn == NULL) { - return; + if (count == ((size_t)-1)) { /* _TRUNCATE */ + count = dstSizeInBytes - 1; } - for (iSample = 0; iSample < sampleCount; iSample += 1) { - pSamplesOut[iSample] = (ma_int32)(pSamplesIn[iSample] * factor); + while (dstSizeInBytes > 0 && src[0] != '\0' && count > 0) { + *dst++ = *src++; + dstSizeInBytes -= 1; + count -= 1; + } + + if (dstSizeInBytes > 0) { + dst[0] = '\0'; + } else { + dstorig[0] = '\0'; + return 34; } + + return 0; } -void ma_copy_and_apply_volume_factor_f32(float* pSamplesOut, const float* pSamplesIn, ma_uint32 sampleCount, float factor) +int ma_itoa_s(int value, char* dst, size_t dstSizeInBytes, int radix) { - ma_uint32 iSample; + int sign; + unsigned int valueU; + char* dstEnd; - if (pSamplesOut == NULL || pSamplesIn == NULL) { - return; + if (dst == NULL || dstSizeInBytes == 0) { + return 22; + } + if (radix < 2 || radix > 36) { + dst[0] = '\0'; + return 22; } - for (iSample = 0; iSample < sampleCount; iSample += 1) { - pSamplesOut[iSample] = pSamplesIn[iSample] * factor; + sign = (value < 0 && radix == 10) ? -1 : 1; /* The negative sign is only used when the base is 10. */ + + if (value < 0) { + valueU = -value; + } else { + valueU = value; } -} -void ma_apply_volume_factor_u8(ma_uint8* pSamples, ma_uint32 sampleCount, float factor) -{ - ma_copy_and_apply_volume_factor_u8(pSamples, pSamples, sampleCount, factor); -} + dstEnd = dst; + do + { + int remainder = valueU % radix; + if (remainder > 9) { + *dstEnd = (char)((remainder - 10) + 'a'); + } else { + *dstEnd = (char)(remainder + '0'); + } -void ma_apply_volume_factor_s16(ma_int16* pSamples, ma_uint32 sampleCount, float factor) -{ - ma_copy_and_apply_volume_factor_s16(pSamples, pSamples, sampleCount, factor); -} + dstEnd += 1; + dstSizeInBytes -= 1; + valueU /= radix; + } while (dstSizeInBytes > 0 && valueU > 0); -void ma_apply_volume_factor_s24(void* pSamples, ma_uint32 sampleCount, float factor) -{ - ma_copy_and_apply_volume_factor_s24(pSamples, pSamples, sampleCount, factor); -} + if (dstSizeInBytes == 0) { + dst[0] = '\0'; + return 22; /* Ran out of room in the output buffer. */ + } -void ma_apply_volume_factor_s32(ma_int32* pSamples, ma_uint32 sampleCount, float factor) -{ - ma_copy_and_apply_volume_factor_s32(pSamples, pSamples, sampleCount, factor); -} + if (sign < 0) { + *dstEnd++ = '-'; + dstSizeInBytes -= 1; + } -void ma_apply_volume_factor_f32(float* pSamples, ma_uint32 sampleCount, float factor) -{ - ma_copy_and_apply_volume_factor_f32(pSamples, pSamples, sampleCount, factor); -} + if (dstSizeInBytes == 0) { + dst[0] = '\0'; + return 22; /* Ran out of room in the output buffer. */ + } + + *dstEnd = '\0'; + + + /* At this point the string will be reversed. */ + dstEnd -= 1; + while (dst < dstEnd) { + char temp = *dst; + *dst = *dstEnd; + *dstEnd = temp; -void ma_copy_and_apply_volume_factor_pcm_frames_u8(ma_uint8* pPCMFramesOut, const ma_uint8* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor) -{ - ma_copy_and_apply_volume_factor_u8(pPCMFramesOut, pPCMFramesIn, frameCount*channels, factor); -} + dst += 1; + dstEnd -= 1; + } -void ma_copy_and_apply_volume_factor_pcm_frames_s16(ma_int16* pPCMFramesOut, const ma_int16* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor) -{ - ma_copy_and_apply_volume_factor_s16(pPCMFramesOut, pPCMFramesIn, frameCount*channels, factor); + return 0; } -void ma_copy_and_apply_volume_factor_pcm_frames_s24(void* pPCMFramesOut, const void* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor) +int ma_strcmp(const char* str1, const char* str2) { - ma_copy_and_apply_volume_factor_s24(pPCMFramesOut, pPCMFramesIn, frameCount*channels, factor); -} + if (str1 == str2) return 0; -void ma_copy_and_apply_volume_factor_pcm_frames_s32(ma_int32* pPCMFramesOut, const ma_int32* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor) -{ - ma_copy_and_apply_volume_factor_s32(pPCMFramesOut, pPCMFramesIn, frameCount*channels, factor); -} + /* These checks differ from the standard implementation. It's not important, but I prefer it just for sanity. */ + if (str1 == NULL) return -1; + if (str2 == NULL) return 1; -void ma_copy_and_apply_volume_factor_pcm_frames_f32(float* pPCMFramesOut, const float* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor) -{ - ma_copy_and_apply_volume_factor_f32(pPCMFramesOut, pPCMFramesIn, frameCount*channels, factor); -} + for (;;) { + if (str1[0] == '\0') { + break; + } + if (str1[0] != str2[0]) { + break; + } -void ma_copy_and_apply_volume_factor_pcm_frames(void* pPCMFramesOut, const void* pPCMFramesIn, ma_uint32 frameCount, ma_format format, ma_uint32 channels, float factor) -{ - switch (format) - { - case ma_format_u8: ma_copy_and_apply_volume_factor_pcm_frames_u8 ((ma_uint8*)pPCMFramesOut, (const ma_uint8*)pPCMFramesIn, frameCount, channels, factor); return; - case ma_format_s16: ma_copy_and_apply_volume_factor_pcm_frames_s16((ma_int16*)pPCMFramesOut, (const ma_int16*)pPCMFramesIn, frameCount, channels, factor); return; - case ma_format_s24: ma_copy_and_apply_volume_factor_pcm_frames_s24( pPCMFramesOut, pPCMFramesIn, frameCount, channels, factor); return; - case ma_format_s32: ma_copy_and_apply_volume_factor_pcm_frames_s32((ma_int32*)pPCMFramesOut, (const ma_int32*)pPCMFramesIn, frameCount, channels, factor); return; - case ma_format_f32: ma_copy_and_apply_volume_factor_pcm_frames_f32( (float*)pPCMFramesOut, (const float*)pPCMFramesIn, frameCount, channels, factor); return; - default: return; /* Do nothing. */ + str1 += 1; + str2 += 1; } -} -void ma_apply_volume_factor_pcm_frames_u8(ma_uint8* pPCMFrames, ma_uint32 frameCount, ma_uint32 channels, float factor) -{ - ma_copy_and_apply_volume_factor_pcm_frames_u8(pPCMFrames, pPCMFrames, frameCount, channels, factor); + return ((unsigned char*)str1)[0] - ((unsigned char*)str2)[0]; } -void ma_apply_volume_factor_pcm_frames_s16(ma_int16* pPCMFrames, ma_uint32 frameCount, ma_uint32 channels, float factor) +int ma_strappend(char* dst, size_t dstSize, const char* srcA, const char* srcB) { - ma_copy_and_apply_volume_factor_pcm_frames_s16(pPCMFrames, pPCMFrames, frameCount, channels, factor); -} + int result; -void ma_apply_volume_factor_pcm_frames_s24(void* pPCMFrames, ma_uint32 frameCount, ma_uint32 channels, float factor) -{ - ma_copy_and_apply_volume_factor_pcm_frames_s24(pPCMFrames, pPCMFrames, frameCount, channels, factor); -} + result = ma_strncpy_s(dst, dstSize, srcA, (size_t)-1); + if (result != 0) { + return result; + } -void ma_apply_volume_factor_pcm_frames_s32(ma_int32* pPCMFrames, ma_uint32 frameCount, ma_uint32 channels, float factor) -{ - ma_copy_and_apply_volume_factor_pcm_frames_s32(pPCMFrames, pPCMFrames, frameCount, channels, factor); -} + result = ma_strncat_s(dst, dstSize, srcB, (size_t)-1); + if (result != 0) { + return result; + } -void ma_apply_volume_factor_pcm_frames_f32(float* pPCMFrames, ma_uint32 frameCount, ma_uint32 channels, float factor) -{ - ma_copy_and_apply_volume_factor_pcm_frames_f32(pPCMFrames, pPCMFrames, frameCount, channels, factor); + return result; } -void ma_apply_volume_factor_pcm_frames(void* pPCMFrames, ma_uint32 frameCount, ma_format format, ma_uint32 channels, float factor) +char* ma_copy_string(const char* src, const ma_allocation_callbacks* pAllocationCallbacks) { - ma_copy_and_apply_volume_factor_pcm_frames(pPCMFrames, pPCMFrames, frameCount, format, channels, factor); -} - + size_t sz = strlen(src)+1; + char* dst = (char*)ma_malloc(sz, pAllocationCallbacks); + if (dst == NULL) { + return NULL; + } -float ma_factor_to_gain_db(float factor) -{ - return (float)(20*log10(factor)); -} + ma_strcpy_s(dst, sz, src); -float ma_gain_db_to_factor(float gain) -{ - return (float)pow(10, gain/20.0); + return dst; } -static MA_INLINE void ma_device__on_data(ma_device* pDevice, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount) +static MA_INLINE void ma_copy_memory_64(void* dst, const void* src, ma_uint64 sizeInBytes) { - ma_device_callback_proc onData; - - onData = pDevice->onData; - if (onData) { - if (!pDevice->noPreZeroedOutputBuffer && pFramesOut != NULL) { - ma_zero_pcm_frames(pFramesOut, frameCount, pDevice->playback.format, pDevice->playback.channels); +#if 0xFFFFFFFFFFFFFFFF <= MA_SIZE_MAX + MA_COPY_MEMORY(dst, src, (size_t)sizeInBytes); +#else + while (sizeInBytes > 0) { + ma_uint64 bytesToCopyNow = sizeInBytes; + if (bytesToCopyNow > MA_SIZE_MAX) { + bytesToCopyNow = MA_SIZE_MAX; } - /* Volume control of input makes things a bit awkward because the input buffer is read-only. We'll need to use a temp buffer and loop in this case. */ - if (pFramesIn != NULL && pDevice->masterVolumeFactor < 1) { - ma_uint8 tempFramesIn[8192]; - ma_uint32 bpfCapture = ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); - ma_uint32 bpfPlayback = ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); - ma_uint32 totalFramesProcessed = 0; - while (totalFramesProcessed < frameCount) { - ma_uint32 framesToProcessThisIteration = frameCount - totalFramesProcessed; - if (framesToProcessThisIteration > sizeof(tempFramesIn)/bpfCapture) { - framesToProcessThisIteration = sizeof(tempFramesIn)/bpfCapture; - } - - ma_copy_and_apply_volume_factor_pcm_frames(tempFramesIn, ma_offset_ptr(pFramesIn, totalFramesProcessed*bpfCapture), framesToProcessThisIteration, pDevice->capture.format, pDevice->capture.channels, pDevice->masterVolumeFactor); + MA_COPY_MEMORY(dst, src, (size_t)bytesToCopyNow); /* Safe cast to size_t. */ - onData(pDevice, ma_offset_ptr(pFramesOut, totalFramesProcessed*bpfPlayback), tempFramesIn, framesToProcessThisIteration); + sizeInBytes -= bytesToCopyNow; + dst = ( void*)(( ma_uint8*)dst + bytesToCopyNow); + src = (const void*)((const ma_uint8*)src + bytesToCopyNow); + } +#endif +} - totalFramesProcessed += framesToProcessThisIteration; - } - } else { - onData(pDevice, pFramesOut, pFramesIn, frameCount); +static MA_INLINE void ma_zero_memory_64(void* dst, ma_uint64 sizeInBytes) +{ +#if 0xFFFFFFFFFFFFFFFF <= MA_SIZE_MAX + MA_ZERO_MEMORY(dst, (size_t)sizeInBytes); +#else + while (sizeInBytes > 0) { + ma_uint64 bytesToZeroNow = sizeInBytes; + if (bytesToZeroNow > MA_SIZE_MAX) { + bytesToZeroNow = MA_SIZE_MAX; } - /* Volume control and clipping for playback devices. */ - if (pFramesOut != NULL) { - if (pDevice->masterVolumeFactor < 1) { - if (pFramesIn == NULL) { /* <-- In full-duplex situations, the volume will have been applied to the input samples before the data callback. Applying it again post-callback will incorrectly compound it. */ - ma_apply_volume_factor_pcm_frames(pFramesOut, frameCount, pDevice->playback.format, pDevice->playback.channels, pDevice->masterVolumeFactor); - } - } + MA_ZERO_MEMORY(dst, (size_t)bytesToZeroNow); /* Safe cast to size_t. */ - if (!pDevice->noClip && pDevice->playback.format == ma_format_f32) { - ma_clip_pcm_frames_f32((float*)pFramesOut, frameCount, pDevice->playback.channels); - } - } + sizeInBytes -= bytesToZeroNow; + dst = (void*)((ma_uint8*)dst + bytesToZeroNow); } +#endif } -/* The callback for reading from the client -> DSP -> device. */ -ma_uint32 ma_device__on_read_from_client(ma_pcm_converter* pDSP, void* pFramesOut, ma_uint32 frameCount, void* pUserData) +/* Thanks to good old Bit Twiddling Hacks for this one: http://graphics.stanford.edu/~seander/bithacks.html#RoundUpPowerOf2 */ +static MA_INLINE unsigned int ma_next_power_of_2(unsigned int x) { - ma_device* pDevice = (ma_device*)pUserData; - - ma_assert(pDevice != NULL); - - ma_device__on_data(pDevice, pFramesOut, NULL, frameCount); + x--; + x |= x >> 1; + x |= x >> 2; + x |= x >> 4; + x |= x >> 8; + x |= x >> 16; + x++; - (void)pDSP; - return frameCount; + return x; } -/* The PCM converter callback for reading from a buffer. */ -ma_uint32 ma_device__pcm_converter__on_read_from_buffer_capture(ma_pcm_converter* pConverter, void* pFramesOut, ma_uint32 frameCount, void* pUserData) +static MA_INLINE unsigned int ma_prev_power_of_2(unsigned int x) { - ma_device* pDevice = (ma_device*)pUserData; - ma_uint32 framesToRead; - ma_uint32 bytesToRead; - - ma_assert(pDevice != NULL); - - if (pDevice->capture._dspFrameCount == 0) { - return 0; /* Nothing left. */ - } - - framesToRead = frameCount; - if (framesToRead > pDevice->capture._dspFrameCount) { - framesToRead = pDevice->capture._dspFrameCount; - } - - bytesToRead = framesToRead * ma_get_bytes_per_frame(pConverter->formatConverterIn.config.formatIn, pConverter->channelRouter.config.channelsIn); + return ma_next_power_of_2(x) >> 1; +} - /* pDevice->capture._dspFrames can be null in which case it should be treated as silence. */ - if (pDevice->capture._dspFrames != NULL) { - ma_copy_memory(pFramesOut, pDevice->capture._dspFrames, bytesToRead); - pDevice->capture._dspFrames += bytesToRead; +static MA_INLINE unsigned int ma_round_to_power_of_2(unsigned int x) +{ + unsigned int prev = ma_prev_power_of_2(x); + unsigned int next = ma_next_power_of_2(x); + if ((next - x) > (x - prev)) { + return prev; } else { - ma_zero_memory(pFramesOut, bytesToRead); + return next; } - - pDevice->capture._dspFrameCount -= framesToRead; - - return framesToRead; } -ma_uint32 ma_device__pcm_converter__on_read_from_buffer_playback(ma_pcm_converter* pConverter, void* pFramesOut, ma_uint32 frameCount, void* pUserData) +static MA_INLINE unsigned int ma_count_set_bits(unsigned int x) { - ma_device* pDevice = (ma_device*)pUserData; - ma_uint32 framesToRead; - ma_uint32 bytesToRead; - - ma_assert(pDevice != NULL); - - if (pDevice->playback._dspFrameCount == 0) { - return 0; /* Nothing left. */ - } - - framesToRead = frameCount; - if (framesToRead > pDevice->playback._dspFrameCount) { - framesToRead = pDevice->playback._dspFrameCount; - } - - bytesToRead = framesToRead * ma_get_bytes_per_frame(pConverter->formatConverterIn.config.formatIn, pConverter->channelRouter.config.channelsIn); - - /* pDevice->playback._dspFrames can be null in which case it should be treated as silence. */ - if (pDevice->playback._dspFrames != NULL) { - ma_copy_memory(pFramesOut, pDevice->playback._dspFrames, bytesToRead); - pDevice->playback._dspFrames += bytesToRead; - } else { - ma_zero_memory(pFramesOut, bytesToRead); + unsigned int count = 0; + while (x != 0) { + if (x & 1) { + count += 1; + } + + x = x >> 1; } - - pDevice->playback._dspFrameCount -= framesToRead; - return framesToRead; + return count; } -/* A helper function for reading sample data from the client. */ -static MA_INLINE void ma_device__read_frames_from_client(ma_device* pDevice, ma_uint32 frameCount, void* pFramesOut) +/* Clamps an f32 sample to -1..1 */ +static MA_INLINE float ma_clip_f32(float x) { - ma_assert(pDevice != NULL); - ma_assert(frameCount > 0); - ma_assert(pFramesOut != NULL); - - if (pDevice->playback.converter.isPassthrough) { - ma_device__on_data(pDevice, pFramesOut, NULL, frameCount); - } else { - ma_pcm_converter_read(&pDevice->playback.converter, pFramesOut, frameCount); - } + if (x < -1) return -1; + if (x > +1) return +1; + return x; } -/* A helper for sending sample data to the client. */ -static MA_INLINE void ma_device__send_frames_to_client(ma_device* pDevice, ma_uint32 frameCount, const void* pFrames) +static MA_INLINE float ma_mix_f32(float x, float y, float a) { - ma_assert(pDevice != NULL); - ma_assert(frameCount > 0); - ma_assert(pFrames != NULL); - - if (pDevice->capture.converter.isPassthrough) { - ma_device__on_data(pDevice, NULL, pFrames, frameCount); - } else { - ma_uint8 chunkBuffer[4096]; - ma_uint32 chunkFrameCount; - - pDevice->capture._dspFrameCount = frameCount; - pDevice->capture._dspFrames = (const ma_uint8*)pFrames; + return x*(1-a) + y*a; +} +static MA_INLINE float ma_mix_f32_fast(float x, float y, float a) +{ + float r0 = (y - x); + float r1 = r0*a; + return x + r1; + /*return x + (y - x)*a;*/ +} - chunkFrameCount = sizeof(chunkBuffer) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); - for (;;) { - ma_uint32 framesJustRead = (ma_uint32)ma_pcm_converter_read(&pDevice->capture.converter, chunkBuffer, chunkFrameCount); - if (framesJustRead == 0) { - break; - } +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE __m128 ma_mix_f32_fast__sse2(__m128 x, __m128 y, __m128 a) +{ + return _mm_add_ps(x, _mm_mul_ps(_mm_sub_ps(y, x), a)); +} +#endif +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE __m256 ma_mix_f32_fast__avx2(__m256 x, __m256 y, __m256 a) +{ + return _mm256_add_ps(x, _mm256_mul_ps(_mm256_sub_ps(y, x), a)); +} +#endif +#if defined(MA_SUPPORT_AVX512) +static MA_INLINE __m512 ma_mix_f32_fast__avx512(__m512 x, __m512 y, __m512 a) +{ + return _mm512_add_ps(x, _mm512_mul_ps(_mm512_sub_ps(y, x), a)); +} +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE float32x4_t ma_mix_f32_fast__neon(float32x4_t x, float32x4_t y, float32x4_t a) +{ + return vaddq_f32(x, vmulq_f32(vsubq_f32(y, x), a)); +} +#endif - ma_device__on_data(pDevice, NULL, chunkBuffer, framesJustRead); - if (framesJustRead < chunkFrameCount) { - break; - } - } - } +static MA_INLINE double ma_mix_f64(double x, double y, double a) +{ + return x*(1-a) + y*a; +} +static MA_INLINE double ma_mix_f64_fast(double x, double y, double a) +{ + return x + (y - x)*a; } -static MA_INLINE ma_result ma_device__handle_duplex_callback_capture(ma_device* pDevice, ma_uint32 frameCount, const void* pFramesInInternalFormat, ma_pcm_rb* pRB) +static MA_INLINE float ma_scale_to_range_f32(float x, float lo, float hi) { - ma_result result; + return lo + x*(hi-lo); +} - ma_assert(pDevice != NULL); - ma_assert(frameCount > 0); - ma_assert(pFramesInInternalFormat != NULL); - ma_assert(pRB != NULL); - - pDevice->capture._dspFrameCount = (ma_uint32)frameCount; - pDevice->capture._dspFrames = (const ma_uint8*)pFramesInInternalFormat; - /* Write to the ring buffer. The ring buffer is in the external format. */ +/* +Greatest common factor using Euclid's algorithm iteratively. +*/ +static MA_INLINE ma_uint32 ma_gcf_u32(ma_uint32 a, ma_uint32 b) +{ for (;;) { - ma_uint32 framesProcessed; - ma_uint32 framesToProcess = 256; - void* pFramesInExternalFormat; - - result = ma_pcm_rb_acquire_write(pRB, &framesToProcess, &pFramesInExternalFormat); - if (result != MA_SUCCESS) { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "Failed to acquire capture PCM frames from ring buffer.", result); - break; - } - - if (framesToProcess == 0) { - if (ma_pcm_rb_pointer_disance(pRB) == (ma_int32)ma_pcm_rb_get_subbuffer_size(pRB)) { - break; /* Overrun. Not enough room in the ring buffer for input frame. Excess frames are dropped. */ - } - } - - /* Convert. */ - framesProcessed = (ma_uint32)ma_pcm_converter_read(&pDevice->capture.converter, pFramesInExternalFormat, framesToProcess); - - result = ma_pcm_rb_commit_write(pRB, framesProcessed, pFramesInExternalFormat); - if (result != MA_SUCCESS) { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "Failed to commit capture PCM frames to ring buffer.", result); + if (b == 0) { break; - } - - if (framesProcessed < framesToProcess) { - break; /* Done. */ + } else { + ma_uint32 t = a; + a = b; + b = t % a; } } - return MA_SUCCESS; + return a; } -static MA_INLINE ma_result ma_device__handle_duplex_callback_playback(ma_device* pDevice, ma_uint32 frameCount, void* pFramesInInternalFormat, ma_pcm_rb* pRB) -{ - ma_result result; - ma_uint8 playbackFramesInExternalFormat[4096]; - ma_uint8 silentInputFrames[4096]; - ma_uint32 totalFramesToReadFromClient; - ma_uint32 totalFramesReadFromClient; - - ma_assert(pDevice != NULL); - ma_assert(frameCount > 0); - ma_assert(pFramesInInternalFormat != NULL); - ma_assert(pRB != NULL); - - /* - Sitting in the ring buffer should be captured data from the capture callback in external format. If there's not enough data in there for - the whole frameCount frames we just use silence instead for the input data. - */ - ma_zero_memory(silentInputFrames, sizeof(silentInputFrames)); - - /* We need to calculate how many output frames are required to be read from the client to completely fill frameCount internal frames. */ - totalFramesToReadFromClient = (ma_uint32)ma_calculate_frame_count_after_src(pDevice->sampleRate, pDevice->playback.internalSampleRate, frameCount); /* ma_pcm_converter_get_required_input_frame_count(&pDevice->playback.converter, (ma_uint32)frameCount); */ - totalFramesReadFromClient = 0; - while (totalFramesReadFromClient < totalFramesToReadFromClient && ma_device_is_started(pDevice)) { - ma_uint32 framesRemainingFromClient; - ma_uint32 framesToProcessFromClient; - ma_uint32 inputFrameCount; - void* pInputFrames; - - framesRemainingFromClient = (totalFramesToReadFromClient - totalFramesReadFromClient); - framesToProcessFromClient = sizeof(playbackFramesInExternalFormat) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); - if (framesToProcessFromClient > framesRemainingFromClient) { - framesToProcessFromClient = framesRemainingFromClient; - } - - /* We need to grab captured samples before firing the callback. If there's not enough input samples we just pass silence. */ - inputFrameCount = framesToProcessFromClient; - result = ma_pcm_rb_acquire_read(pRB, &inputFrameCount, &pInputFrames); - if (result == MA_SUCCESS) { - if (inputFrameCount > 0) { - /* Use actual input frames. */ - ma_device__on_data(pDevice, playbackFramesInExternalFormat, pInputFrames, inputFrameCount); - } else { - if (ma_pcm_rb_pointer_disance(pRB) == 0) { - break; /* Underrun. */ - } - } - - /* We're done with the captured samples. */ - result = ma_pcm_rb_commit_read(pRB, inputFrameCount, pInputFrames); - if (result != MA_SUCCESS) { - break; /* Don't know what to do here... Just abandon ship. */ - } - } else { - /* Use silent input frames. */ - inputFrameCount = ma_min( - sizeof(playbackFramesInExternalFormat) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels), - sizeof(silentInputFrames) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels) - ); - ma_device__on_data(pDevice, playbackFramesInExternalFormat, silentInputFrames, inputFrameCount); - } +/* +Random Number Generation - /* We have samples in external format so now we need to convert to internal format and output to the device. */ - pDevice->playback._dspFrameCount = inputFrameCount; - pDevice->playback._dspFrames = (const ma_uint8*)playbackFramesInExternalFormat; - ma_pcm_converter_read(&pDevice->playback.converter, pFramesInInternalFormat, inputFrameCount); +miniaudio uses the LCG random number generation algorithm. This is good enough for audio. - totalFramesReadFromClient += inputFrameCount; - pFramesInInternalFormat = ma_offset_ptr(pFramesInInternalFormat, inputFrameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); - } +Note that miniaudio's LCG implementation uses global state which is _not_ thread-local. When this is called across +multiple threads, results will be unpredictable. However, it won't crash and results will still be random enough +for miniaudio's purposes. +*/ +#define MA_LCG_M 2147483647 +#define MA_LCG_A 48271 +#define MA_LCG_C 0 +static ma_int32 g_maLCG = 4321; /* Non-zero initial seed. Use ma_seed() to use an explicit seed. */ - return MA_SUCCESS; +static MA_INLINE void ma_seed(ma_int32 seed) +{ + g_maLCG = seed; } -/* A helper for changing the state of the device. */ -static MA_INLINE void ma_device__set_state(ma_device* pDevice, ma_uint32 newState) +static MA_INLINE ma_int32 ma_rand_s32() { - ma_atomic_exchange_32(&pDevice->state, newState); + ma_int32 lcg = g_maLCG; + ma_int32 r = (MA_LCG_A * lcg + MA_LCG_C) % MA_LCG_M; + g_maLCG = r; + return r; } -/* A helper for getting the state of the device. */ -static MA_INLINE ma_uint32 ma_device__get_state(ma_device* pDevice) +static MA_INLINE ma_uint32 ma_rand_u32() { - ma_uint32 state; - ma_atomic_exchange_32(&state, pDevice->state); - - return state; + return (ma_uint32)ma_rand_s32(); } - -#ifdef MA_WIN32 - GUID MA_GUID_KSDATAFORMAT_SUBTYPE_PCM = {0x00000001, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}}; - GUID MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = {0x00000003, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}}; - /*GUID MA_GUID_KSDATAFORMAT_SUBTYPE_ALAW = {0x00000006, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}};*/ - /*GUID MA_GUID_KSDATAFORMAT_SUBTYPE_MULAW = {0x00000007, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}};*/ -#endif - - -ma_bool32 ma_context__device_id_equal(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) +static MA_INLINE double ma_rand_f64() { - ma_assert(pContext != NULL); + return ma_rand_s32() / (double)0x7FFFFFFF; +} - if (pID0 == pID1) return MA_TRUE; +static MA_INLINE float ma_rand_f32() +{ + return (float)ma_rand_f64(); +} - if ((pID0 == NULL && pID1 != NULL) || - (pID0 != NULL && pID1 == NULL)) { - return MA_FALSE; - } +static MA_INLINE float ma_rand_range_f32(float lo, float hi) +{ + return ma_scale_to_range_f32(ma_rand_f32(), lo, hi); +} - if (pContext->onDeviceIDEqual) { - return pContext->onDeviceIDEqual(pContext, pID0, pID1); +static MA_INLINE ma_int32 ma_rand_range_s32(ma_int32 lo, ma_int32 hi) +{ + if (lo == hi) { + return lo; } - return MA_FALSE; + return lo + ma_rand_u32() / (0xFFFFFFFF / (hi - lo + 1) + 1); } -typedef struct +static MA_INLINE float ma_dither_f32_rectangle(float ditherMin, float ditherMax) { - ma_device_type deviceType; - const ma_device_id* pDeviceID; - char* pName; - size_t nameBufferSize; - ma_bool32 foundDevice; -} ma_context__try_get_device_name_by_id__enum_callback_data; + return ma_rand_range_f32(ditherMin, ditherMax); +} -ma_bool32 ma_context__try_get_device_name_by_id__enum_callback(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pDeviceInfo, void* pUserData) +static MA_INLINE float ma_dither_f32_triangle(float ditherMin, float ditherMax) { - ma_context__try_get_device_name_by_id__enum_callback_data* pData = (ma_context__try_get_device_name_by_id__enum_callback_data*)pUserData; - ma_assert(pData != NULL); + float a = ma_rand_range_f32(ditherMin, 0); + float b = ma_rand_range_f32(0, ditherMax); + return a + b; +} - if (pData->deviceType == deviceType) { - if (pContext->onDeviceIDEqual(pContext, pData->pDeviceID, &pDeviceInfo->id)) { - ma_strncpy_s(pData->pName, pData->nameBufferSize, pDeviceInfo->name, (size_t)-1); - pData->foundDevice = MA_TRUE; - } +static MA_INLINE float ma_dither_f32(ma_dither_mode ditherMode, float ditherMin, float ditherMax) +{ + if (ditherMode == ma_dither_mode_rectangle) { + return ma_dither_f32_rectangle(ditherMin, ditherMax); + } + if (ditherMode == ma_dither_mode_triangle) { + return ma_dither_f32_triangle(ditherMin, ditherMax); } - return !pData->foundDevice; + return 0; } -/* -Generic function for retrieving the name of a device by it's ID. - -This function simply enumerates every device and then retrieves the name of the first device that has the same ID. -*/ -ma_result ma_context__try_get_device_name_by_id(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, char* pName, size_t nameBufferSize) +static MA_INLINE ma_int32 ma_dither_s32(ma_dither_mode ditherMode, ma_int32 ditherMin, ma_int32 ditherMax) { - ma_result result; - ma_context__try_get_device_name_by_id__enum_callback_data data; - - ma_assert(pContext != NULL); - ma_assert(pName != NULL); - - if (pDeviceID == NULL) { - return MA_NO_DEVICE; + if (ditherMode == ma_dither_mode_rectangle) { + ma_int32 a = ma_rand_range_s32(ditherMin, ditherMax); + return a; } - - data.deviceType = deviceType; - data.pDeviceID = pDeviceID; - data.pName = pName; - data.nameBufferSize = nameBufferSize; - data.foundDevice = MA_FALSE; - result = ma_context_enumerate_devices(pContext, ma_context__try_get_device_name_by_id__enum_callback, &data); - if (result != MA_SUCCESS) { - return result; + if (ditherMode == ma_dither_mode_triangle) { + ma_int32 a = ma_rand_range_s32(ditherMin, 0); + ma_int32 b = ma_rand_range_s32(0, ditherMax); + return a + b; } - if (!data.foundDevice) { - return MA_NO_DEVICE; - } else { - return MA_SUCCESS; - } + return 0; } -ma_uint32 ma_get_format_priority_index(ma_format format) /* Lower = better. */ -{ - ma_uint32 i; - for (i = 0; i < ma_countof(g_maFormatPriorities); ++i) { - if (g_maFormatPriorities[i] == format) { - return i; - } - } +/****************************************************************************** - /* Getting here means the format could not be found or is equal to ma_format_unknown. */ - return (ma_uint32)-1; -} +Atomics -void ma_device__post_init_setup(ma_device* pDevice, ma_device_type deviceType); +******************************************************************************/ +#if defined(__clang__) + #if defined(__has_builtin) + #if __has_builtin(__sync_swap) + #define MA_HAS_SYNC_SWAP + #endif + #endif +#elif defined(__GNUC__) + #if __GNUC__ > 4 || (__GNUC__ == 4 && __GNUC__ >= 7) + #define MA_HAS_GNUC_ATOMICS + #endif +#endif +#if defined(_WIN32) && !defined(__GNUC__) && !defined(__clang__) +#define ma_memory_barrier() MemoryBarrier() +#define ma_atomic_exchange_32(a, b) InterlockedExchange((LONG*)a, (LONG)b) +#define ma_atomic_exchange_64(a, b) InterlockedExchange64((LONGLONG*)a, (LONGLONG)b) +#define ma_atomic_increment_32(a) InterlockedIncrement((LONG*)a) +#define ma_atomic_decrement_32(a) InterlockedDecrement((LONG*)a) +#else +#define ma_memory_barrier() __sync_synchronize() +#if defined(MA_HAS_SYNC_SWAP) + #define ma_atomic_exchange_32(a, b) __sync_swap(a, b) + #define ma_atomic_exchange_64(a, b) __sync_swap(a, b) +#elif defined(MA_HAS_GNUC_ATOMICS) + #define ma_atomic_exchange_32(a, b) (void)__atomic_exchange_n(a, b, __ATOMIC_ACQ_REL) + #define ma_atomic_exchange_64(a, b) (void)__atomic_exchange_n(a, b, __ATOMIC_ACQ_REL) +#else + #define ma_atomic_exchange_32(a, b) __sync_synchronize(); (void)__sync_lock_test_and_set(a, b) + #define ma_atomic_exchange_64(a, b) __sync_synchronize(); (void)__sync_lock_test_and_set(a, b) +#endif +#define ma_atomic_increment_32(a) __sync_add_and_fetch(a, 1) +#define ma_atomic_decrement_32(a) __sync_sub_and_fetch(a, 1) +#endif -/******************************************************************************* +#ifdef MA_64BIT +#define ma_atomic_exchange_ptr ma_atomic_exchange_64 +#endif +#ifdef MA_32BIT +#define ma_atomic_exchange_ptr ma_atomic_exchange_32 +#endif -Null Backend -*******************************************************************************/ -#ifdef MA_HAS_NULL +static void* ma__malloc_default(size_t sz, void* pUserData) +{ + (void)pUserData; + return MA_MALLOC(sz); +} -#define MA_DEVICE_OP_NONE__NULL 0 -#define MA_DEVICE_OP_START__NULL 1 -#define MA_DEVICE_OP_SUSPEND__NULL 2 -#define MA_DEVICE_OP_KILL__NULL 3 +static void* ma__realloc_default(void* p, size_t sz, void* pUserData) +{ + (void)pUserData; + return MA_REALLOC(p, sz); +} -ma_thread_result MA_THREADCALL ma_device_thread__null(void* pData) +static void ma__free_default(void* p, void* pUserData) { - ma_device* pDevice = (ma_device*)pData; - ma_assert(pDevice != NULL); + (void)pUserData; + MA_FREE(p); +} - for (;;) { /* Keep the thread alive until the device is uninitialized. */ - /* Wait for an operation to be requested. */ - ma_event_wait(&pDevice->null_device.operationEvent); - /* At this point an event should have been triggered. */ +static void* ma__malloc_from_callbacks(size_t sz, const ma_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks == NULL) { + return NULL; + } - /* Starting the device needs to put the thread into a loop. */ - if (pDevice->null_device.operation == MA_DEVICE_OP_START__NULL) { - ma_atomic_exchange_32(&pDevice->null_device.operation, MA_DEVICE_OP_NONE__NULL); + if (pAllocationCallbacks->onMalloc != NULL) { + return pAllocationCallbacks->onMalloc(sz, pAllocationCallbacks->pUserData); + } - /* Reset the timer just in case. */ - ma_timer_init(&pDevice->null_device.timer); + /* Try using realloc(). */ + if (pAllocationCallbacks->onRealloc != NULL) { + return pAllocationCallbacks->onRealloc(NULL, sz, pAllocationCallbacks->pUserData); + } - /* Keep looping until an operation has been requested. */ - while (pDevice->null_device.operation != MA_DEVICE_OP_NONE__NULL && pDevice->null_device.operation != MA_DEVICE_OP_START__NULL) { - ma_sleep(10); /* Don't hog the CPU. */ - } + return NULL; +} - /* Getting here means a suspend or kill operation has been requested. */ - ma_atomic_exchange_32(&pDevice->null_device.operationResult, MA_SUCCESS); - ma_event_signal(&pDevice->null_device.operationCompletionEvent); - continue; - } +static void* ma__realloc_from_callbacks(void* p, size_t szNew, size_t szOld, const ma_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks == NULL) { + return NULL; + } - /* Suspending the device means we need to stop the timer and just continue the loop. */ - if (pDevice->null_device.operation == MA_DEVICE_OP_SUSPEND__NULL) { - ma_atomic_exchange_32(&pDevice->null_device.operation, MA_DEVICE_OP_NONE__NULL); + if (pAllocationCallbacks->onRealloc != NULL) { + return pAllocationCallbacks->onRealloc(p, szNew, pAllocationCallbacks->pUserData); + } - /* We need to add the current run time to the prior run time, then reset the timer. */ - pDevice->null_device.priorRunTime += ma_timer_get_time_in_seconds(&pDevice->null_device.timer); - ma_timer_init(&pDevice->null_device.timer); + /* Try emulating realloc() in terms of malloc()/free(). */ + if (pAllocationCallbacks->onMalloc != NULL && pAllocationCallbacks->onFree != NULL) { + void* p2; - /* We're done. */ - ma_atomic_exchange_32(&pDevice->null_device.operationResult, MA_SUCCESS); - ma_event_signal(&pDevice->null_device.operationCompletionEvent); - continue; + p2 = pAllocationCallbacks->onMalloc(szNew, pAllocationCallbacks->pUserData); + if (p2 == NULL) { + return NULL; } - /* Killing the device means we need to get out of this loop so that this thread can terminate. */ - if (pDevice->null_device.operation == MA_DEVICE_OP_KILL__NULL) { - ma_atomic_exchange_32(&pDevice->null_device.operation, MA_DEVICE_OP_NONE__NULL); - ma_atomic_exchange_32(&pDevice->null_device.operationResult, MA_SUCCESS); - ma_event_signal(&pDevice->null_device.operationCompletionEvent); - break; + if (p != NULL) { + MA_COPY_MEMORY(p2, p, szOld); + pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData); } - /* Getting a signal on a "none" operation probably means an error. Return invalid operation. */ - if (pDevice->null_device.operation == MA_DEVICE_OP_NONE__NULL) { - ma_assert(MA_FALSE); /* <-- Trigger this in debug mode to ensure developers are aware they're doing something wrong (or there's a bug in a miniaudio). */ - ma_atomic_exchange_32(&pDevice->null_device.operationResult, MA_INVALID_OPERATION); - ma_event_signal(&pDevice->null_device.operationCompletionEvent); - continue; /* Continue the loop. Don't terminate. */ - } + return p2; } - return (ma_thread_result)0; + return NULL; } -ma_result ma_device_do_operation__null(ma_device* pDevice, ma_uint32 operation) +static MA_INLINE void* ma__calloc_from_callbacks(size_t sz, const ma_allocation_callbacks* pAllocationCallbacks) { - ma_atomic_exchange_32(&pDevice->null_device.operation, operation); - if (!ma_event_signal(&pDevice->null_device.operationEvent)) { - return MA_ERROR; - } - - if (!ma_event_wait(&pDevice->null_device.operationCompletionEvent)) { - return MA_ERROR; + void* p = ma__malloc_from_callbacks(sz, pAllocationCallbacks); + if (p != NULL) { + MA_ZERO_MEMORY(p, sz); } - return pDevice->null_device.operationResult; + return p; } -ma_uint64 ma_device_get_total_run_time_in_frames__null(ma_device* pDevice) +static void ma__free_from_callbacks(void* p, const ma_allocation_callbacks* pAllocationCallbacks) { - ma_uint32 internalSampleRate; - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - internalSampleRate = pDevice->capture.internalSampleRate; - } else { - internalSampleRate = pDevice->playback.internalSampleRate; + if (p == NULL || pAllocationCallbacks == NULL) { + return; } - - return (ma_uint64)((pDevice->null_device.priorRunTime + ma_timer_get_time_in_seconds(&pDevice->null_device.timer)) * internalSampleRate); + if (pAllocationCallbacks->onFree != NULL) { + pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData); + } } -ma_bool32 ma_context_is_device_id_equal__null(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) +static ma_allocation_callbacks ma_allocation_callbacks_init_default() { - ma_assert(pContext != NULL); - ma_assert(pID0 != NULL); - ma_assert(pID1 != NULL); - (void)pContext; + ma_allocation_callbacks callbacks; + callbacks.pUserData = NULL; + callbacks.onMalloc = ma__malloc_default; + callbacks.onRealloc = ma__realloc_default; + callbacks.onFree = ma__free_default; - return pID0->nullbackend == pID1->nullbackend; + return callbacks; } -ma_result ma_context_enumerate_devices__null(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +static ma_result ma_allocation_callbacks_init_copy(ma_allocation_callbacks* pDst, const ma_allocation_callbacks* pSrc) { - ma_bool32 cbResult = MA_TRUE; - - ma_assert(pContext != NULL); - ma_assert(callback != NULL); - - /* Playback. */ - if (cbResult) { - ma_device_info deviceInfo; - ma_zero_object(&deviceInfo); - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), "NULL Playback Device", (size_t)-1); - cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); + if (pDst == NULL) { + return MA_INVALID_ARGS; } - /* Capture. */ - if (cbResult) { - ma_device_info deviceInfo; - ma_zero_object(&deviceInfo); - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), "NULL Capture Device", (size_t)-1); - cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); + if (pSrc == NULL) { + *pDst = ma_allocation_callbacks_init_default(); + } else { + if (pSrc->pUserData == NULL && pSrc->onFree == NULL && pSrc->onMalloc == NULL && pSrc->onRealloc == NULL) { + *pDst = ma_allocation_callbacks_init_default(); + } else { + if (pSrc->onFree == NULL || (pSrc->onMalloc == NULL && pSrc->onRealloc == NULL)) { + return MA_INVALID_ARGS; /* Invalid allocation callbacks. */ + } else { + *pDst = *pSrc; + } + } } return MA_SUCCESS; } -ma_result ma_context_get_device_info__null(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) -{ - ma_uint32 iFormat; - ma_assert(pContext != NULL); +ma_uint64 ma_calculate_frame_count_after_resampling(ma_uint32 sampleRateOut, ma_uint32 sampleRateIn, ma_uint64 frameCountIn) +{ + /* For robustness we're going to use a resampler object to calculate this since that already has a way of calculating this. */ + ma_result result; + ma_uint64 frameCountOut; + ma_resampler_config config; + ma_resampler resampler; - if (pDeviceID != NULL && pDeviceID->nullbackend != 0) { - return MA_NO_DEVICE; /* Don't know the device. */ + config = ma_resampler_config_init(ma_format_s16, 1, sampleRateIn, sampleRateOut, ma_resample_algorithm_linear); + result = ma_resampler_init(&config, &resampler); + if (result != MA_SUCCESS) { + return 0; } - /* Name / Description */ - if (deviceType == ma_device_type_playback) { - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), "NULL Playback Device", (size_t)-1); - } else { - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), "NULL Capture Device", (size_t)-1); + frameCountOut = ma_resampler_get_expected_output_frame_count(&resampler, frameCountIn); + + ma_resampler_uninit(&resampler); + return frameCountOut; +} + +#ifndef MA_DATA_CONVERTER_STACK_BUFFER_SIZE +#define MA_DATA_CONVERTER_STACK_BUFFER_SIZE 4096 +#endif + +/************************************************************************************************************************************************************ +************************************************************************************************************************************************************* + +DEVICE I/O +========== + +************************************************************************************************************************************************************* +************************************************************************************************************************************************************/ +#ifndef MA_NO_DEVICE_IO +/* +Unfortunately using runtime linking for pthreads causes problems. This has occurred for me when testing on FreeBSD. When +using runtime linking, deadlocks can occur (for me it happens when loading data from fread()). It turns out that doing +compile-time linking fixes this. I'm not sure why this happens, but the safest way I can think of to fix this is to simply +disable runtime linking by default. To enable runtime linking, #define this before the implementation of this file. I am +not officially supporting this, but I'm leaving it here in case it's useful for somebody, somewhere. +*/ +/*#define MA_USE_RUNTIME_LINKING_FOR_PTHREAD*/ + +/* Disable run-time linking on certain backends. */ +#ifndef MA_NO_RUNTIME_LINKING + #if defined(MA_ANDROID) || defined(MA_EMSCRIPTEN) + #define MA_NO_RUNTIME_LINKING + #endif +#endif + +/* +Check if we have the necessary development packages for each backend at the top so we can use this to determine whether or not +certain unused functions and variables can be excluded from the build to avoid warnings. +*/ +#ifdef MA_ENABLE_WASAPI + #define MA_HAS_WASAPI /* Every compiler should support WASAPI */ +#endif +#ifdef MA_ENABLE_DSOUND + #define MA_HAS_DSOUND /* Every compiler should support DirectSound. */ +#endif +#ifdef MA_ENABLE_WINMM + #define MA_HAS_WINMM /* Every compiler I'm aware of supports WinMM. */ +#endif +#ifdef MA_ENABLE_ALSA + #define MA_HAS_ALSA + #ifdef MA_NO_RUNTIME_LINKING + #ifdef __has_include + #if !__has_include() + #undef MA_HAS_ALSA + #endif + #endif + #endif +#endif +#ifdef MA_ENABLE_PULSEAUDIO + #define MA_HAS_PULSEAUDIO + #ifdef MA_NO_RUNTIME_LINKING + #ifdef __has_include + #if !__has_include() + #undef MA_HAS_PULSEAUDIO + #endif + #endif + #endif +#endif +#ifdef MA_ENABLE_JACK + #define MA_HAS_JACK + #ifdef MA_NO_RUNTIME_LINKING + #ifdef __has_include + #if !__has_include() + #undef MA_HAS_JACK + #endif + #endif + #endif +#endif +#ifdef MA_ENABLE_COREAUDIO + #define MA_HAS_COREAUDIO +#endif +#ifdef MA_ENABLE_SNDIO + #define MA_HAS_SNDIO +#endif +#ifdef MA_ENABLE_AUDIO4 + #define MA_HAS_AUDIO4 +#endif +#ifdef MA_ENABLE_OSS + #define MA_HAS_OSS +#endif +#ifdef MA_ENABLE_AAUDIO + #define MA_HAS_AAUDIO +#endif +#ifdef MA_ENABLE_OPENSL + #define MA_HAS_OPENSL +#endif +#ifdef MA_ENABLE_WEBAUDIO + #define MA_HAS_WEBAUDIO +#endif +#ifdef MA_ENABLE_NULL + #define MA_HAS_NULL /* Everything supports the null backend. */ +#endif + +const char* ma_get_backend_name(ma_backend backend) +{ + switch (backend) + { + case ma_backend_wasapi: return "WASAPI"; + case ma_backend_dsound: return "DirectSound"; + case ma_backend_winmm: return "WinMM"; + case ma_backend_coreaudio: return "Core Audio"; + case ma_backend_sndio: return "sndio"; + case ma_backend_audio4: return "audio(4)"; + case ma_backend_oss: return "OSS"; + case ma_backend_pulseaudio: return "PulseAudio"; + case ma_backend_alsa: return "ALSA"; + case ma_backend_jack: return "JACK"; + case ma_backend_aaudio: return "AAudio"; + case ma_backend_opensl: return "OpenSL|ES"; + case ma_backend_webaudio: return "Web Audio"; + case ma_backend_null: return "Null"; + default: return "Unknown"; } +} - /* Support everything on the null backend. */ - pDeviceInfo->formatCount = ma_format_count - 1; /* Minus one because we don't want to include ma_format_unknown. */ - for (iFormat = 0; iFormat < pDeviceInfo->formatCount; ++iFormat) { - pDeviceInfo->formats[iFormat] = (ma_format)(iFormat + 1); /* +1 to skip over ma_format_unknown. */ +ma_bool32 ma_is_loopback_supported(ma_backend backend) +{ + switch (backend) + { + case ma_backend_wasapi: return MA_TRUE; + case ma_backend_dsound: return MA_FALSE; + case ma_backend_winmm: return MA_FALSE; + case ma_backend_coreaudio: return MA_FALSE; + case ma_backend_sndio: return MA_FALSE; + case ma_backend_audio4: return MA_FALSE; + case ma_backend_oss: return MA_FALSE; + case ma_backend_pulseaudio: return MA_FALSE; + case ma_backend_alsa: return MA_FALSE; + case ma_backend_jack: return MA_FALSE; + case ma_backend_aaudio: return MA_FALSE; + case ma_backend_opensl: return MA_FALSE; + case ma_backend_webaudio: return MA_FALSE; + case ma_backend_null: return MA_FALSE; + default: return MA_FALSE; } - - pDeviceInfo->minChannels = 1; - pDeviceInfo->maxChannels = MA_MAX_CHANNELS; - pDeviceInfo->minSampleRate = MA_SAMPLE_RATE_8000; - pDeviceInfo->maxSampleRate = MA_SAMPLE_RATE_384000; - - (void)pContext; - (void)shareMode; - return MA_SUCCESS; } -void ma_device_uninit__null(ma_device* pDevice) -{ - ma_assert(pDevice != NULL); - /* Keep it clean and wait for the device thread to finish before returning. */ - ma_device_do_operation__null(pDevice, MA_DEVICE_OP_KILL__NULL); +#ifdef MA_WIN32 + #define MA_THREADCALL WINAPI + typedef unsigned long ma_thread_result; +#else + #define MA_THREADCALL + typedef void* ma_thread_result; +#endif +typedef ma_thread_result (MA_THREADCALL * ma_thread_entry_proc)(void* pData); - /* At this point the loop in the device thread is as good as terminated so we can uninitialize our events. */ - ma_event_uninit(&pDevice->null_device.operationCompletionEvent); - ma_event_uninit(&pDevice->null_device.operationEvent); -} +#ifdef MA_WIN32 +typedef HRESULT (WINAPI * MA_PFN_CoInitializeEx)(LPVOID pvReserved, DWORD dwCoInit); +typedef void (WINAPI * MA_PFN_CoUninitialize)(); +typedef HRESULT (WINAPI * MA_PFN_CoCreateInstance)(REFCLSID rclsid, LPUNKNOWN pUnkOuter, DWORD dwClsContext, REFIID riid, LPVOID *ppv); +typedef void (WINAPI * MA_PFN_CoTaskMemFree)(LPVOID pv); +typedef HRESULT (WINAPI * MA_PFN_PropVariantClear)(PROPVARIANT *pvar); +typedef int (WINAPI * MA_PFN_StringFromGUID2)(const GUID* const rguid, LPOLESTR lpsz, int cchMax); -ma_result ma_device_init__null(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) -{ - ma_result result; - ma_uint32 bufferSizeInFrames; +typedef HWND (WINAPI * MA_PFN_GetForegroundWindow)(); +typedef HWND (WINAPI * MA_PFN_GetDesktopWindow)(); - ma_assert(pDevice != NULL); +/* Microsoft documents these APIs as returning LSTATUS, but the Win32 API shipping with some compilers do not define it. It's just a LONG. */ +typedef LONG (WINAPI * MA_PFN_RegOpenKeyExA)(HKEY hKey, LPCSTR lpSubKey, DWORD ulOptions, REGSAM samDesired, PHKEY phkResult); +typedef LONG (WINAPI * MA_PFN_RegCloseKey)(HKEY hKey); +typedef LONG (WINAPI * MA_PFN_RegQueryValueExA)(HKEY hKey, LPCSTR lpValueName, LPDWORD lpReserved, LPDWORD lpType, LPBYTE lpData, LPDWORD lpcbData); +#endif - ma_zero_object(&pDevice->null_device); - if (pConfig->deviceType == ma_device_type_loopback) { - return MA_DEVICE_TYPE_NOT_SUPPORTED; - } +#define MA_STATE_UNINITIALIZED 0 +#define MA_STATE_STOPPED 1 /* The device's default state after initialization. */ +#define MA_STATE_STARTED 2 /* The worker thread is in it's main loop waiting for the driver to request or deliver audio data. */ +#define MA_STATE_STARTING 3 /* Transitioning from a stopped state to started. */ +#define MA_STATE_STOPPING 4 /* Transitioning from a started state to stopped. */ - bufferSizeInFrames = pConfig->bufferSizeInFrames; - if (bufferSizeInFrames == 0) { - bufferSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->bufferSizeInMilliseconds, pConfig->sampleRate); - } +#define MA_DEFAULT_PLAYBACK_DEVICE_NAME "Default Playback Device" +#define MA_DEFAULT_CAPTURE_DEVICE_NAME "Default Capture Device" - if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { - ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), "NULL Capture Device", (size_t)-1); - pDevice->capture.internalFormat = pConfig->capture.format; - pDevice->capture.internalChannels = pConfig->capture.channels; - ma_channel_map_copy(pDevice->capture.internalChannelMap, pConfig->capture.channelMap, pConfig->capture.channels); - pDevice->capture.internalBufferSizeInFrames = bufferSizeInFrames; - pDevice->capture.internalPeriods = pConfig->periods; - } - if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { - ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), "NULL Playback Device", (size_t)-1); - pDevice->playback.internalFormat = pConfig->playback.format; - pDevice->playback.internalChannels = pConfig->playback.channels; - ma_channel_map_copy(pDevice->playback.internalChannelMap, pConfig->playback.channelMap, pConfig->playback.channels); - pDevice->playback.internalBufferSizeInFrames = bufferSizeInFrames; - pDevice->playback.internalPeriods = pConfig->periods; - } - /* - In order to get timing right, we need to create a thread that does nothing but keeps track of the timer. This timer is started when the - first period is "written" to it, and then stopped in ma_device_stop__null(). - */ - result = ma_event_init(pContext, &pDevice->null_device.operationEvent); - if (result != MA_SUCCESS) { - return result; +const char* ma_log_level_to_string(ma_uint32 logLevel) +{ + switch (logLevel) + { + case MA_LOG_LEVEL_VERBOSE: return ""; + case MA_LOG_LEVEL_INFO: return "INFO"; + case MA_LOG_LEVEL_WARNING: return "WARNING"; + case MA_LOG_LEVEL_ERROR: return "ERROR"; + default: return "ERROR"; } +} - result = ma_event_init(pContext, &pDevice->null_device.operationCompletionEvent); - if (result != MA_SUCCESS) { - return result; +/* Posts a log message. */ +static void ma_post_log_message(ma_context* pContext, ma_device* pDevice, ma_uint32 logLevel, const char* message) +{ + if (pContext == NULL) { + return; } + +#if defined(MA_LOG_LEVEL) + if (logLevel <= MA_LOG_LEVEL) { + ma_log_proc onLog; - result = ma_thread_create(pContext, &pDevice->thread, ma_device_thread__null, pDevice); - if (result != MA_SUCCESS) { - return result; + #if defined(MA_DEBUG_OUTPUT) + if (logLevel <= MA_LOG_LEVEL) { + printf("%s: %s\n", ma_log_level_to_string(logLevel), message); + } + #endif + + onLog = pContext->logCallback; + if (onLog) { + onLog(pContext, pDevice, logLevel, message); + } } - - return MA_SUCCESS; +#endif } -ma_result ma_device_start__null(ma_device* pDevice) +/* Posts an log message. Throw a breakpoint in here if you're needing to debug. The return value is always "resultCode". */ +static ma_result ma_context_post_error(ma_context* pContext, ma_device* pDevice, ma_uint32 logLevel, const char* message, ma_result resultCode) { - ma_assert(pDevice != NULL); - - ma_device_do_operation__null(pDevice, MA_DEVICE_OP_START__NULL); + /* Derive the context from the device if necessary. */ + if (pContext == NULL) { + if (pDevice != NULL) { + pContext = pDevice->pContext; + } + } - ma_atomic_exchange_32(&pDevice->null_device.isStarted, MA_TRUE); - return MA_SUCCESS; + ma_post_log_message(pContext, pDevice, logLevel, message); + return resultCode; } -ma_result ma_device_stop__null(ma_device* pDevice) +static ma_result ma_post_error(ma_device* pDevice, ma_uint32 logLevel, const char* message, ma_result resultCode) { - ma_assert(pDevice != NULL); - - ma_device_do_operation__null(pDevice, MA_DEVICE_OP_SUSPEND__NULL); - - ma_atomic_exchange_32(&pDevice->null_device.isStarted, MA_FALSE); - return MA_SUCCESS; + return ma_context_post_error(NULL, pDevice, logLevel, message, resultCode); } -ma_result ma_device_write__null(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) -{ - ma_result result = MA_SUCCESS; - ma_uint32 totalPCMFramesProcessed; - ma_bool32 wasStartedOnEntry; - - if (pFramesWritten != NULL) { - *pFramesWritten = 0; - } - - wasStartedOnEntry = pDevice->null_device.isStarted; - - /* Keep going until everything has been read. */ - totalPCMFramesProcessed = 0; - while (totalPCMFramesProcessed < frameCount) { - ma_uint64 targetFrame; - - /* If there are any frames remaining in the current period, consume those first. */ - if (pDevice->null_device.currentPeriodFramesRemainingPlayback > 0) { - ma_uint32 framesRemaining = (frameCount - totalPCMFramesProcessed); - ma_uint32 framesToProcess = pDevice->null_device.currentPeriodFramesRemainingPlayback; - if (framesToProcess > framesRemaining) { - framesToProcess = framesRemaining; - } - - /* We don't actually do anything with pPCMFrames, so just mark it as unused to prevent a warning. */ - (void)pPCMFrames; - - pDevice->null_device.currentPeriodFramesRemainingPlayback -= framesToProcess; - totalPCMFramesProcessed += framesToProcess; - } - - /* If we've consumed the current period we'll need to mark it as such an ensure the device is started if it's not already. */ - if (pDevice->null_device.currentPeriodFramesRemainingPlayback == 0) { - pDevice->null_device.currentPeriodFramesRemainingPlayback = 0; - - if (!pDevice->null_device.isStarted && !wasStartedOnEntry) { - result = ma_device_start__null(pDevice); - if (result != MA_SUCCESS) { - break; - } - } - } - /* If we've consumed the whole buffer we can return now. */ - ma_assert(totalPCMFramesProcessed <= frameCount); - if (totalPCMFramesProcessed == frameCount) { - break; - } +/******************************************************************************* - /* Getting here means we've still got more frames to consume, we but need to wait for it to become available. */ - targetFrame = pDevice->null_device.lastProcessedFramePlayback; - for (;;) { - ma_uint64 currentFrame; +Timing - /* Stop waiting if the device has been stopped. */ - if (!pDevice->null_device.isStarted) { - break; - } +*******************************************************************************/ +#ifdef MA_WIN32 +LARGE_INTEGER g_ma_TimerFrequency = {{0}}; +static void ma_timer_init(ma_timer* pTimer) +{ + LARGE_INTEGER counter; - currentFrame = ma_device_get_total_run_time_in_frames__null(pDevice); - if (currentFrame >= targetFrame) { - break; - } + if (g_ma_TimerFrequency.QuadPart == 0) { + QueryPerformanceFrequency(&g_ma_TimerFrequency); + } - /* Getting here means we haven't yet reached the target sample, so continue waiting. */ - ma_sleep(10); - } + QueryPerformanceCounter(&counter); + pTimer->counter = counter.QuadPart; +} - pDevice->null_device.lastProcessedFramePlayback += pDevice->playback.internalBufferSizeInFrames / pDevice->playback.internalPeriods; - pDevice->null_device.currentPeriodFramesRemainingPlayback = pDevice->playback.internalBufferSizeInFrames / pDevice->playback.internalPeriods; +static double ma_timer_get_time_in_seconds(ma_timer* pTimer) +{ + LARGE_INTEGER counter; + if (!QueryPerformanceCounter(&counter)) { + return 0; } - if (pFramesWritten != NULL) { - *pFramesWritten = totalPCMFramesProcessed; - } + return (double)(counter.QuadPart - pTimer->counter) / g_ma_TimerFrequency.QuadPart; +} +#elif defined(MA_APPLE) && (__MAC_OS_X_VERSION_MIN_REQUIRED < 101200) +ma_uint64 g_ma_TimerFrequency = 0; +static void ma_timer_init(ma_timer* pTimer) +{ + mach_timebase_info_data_t baseTime; + mach_timebase_info(&baseTime); + g_ma_TimerFrequency = (baseTime.denom * 1e9) / baseTime.numer; - return result; + pTimer->counter = mach_absolute_time(); } -ma_result ma_device_read__null(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead) +static double ma_timer_get_time_in_seconds(ma_timer* pTimer) { - ma_result result = MA_SUCCESS; - ma_uint32 totalPCMFramesProcessed; - - if (pFramesRead != NULL) { - *pFramesRead = 0; - } + ma_uint64 newTimeCounter = mach_absolute_time(); + ma_uint64 oldTimeCounter = pTimer->counter; - /* Keep going until everything has been read. */ - totalPCMFramesProcessed = 0; - while (totalPCMFramesProcessed < frameCount) { - ma_uint64 targetFrame; + return (newTimeCounter - oldTimeCounter) / g_ma_TimerFrequency; +} +#elif defined(MA_EMSCRIPTEN) +static MA_INLINE void ma_timer_init(ma_timer* pTimer) +{ + pTimer->counterD = emscripten_get_now(); +} - /* If there are any frames remaining in the current period, consume those first. */ - if (pDevice->null_device.currentPeriodFramesRemainingCapture > 0) { - ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 framesRemaining = (frameCount - totalPCMFramesProcessed); - ma_uint32 framesToProcess = pDevice->null_device.currentPeriodFramesRemainingCapture; - if (framesToProcess > framesRemaining) { - framesToProcess = framesRemaining; - } +static MA_INLINE double ma_timer_get_time_in_seconds(ma_timer* pTimer) +{ + return (emscripten_get_now() - pTimer->counterD) / 1000; /* Emscripten is in milliseconds. */ +} +#else +#if _POSIX_C_SOURCE >= 199309L +#if defined(CLOCK_MONOTONIC) + #define MA_CLOCK_ID CLOCK_MONOTONIC +#else + #define MA_CLOCK_ID CLOCK_REALTIME +#endif - /* We need to ensured the output buffer is zeroed. */ - ma_zero_memory(ma_offset_ptr(pPCMFrames, totalPCMFramesProcessed*bpf), framesToProcess*bpf); +static void ma_timer_init(ma_timer* pTimer) +{ + struct timespec newTime; + clock_gettime(MA_CLOCK_ID, &newTime); - pDevice->null_device.currentPeriodFramesRemainingCapture -= framesToProcess; - totalPCMFramesProcessed += framesToProcess; - } + pTimer->counter = (newTime.tv_sec * 1000000000) + newTime.tv_nsec; +} - /* If we've consumed the current period we'll need to mark it as such an ensure the device is started if it's not already. */ - if (pDevice->null_device.currentPeriodFramesRemainingCapture == 0) { - pDevice->null_device.currentPeriodFramesRemainingCapture = 0; - } +static double ma_timer_get_time_in_seconds(ma_timer* pTimer) +{ + ma_uint64 newTimeCounter; + ma_uint64 oldTimeCounter; - /* If we've consumed the whole buffer we can return now. */ - ma_assert(totalPCMFramesProcessed <= frameCount); - if (totalPCMFramesProcessed == frameCount) { - break; - } + struct timespec newTime; + clock_gettime(MA_CLOCK_ID, &newTime); - /* Getting here means we've still got more frames to consume, we but need to wait for it to become available. */ - targetFrame = pDevice->null_device.lastProcessedFrameCapture + (pDevice->capture.internalBufferSizeInFrames / pDevice->capture.internalPeriods); - for (;;) { - ma_uint64 currentFrame; + newTimeCounter = (newTime.tv_sec * 1000000000) + newTime.tv_nsec; + oldTimeCounter = pTimer->counter; - /* Stop waiting if the device has been stopped. */ - if (!pDevice->null_device.isStarted) { - break; - } + return (newTimeCounter - oldTimeCounter) / 1000000000.0; +} +#else +static void ma_timer_init(ma_timer* pTimer) +{ + struct timeval newTime; + gettimeofday(&newTime, NULL); - currentFrame = ma_device_get_total_run_time_in_frames__null(pDevice); - if (currentFrame >= targetFrame) { - break; - } + pTimer->counter = (newTime.tv_sec * 1000000) + newTime.tv_usec; +} - /* Getting here means we haven't yet reached the target sample, so continue waiting. */ - ma_sleep(10); - } +static double ma_timer_get_time_in_seconds(ma_timer* pTimer) +{ + ma_uint64 newTimeCounter; + ma_uint64 oldTimeCounter; - pDevice->null_device.lastProcessedFrameCapture += pDevice->capture.internalBufferSizeInFrames / pDevice->capture.internalPeriods; - pDevice->null_device.currentPeriodFramesRemainingCapture = pDevice->capture.internalBufferSizeInFrames / pDevice->capture.internalPeriods; - } + struct timeval newTime; + gettimeofday(&newTime, NULL); - if (pFramesRead != NULL) { - *pFramesRead = totalPCMFramesProcessed; - } + newTimeCounter = (newTime.tv_sec * 1000000) + newTime.tv_usec; + oldTimeCounter = pTimer->counter; - return result; + return (newTimeCounter - oldTimeCounter) / 1000000.0; } +#endif +#endif -ma_result ma_device_main_loop__null(ma_device* pDevice) + +/******************************************************************************* + +Dynamic Linking + +*******************************************************************************/ +ma_handle ma_dlopen(ma_context* pContext, const char* filename) { - ma_result result = MA_SUCCESS; - ma_bool32 exitLoop = MA_FALSE; - - ma_assert(pDevice != NULL); + ma_handle handle; - /* The capture device needs to be started immediately. */ - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - result = ma_device_start__null(pDevice); - if (result != MA_SUCCESS) { - return result; - } +#if MA_LOG_LEVEL >= MA_LOG_LEVEL_VERBOSE + if (pContext != NULL) { + char message[256]; + ma_strappend(message, sizeof(message), "Loading library: ", filename); + ma_post_log_message(pContext, NULL, MA_LOG_LEVEL_VERBOSE, message); } +#endif - while (ma_device__get_state(pDevice) == MA_STATE_STARTED && !exitLoop) { - switch (pDevice->type) - { - case ma_device_type_duplex: - { - /* The process is: device_read -> convert -> callback -> convert -> device_write */ - ma_uint8 capturedDeviceData[8192]; - ma_uint8 playbackDeviceData[8192]; - ma_uint32 capturedDeviceDataCapInFrames = sizeof(capturedDeviceData) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 playbackDeviceDataCapInFrames = sizeof(playbackDeviceData) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); +#ifdef _WIN32 +#ifdef MA_WIN32_DESKTOP + handle = (ma_handle)LoadLibraryA(filename); +#else + /* *sigh* It appears there is no ANSI version of LoadPackagedLibrary()... */ + WCHAR filenameW[4096]; + if (MultiByteToWideChar(CP_UTF8, 0, filename, -1, filenameW, sizeof(filenameW)) == 0) { + handle = NULL; + } else { + handle = (ma_handle)LoadPackagedLibrary(filenameW, 0); + } +#endif +#else + handle = (ma_handle)dlopen(filename, RTLD_NOW); +#endif - ma_uint32 totalFramesProcessed = 0; - ma_uint32 periodSizeInFrames = ma_min(pDevice->capture.internalBufferSizeInFrames/pDevice->capture.internalPeriods, pDevice->playback.internalBufferSizeInFrames/pDevice->playback.internalPeriods); - - while (totalFramesProcessed < periodSizeInFrames) { - ma_uint32 framesRemaining = periodSizeInFrames - totalFramesProcessed; - ma_uint32 framesProcessed; - ma_uint32 framesToProcess = framesRemaining; - if (framesToProcess > capturedDeviceDataCapInFrames) { - framesToProcess = capturedDeviceDataCapInFrames; - } + /* + I'm not considering failure to load a library an error nor a warning because seamlessly falling through to a lower-priority + backend is a deliberate design choice. Instead I'm logging it as an informational message. + */ +#if MA_LOG_LEVEL >= MA_LOG_LEVEL_INFO + if (handle == NULL) { + char message[256]; + ma_strappend(message, sizeof(message), "Failed to load library: ", filename); + ma_post_log_message(pContext, NULL, MA_LOG_LEVEL_INFO, message); + } +#endif - result = ma_device_read__null(pDevice, capturedDeviceData, framesToProcess, &framesProcessed); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } + (void)pContext; /* It's possible for pContext to be unused. */ + return handle; +} - pDevice->capture._dspFrameCount = framesToProcess; - pDevice->capture._dspFrames = capturedDeviceData; +void ma_dlclose(ma_context* pContext, ma_handle handle) +{ +#ifdef _WIN32 + FreeLibrary((HMODULE)handle); +#else + dlclose((void*)handle); +#endif - for (;;) { - ma_uint8 capturedData[8192]; - ma_uint8 playbackData[8192]; - ma_uint32 capturedDataCapInFrames = sizeof(capturedData) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); - ma_uint32 playbackDataCapInFrames = sizeof(playbackData) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); - - ma_uint32 capturedFramesToTryProcessing = ma_min(capturedDataCapInFrames, playbackDataCapInFrames); - ma_uint32 capturedFramesToProcess = (ma_uint32)ma_pcm_converter_read(&pDevice->capture.converter, capturedData, capturedFramesToTryProcessing); - if (capturedFramesToProcess == 0) { - break; /* Don't fire the data callback with zero frames. */ - } + (void)pContext; +} - ma_device__on_data(pDevice, playbackData, capturedData, capturedFramesToProcess); +ma_proc ma_dlsym(ma_context* pContext, ma_handle handle, const char* symbol) +{ + ma_proc proc; - /* At this point the playbackData buffer should be holding data that needs to be written to the device. */ - pDevice->playback._dspFrameCount = capturedFramesToProcess; - pDevice->playback._dspFrames = playbackData; - for (;;) { - ma_uint32 playbackDeviceFramesCount = (ma_uint32)ma_pcm_converter_read(&pDevice->playback.converter, playbackDeviceData, playbackDeviceDataCapInFrames); - if (playbackDeviceFramesCount == 0) { - break; - } +#if MA_LOG_LEVEL >= MA_LOG_LEVEL_VERBOSE + if (pContext != NULL) { + char message[256]; + ma_strappend(message, sizeof(message), "Loading symbol: ", symbol); + ma_post_log_message(pContext, NULL, MA_LOG_LEVEL_VERBOSE, message); + } +#endif - result = ma_device_write__null(pDevice, playbackDeviceData, playbackDeviceFramesCount, NULL); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } +#ifdef _WIN32 + proc = (ma_proc)GetProcAddress((HMODULE)handle, symbol); +#else +#if defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)) + #pragma GCC diagnostic push + #pragma GCC diagnostic ignored "-Wpedantic" +#endif + proc = (ma_proc)dlsym((void*)handle, symbol); +#if defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)) + #pragma GCC diagnostic pop +#endif +#endif - if (playbackDeviceFramesCount < playbackDeviceDataCapInFrames) { - break; - } - } +#if MA_LOG_LEVEL >= MA_LOG_LEVEL_WARNING + if (handle == NULL) { + char message[256]; + ma_strappend(message, sizeof(message), "Failed to load symbol: ", symbol); + ma_post_log_message(pContext, NULL, MA_LOG_LEVEL_WARNING, message); + } +#endif - if (capturedFramesToProcess < capturedFramesToTryProcessing) { - break; - } + (void)pContext; /* It's possible for pContext to be unused. */ + return proc; +} - /* In case an error happened from ma_device_write2__alsa()... */ - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } - } - totalFramesProcessed += framesProcessed; - } - } break; +/******************************************************************************* - case ma_device_type_capture: - { - /* We read in chunks of the period size, but use a stack allocated buffer for the intermediary. */ - ma_uint8 intermediaryBuffer[8192]; - ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 periodSizeInFrames = pDevice->capture.internalBufferSizeInFrames / pDevice->capture.internalPeriods; - ma_uint32 framesReadThisPeriod = 0; - while (framesReadThisPeriod < periodSizeInFrames) { - ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesReadThisPeriod; - ma_uint32 framesProcessed; - ma_uint32 framesToReadThisIteration = framesRemainingInPeriod; - if (framesToReadThisIteration > intermediaryBufferSizeInFrames) { - framesToReadThisIteration = intermediaryBufferSizeInFrames; - } +Threading - result = ma_device_read__null(pDevice, intermediaryBuffer, framesToReadThisIteration, &framesProcessed); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } +*******************************************************************************/ +#ifdef MA_WIN32 +static int ma_thread_priority_to_win32(ma_thread_priority priority) +{ + switch (priority) { + case ma_thread_priority_idle: return THREAD_PRIORITY_IDLE; + case ma_thread_priority_lowest: return THREAD_PRIORITY_LOWEST; + case ma_thread_priority_low: return THREAD_PRIORITY_BELOW_NORMAL; + case ma_thread_priority_normal: return THREAD_PRIORITY_NORMAL; + case ma_thread_priority_high: return THREAD_PRIORITY_ABOVE_NORMAL; + case ma_thread_priority_highest: return THREAD_PRIORITY_HIGHEST; + case ma_thread_priority_realtime: return THREAD_PRIORITY_TIME_CRITICAL; + default: return THREAD_PRIORITY_NORMAL; + } +} - ma_device__send_frames_to_client(pDevice, framesProcessed, intermediaryBuffer); +static ma_result ma_thread_create__win32(ma_context* pContext, ma_thread* pThread, ma_thread_entry_proc entryProc, void* pData) +{ + pThread->win32.hThread = CreateThread(NULL, 0, entryProc, pData, 0, NULL); + if (pThread->win32.hThread == NULL) { + return MA_FAILED_TO_CREATE_THREAD; + } - framesReadThisPeriod += framesProcessed; - } - } break; + SetThreadPriority((HANDLE)pThread->win32.hThread, ma_thread_priority_to_win32(pContext->threadPriority)); - case ma_device_type_playback: - { - /* We write in chunks of the period size, but use a stack allocated buffer for the intermediary. */ - ma_uint8 intermediaryBuffer[8192]; - ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - ma_uint32 periodSizeInFrames = pDevice->playback.internalBufferSizeInFrames / pDevice->playback.internalPeriods; - ma_uint32 framesWrittenThisPeriod = 0; - while (framesWrittenThisPeriod < periodSizeInFrames) { - ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesWrittenThisPeriod; - ma_uint32 framesProcessed; - ma_uint32 framesToWriteThisIteration = framesRemainingInPeriod; - if (framesToWriteThisIteration > intermediaryBufferSizeInFrames) { - framesToWriteThisIteration = intermediaryBufferSizeInFrames; - } + return MA_SUCCESS; +} - ma_device__read_frames_from_client(pDevice, framesToWriteThisIteration, intermediaryBuffer); +static void ma_thread_wait__win32(ma_thread* pThread) +{ + WaitForSingleObject(pThread->win32.hThread, INFINITE); +} - result = ma_device_write__null(pDevice, intermediaryBuffer, framesToWriteThisIteration, &framesProcessed); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } +static void ma_sleep__win32(ma_uint32 milliseconds) +{ + Sleep((DWORD)milliseconds); +} - framesWrittenThisPeriod += framesProcessed; - } - } break; - /* To silence a warning. Will never hit this. */ - case ma_device_type_loopback: - default: break; - } +static ma_result ma_mutex_init__win32(ma_context* pContext, ma_mutex* pMutex) +{ + (void)pContext; + + pMutex->win32.hMutex = CreateEventA(NULL, FALSE, TRUE, NULL); + if (pMutex->win32.hMutex == NULL) { + return MA_FAILED_TO_CREATE_MUTEX; } + return MA_SUCCESS; +} - /* Here is where the device is started. */ - ma_device_stop__null(pDevice); +static void ma_mutex_uninit__win32(ma_mutex* pMutex) +{ + CloseHandle(pMutex->win32.hMutex); +} - return result; +static void ma_mutex_lock__win32(ma_mutex* pMutex) +{ + WaitForSingleObject(pMutex->win32.hMutex, INFINITE); } -ma_result ma_context_uninit__null(ma_context* pContext) +static void ma_mutex_unlock__win32(ma_mutex* pMutex) { - ma_assert(pContext != NULL); - ma_assert(pContext->backend == ma_backend_null); + SetEvent(pMutex->win32.hMutex); +} + +static ma_result ma_event_init__win32(ma_context* pContext, ma_event* pEvent) +{ (void)pContext; + + pEvent->win32.hEvent = CreateEventW(NULL, FALSE, FALSE, NULL); + if (pEvent->win32.hEvent == NULL) { + return MA_FAILED_TO_CREATE_EVENT; + } + return MA_SUCCESS; } -ma_result ma_context_init__null(const ma_context_config* pConfig, ma_context* pContext) +static void ma_event_uninit__win32(ma_event* pEvent) { - ma_assert(pContext != NULL); + CloseHandle(pEvent->win32.hEvent); +} - (void)pConfig; +static ma_bool32 ma_event_wait__win32(ma_event* pEvent) +{ + return WaitForSingleObject(pEvent->win32.hEvent, INFINITE) == WAIT_OBJECT_0; +} - pContext->onUninit = ma_context_uninit__null; - pContext->onDeviceIDEqual = ma_context_is_device_id_equal__null; - pContext->onEnumDevices = ma_context_enumerate_devices__null; - pContext->onGetDeviceInfo = ma_context_get_device_info__null; - pContext->onDeviceInit = ma_device_init__null; - pContext->onDeviceUninit = ma_device_uninit__null; - pContext->onDeviceStart = NULL; /* Not required for synchronous backends. */ - pContext->onDeviceStop = NULL; /* Not required for synchronous backends. */ - pContext->onDeviceMainLoop = ma_device_main_loop__null; +static ma_bool32 ma_event_signal__win32(ma_event* pEvent) +{ + return SetEvent(pEvent->win32.hEvent); +} + + +static ma_result ma_semaphore_init__win32(ma_context* pContext, int initialValue, ma_semaphore* pSemaphore) +{ + (void)pContext; + + pSemaphore->win32.hSemaphore = CreateSemaphoreA(NULL, (LONG)initialValue, LONG_MAX, NULL); + if (pSemaphore->win32.hSemaphore == NULL) { + return MA_FAILED_TO_CREATE_SEMAPHORE; + } - /* The null backend always works. */ return MA_SUCCESS; } -#endif - -/******************************************************************************* +static void ma_semaphore_uninit__win32(ma_semaphore* pSemaphore) +{ + CloseHandle((HANDLE)pSemaphore->win32.hSemaphore); +} -WIN32 COMMON +static ma_bool32 ma_semaphore_wait__win32(ma_semaphore* pSemaphore) +{ + return WaitForSingleObject((HANDLE)pSemaphore->win32.hSemaphore, INFINITE) == WAIT_OBJECT_0; +} -*******************************************************************************/ -#if defined(MA_WIN32) -#if defined(MA_WIN32_DESKTOP) - #define ma_CoInitializeEx(pContext, pvReserved, dwCoInit) ((MA_PFN_CoInitializeEx)pContext->win32.CoInitializeEx)(pvReserved, dwCoInit) - #define ma_CoUninitialize(pContext) ((MA_PFN_CoUninitialize)pContext->win32.CoUninitialize)() - #define ma_CoCreateInstance(pContext, rclsid, pUnkOuter, dwClsContext, riid, ppv) ((MA_PFN_CoCreateInstance)pContext->win32.CoCreateInstance)(rclsid, pUnkOuter, dwClsContext, riid, ppv) - #define ma_CoTaskMemFree(pContext, pv) ((MA_PFN_CoTaskMemFree)pContext->win32.CoTaskMemFree)(pv) - #define ma_PropVariantClear(pContext, pvar) ((MA_PFN_PropVariantClear)pContext->win32.PropVariantClear)(pvar) -#else - #define ma_CoInitializeEx(pContext, pvReserved, dwCoInit) CoInitializeEx(pvReserved, dwCoInit) - #define ma_CoUninitialize(pContext) CoUninitialize() - #define ma_CoCreateInstance(pContext, rclsid, pUnkOuter, dwClsContext, riid, ppv) CoCreateInstance(rclsid, pUnkOuter, dwClsContext, riid, ppv) - #define ma_CoTaskMemFree(pContext, pv) CoTaskMemFree(pv) - #define ma_PropVariantClear(pContext, pvar) PropVariantClear(pvar) +static ma_bool32 ma_semaphore_release__win32(ma_semaphore* pSemaphore) +{ + return ReleaseSemaphore((HANDLE)pSemaphore->win32.hSemaphore, 1, NULL) != 0; +} #endif -#if !defined(MAXULONG_PTR) -typedef size_t DWORD_PTR; -#endif -#if !defined(WAVE_FORMAT_44M08) -#define WAVE_FORMAT_44M08 0x00000100 -#define WAVE_FORMAT_44S08 0x00000200 -#define WAVE_FORMAT_44M16 0x00000400 -#define WAVE_FORMAT_44S16 0x00000800 -#define WAVE_FORMAT_48M08 0x00001000 -#define WAVE_FORMAT_48S08 0x00002000 -#define WAVE_FORMAT_48M16 0x00004000 -#define WAVE_FORMAT_48S16 0x00008000 -#define WAVE_FORMAT_96M08 0x00010000 -#define WAVE_FORMAT_96S08 0x00020000 -#define WAVE_FORMAT_96M16 0x00040000 -#define WAVE_FORMAT_96S16 0x00080000 -#endif +#ifdef MA_POSIX +#include -#ifndef SPEAKER_FRONT_LEFT -#define SPEAKER_FRONT_LEFT 0x1 -#define SPEAKER_FRONT_RIGHT 0x2 -#define SPEAKER_FRONT_CENTER 0x4 -#define SPEAKER_LOW_FREQUENCY 0x8 -#define SPEAKER_BACK_LEFT 0x10 -#define SPEAKER_BACK_RIGHT 0x20 -#define SPEAKER_FRONT_LEFT_OF_CENTER 0x40 -#define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80 -#define SPEAKER_BACK_CENTER 0x100 -#define SPEAKER_SIDE_LEFT 0x200 -#define SPEAKER_SIDE_RIGHT 0x400 -#define SPEAKER_TOP_CENTER 0x800 -#define SPEAKER_TOP_FRONT_LEFT 0x1000 -#define SPEAKER_TOP_FRONT_CENTER 0x2000 -#define SPEAKER_TOP_FRONT_RIGHT 0x4000 -#define SPEAKER_TOP_BACK_LEFT 0x8000 -#define SPEAKER_TOP_BACK_CENTER 0x10000 -#define SPEAKER_TOP_BACK_RIGHT 0x20000 -#endif +typedef int (* ma_pthread_create_proc)(pthread_t *thread, const pthread_attr_t *attr, void *(*start_routine) (void *), void *arg); +typedef int (* ma_pthread_join_proc)(pthread_t thread, void **retval); +typedef int (* ma_pthread_mutex_init_proc)(pthread_mutex_t *__mutex, const pthread_mutexattr_t *__mutexattr); +typedef int (* ma_pthread_mutex_destroy_proc)(pthread_mutex_t *__mutex); +typedef int (* ma_pthread_mutex_lock_proc)(pthread_mutex_t *__mutex); +typedef int (* ma_pthread_mutex_unlock_proc)(pthread_mutex_t *__mutex); +typedef int (* ma_pthread_cond_init_proc)(pthread_cond_t *__restrict __cond, const pthread_condattr_t *__restrict __cond_attr); +typedef int (* ma_pthread_cond_destroy_proc)(pthread_cond_t *__cond); +typedef int (* ma_pthread_cond_signal_proc)(pthread_cond_t *__cond); +typedef int (* ma_pthread_cond_wait_proc)(pthread_cond_t *__restrict __cond, pthread_mutex_t *__restrict __mutex); +typedef int (* ma_pthread_attr_init_proc)(pthread_attr_t *attr); +typedef int (* ma_pthread_attr_destroy_proc)(pthread_attr_t *attr); +typedef int (* ma_pthread_attr_setschedpolicy_proc)(pthread_attr_t *attr, int policy); +typedef int (* ma_pthread_attr_getschedparam_proc)(const pthread_attr_t *attr, struct sched_param *param); +typedef int (* ma_pthread_attr_setschedparam_proc)(pthread_attr_t *attr, const struct sched_param *param); -/* -The SDK that comes with old versions of MSVC (VC6, for example) does not appear to define WAVEFORMATEXTENSIBLE. We -define our own implementation in this case. -*/ -#if (defined(_MSC_VER) && !defined(_WAVEFORMATEXTENSIBLE_)) || defined(__DMC__) -typedef struct +static ma_result ma_thread_create__posix(ma_context* pContext, ma_thread* pThread, ma_thread_entry_proc entryProc, void* pData) { - WAVEFORMATEX Format; - union - { - WORD wValidBitsPerSample; - WORD wSamplesPerBlock; - WORD wReserved; - } Samples; - DWORD dwChannelMask; - GUID SubFormat; -} WAVEFORMATEXTENSIBLE; -#endif + int result; + pthread_attr_t* pAttr = NULL; -#ifndef WAVE_FORMAT_EXTENSIBLE -#define WAVE_FORMAT_EXTENSIBLE 0xFFFE +#if !defined(__EMSCRIPTEN__) + /* Try setting the thread priority. It's not critical if anything fails here. */ + pthread_attr_t attr; + if (((ma_pthread_attr_init_proc)pContext->posix.pthread_attr_init)(&attr) == 0) { + int scheduler = -1; + if (pContext->threadPriority == ma_thread_priority_idle) { +#ifdef SCHED_IDLE + if (((ma_pthread_attr_setschedpolicy_proc)pContext->posix.pthread_attr_setschedpolicy)(&attr, SCHED_IDLE) == 0) { + scheduler = SCHED_IDLE; + } #endif - -#ifndef WAVE_FORMAT_IEEE_FLOAT -#define WAVE_FORMAT_IEEE_FLOAT 0x0003 + } else if (pContext->threadPriority == ma_thread_priority_realtime) { +#ifdef SCHED_FIFO + if (((ma_pthread_attr_setschedpolicy_proc)pContext->posix.pthread_attr_setschedpolicy)(&attr, SCHED_FIFO) == 0) { + scheduler = SCHED_FIFO; + } +#endif +#ifdef MA_LINUX + } else { + scheduler = sched_getscheduler(0); #endif + } -GUID MA_GUID_NULL = {0x00000000, 0x0000, 0x0000, {0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00}}; + if (scheduler != -1) { + int priorityMin = sched_get_priority_min(scheduler); + int priorityMax = sched_get_priority_max(scheduler); + int priorityStep = (priorityMax - priorityMin) / 7; /* 7 = number of priorities supported by miniaudio. */ -/* Converts an individual Win32-style channel identifier (SPEAKER_FRONT_LEFT, etc.) to miniaudio. */ -ma_uint8 ma_channel_id_to_ma__win32(DWORD id) -{ - switch (id) - { - case SPEAKER_FRONT_LEFT: return MA_CHANNEL_FRONT_LEFT; - case SPEAKER_FRONT_RIGHT: return MA_CHANNEL_FRONT_RIGHT; - case SPEAKER_FRONT_CENTER: return MA_CHANNEL_FRONT_CENTER; - case SPEAKER_LOW_FREQUENCY: return MA_CHANNEL_LFE; - case SPEAKER_BACK_LEFT: return MA_CHANNEL_BACK_LEFT; - case SPEAKER_BACK_RIGHT: return MA_CHANNEL_BACK_RIGHT; - case SPEAKER_FRONT_LEFT_OF_CENTER: return MA_CHANNEL_FRONT_LEFT_CENTER; - case SPEAKER_FRONT_RIGHT_OF_CENTER: return MA_CHANNEL_FRONT_RIGHT_CENTER; - case SPEAKER_BACK_CENTER: return MA_CHANNEL_BACK_CENTER; - case SPEAKER_SIDE_LEFT: return MA_CHANNEL_SIDE_LEFT; - case SPEAKER_SIDE_RIGHT: return MA_CHANNEL_SIDE_RIGHT; - case SPEAKER_TOP_CENTER: return MA_CHANNEL_TOP_CENTER; - case SPEAKER_TOP_FRONT_LEFT: return MA_CHANNEL_TOP_FRONT_LEFT; - case SPEAKER_TOP_FRONT_CENTER: return MA_CHANNEL_TOP_FRONT_CENTER; - case SPEAKER_TOP_FRONT_RIGHT: return MA_CHANNEL_TOP_FRONT_RIGHT; - case SPEAKER_TOP_BACK_LEFT: return MA_CHANNEL_TOP_BACK_LEFT; - case SPEAKER_TOP_BACK_CENTER: return MA_CHANNEL_TOP_BACK_CENTER; - case SPEAKER_TOP_BACK_RIGHT: return MA_CHANNEL_TOP_BACK_RIGHT; - default: return 0; - } -} + struct sched_param sched; + if (((ma_pthread_attr_getschedparam_proc)pContext->posix.pthread_attr_getschedparam)(&attr, &sched) == 0) { + if (pContext->threadPriority == ma_thread_priority_idle) { + sched.sched_priority = priorityMin; + } else if (pContext->threadPriority == ma_thread_priority_realtime) { + sched.sched_priority = priorityMax; + } else { + sched.sched_priority += ((int)pContext->threadPriority + 5) * priorityStep; /* +5 because the lowest priority is -5. */ + if (sched.sched_priority < priorityMin) { + sched.sched_priority = priorityMin; + } + if (sched.sched_priority > priorityMax) { + sched.sched_priority = priorityMax; + } + } -/* Converts an individual miniaudio channel identifier (MA_CHANNEL_FRONT_LEFT, etc.) to Win32-style. */ -DWORD ma_channel_id_to_win32(DWORD id) -{ - switch (id) - { - case MA_CHANNEL_MONO: return SPEAKER_FRONT_CENTER; - case MA_CHANNEL_FRONT_LEFT: return SPEAKER_FRONT_LEFT; - case MA_CHANNEL_FRONT_RIGHT: return SPEAKER_FRONT_RIGHT; - case MA_CHANNEL_FRONT_CENTER: return SPEAKER_FRONT_CENTER; - case MA_CHANNEL_LFE: return SPEAKER_LOW_FREQUENCY; - case MA_CHANNEL_BACK_LEFT: return SPEAKER_BACK_LEFT; - case MA_CHANNEL_BACK_RIGHT: return SPEAKER_BACK_RIGHT; - case MA_CHANNEL_FRONT_LEFT_CENTER: return SPEAKER_FRONT_LEFT_OF_CENTER; - case MA_CHANNEL_FRONT_RIGHT_CENTER: return SPEAKER_FRONT_RIGHT_OF_CENTER; - case MA_CHANNEL_BACK_CENTER: return SPEAKER_BACK_CENTER; - case MA_CHANNEL_SIDE_LEFT: return SPEAKER_SIDE_LEFT; - case MA_CHANNEL_SIDE_RIGHT: return SPEAKER_SIDE_RIGHT; - case MA_CHANNEL_TOP_CENTER: return SPEAKER_TOP_CENTER; - case MA_CHANNEL_TOP_FRONT_LEFT: return SPEAKER_TOP_FRONT_LEFT; - case MA_CHANNEL_TOP_FRONT_CENTER: return SPEAKER_TOP_FRONT_CENTER; - case MA_CHANNEL_TOP_FRONT_RIGHT: return SPEAKER_TOP_FRONT_RIGHT; - case MA_CHANNEL_TOP_BACK_LEFT: return SPEAKER_TOP_BACK_LEFT; - case MA_CHANNEL_TOP_BACK_CENTER: return SPEAKER_TOP_BACK_CENTER; - case MA_CHANNEL_TOP_BACK_RIGHT: return SPEAKER_TOP_BACK_RIGHT; - default: return 0; + if (((ma_pthread_attr_setschedparam_proc)pContext->posix.pthread_attr_setschedparam)(&attr, &sched) == 0) { + pAttr = &attr; + } + } + } + + ((ma_pthread_attr_destroy_proc)pContext->posix.pthread_attr_destroy)(&attr); + } +#endif + + result = ((ma_pthread_create_proc)pContext->posix.pthread_create)(&pThread->posix.thread, pAttr, entryProc, pData); + if (result != 0) { + return MA_FAILED_TO_CREATE_THREAD; } + + return MA_SUCCESS; } -/* Converts a channel mapping to a Win32-style channel mask. */ -DWORD ma_channel_map_to_channel_mask__win32(const ma_channel channelMap[MA_MAX_CHANNELS], ma_uint32 channels) +static void ma_thread_wait__posix(ma_thread* pThread) { - DWORD dwChannelMask = 0; - ma_uint32 iChannel; + ((ma_pthread_join_proc)pThread->pContext->posix.pthread_join)(pThread->posix.thread, NULL); +} - for (iChannel = 0; iChannel < channels; ++iChannel) { - dwChannelMask |= ma_channel_id_to_win32(channelMap[iChannel]); +#if !defined(MA_EMSCRIPTEN) +static void ma_sleep__posix(ma_uint32 milliseconds) +{ +#ifdef MA_EMSCRIPTEN + (void)milliseconds; + MA_ASSERT(MA_FALSE); /* The Emscripten build should never sleep. */ +#else + #if _POSIX_C_SOURCE >= 199309L + struct timespec ts; + ts.tv_sec = milliseconds / 1000000; + ts.tv_nsec = milliseconds % 1000000 * 1000000; + nanosleep(&ts, NULL); + #else + struct timeval tv; + tv.tv_sec = milliseconds / 1000; + tv.tv_usec = milliseconds % 1000 * 1000; + select(0, NULL, NULL, NULL, &tv); + #endif +#endif +} +#endif /* MA_EMSCRIPTEN */ + + +static ma_result ma_mutex_init__posix(ma_context* pContext, ma_mutex* pMutex) +{ + int result = ((ma_pthread_mutex_init_proc)pContext->posix.pthread_mutex_init)(&pMutex->posix.mutex, NULL); + if (result != 0) { + return MA_FAILED_TO_CREATE_MUTEX; } - return dwChannelMask; + return MA_SUCCESS; } -/* Converts a Win32-style channel mask to a miniaudio channel map. */ -void ma_channel_mask_to_channel_map__win32(DWORD dwChannelMask, ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) +static void ma_mutex_uninit__posix(ma_mutex* pMutex) { - if (channels == 1 && dwChannelMask == 0) { - channelMap[0] = MA_CHANNEL_MONO; - } else if (channels == 2 && dwChannelMask == 0) { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - } else { - if (channels == 1 && (dwChannelMask & SPEAKER_FRONT_CENTER) != 0) { - channelMap[0] = MA_CHANNEL_MONO; - } else { - /* Just iterate over each bit. */ - ma_uint32 iChannel = 0; - ma_uint32 iBit; + ((ma_pthread_mutex_destroy_proc)pMutex->pContext->posix.pthread_mutex_destroy)(&pMutex->posix.mutex); +} - for (iBit = 0; iBit < 32; ++iBit) { - DWORD bitValue = (dwChannelMask & (1UL << iBit)); - if (bitValue != 0) { - /* The bit is set. */ - channelMap[iChannel] = ma_channel_id_to_ma__win32(bitValue); - iChannel += 1; - } - } - } - } +static void ma_mutex_lock__posix(ma_mutex* pMutex) +{ + ((ma_pthread_mutex_lock_proc)pMutex->pContext->posix.pthread_mutex_lock)(&pMutex->posix.mutex); } -#ifdef __cplusplus -ma_bool32 ma_is_guid_equal(const void* a, const void* b) +static void ma_mutex_unlock__posix(ma_mutex* pMutex) { - return IsEqualGUID(*(const GUID*)a, *(const GUID*)b); + ((ma_pthread_mutex_unlock_proc)pMutex->pContext->posix.pthread_mutex_unlock)(&pMutex->posix.mutex); } -#else -#define ma_is_guid_equal(a, b) IsEqualGUID((const GUID*)a, (const GUID*)b) -#endif -ma_format ma_format_from_WAVEFORMATEX(const WAVEFORMATEX* pWF) + +static ma_result ma_event_init__posix(ma_context* pContext, ma_event* pEvent) { - ma_assert(pWF != NULL); + if (((ma_pthread_mutex_init_proc)pContext->posix.pthread_mutex_init)(&pEvent->posix.mutex, NULL) != 0) { + return MA_FAILED_TO_CREATE_MUTEX; + } - if (pWF->wFormatTag == WAVE_FORMAT_EXTENSIBLE) { - const WAVEFORMATEXTENSIBLE* pWFEX = (const WAVEFORMATEXTENSIBLE*)pWF; - if (ma_is_guid_equal(&pWFEX->SubFormat, &MA_GUID_KSDATAFORMAT_SUBTYPE_PCM)) { - if (pWFEX->Samples.wValidBitsPerSample == 32) { - return ma_format_s32; - } - if (pWFEX->Samples.wValidBitsPerSample == 24) { - if (pWFEX->Format.wBitsPerSample == 32) { - /*return ma_format_s24_32;*/ - } - if (pWFEX->Format.wBitsPerSample == 24) { - return ma_format_s24; - } - } - if (pWFEX->Samples.wValidBitsPerSample == 16) { - return ma_format_s16; - } - if (pWFEX->Samples.wValidBitsPerSample == 8) { - return ma_format_u8; - } - } - if (ma_is_guid_equal(&pWFEX->SubFormat, &MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)) { - if (pWFEX->Samples.wValidBitsPerSample == 32) { - return ma_format_f32; - } - /* - if (pWFEX->Samples.wValidBitsPerSample == 64) { - return ma_format_f64; - } - */ - } - } else { - if (pWF->wFormatTag == WAVE_FORMAT_PCM) { - if (pWF->wBitsPerSample == 32) { - return ma_format_s32; - } - if (pWF->wBitsPerSample == 24) { - return ma_format_s24; - } - if (pWF->wBitsPerSample == 16) { - return ma_format_s16; - } - if (pWF->wBitsPerSample == 8) { - return ma_format_u8; - } - } - if (pWF->wFormatTag == WAVE_FORMAT_IEEE_FLOAT) { - if (pWF->wBitsPerSample == 32) { - return ma_format_f32; - } - if (pWF->wBitsPerSample == 64) { - /*return ma_format_f64;*/ - } - } + if (((ma_pthread_cond_init_proc)pContext->posix.pthread_cond_init)(&pEvent->posix.condition, NULL) != 0) { + return MA_FAILED_TO_CREATE_EVENT; } - return ma_format_unknown; + pEvent->posix.value = 0; + return MA_SUCCESS; } -#endif - -/******************************************************************************* - -WASAPI Backend +static void ma_event_uninit__posix(ma_event* pEvent) +{ + ((ma_pthread_cond_destroy_proc)pEvent->pContext->posix.pthread_cond_destroy)(&pEvent->posix.condition); + ((ma_pthread_mutex_destroy_proc)pEvent->pContext->posix.pthread_mutex_destroy)(&pEvent->posix.mutex); +} -*******************************************************************************/ -#ifdef MA_HAS_WASAPI -#if 0 -#if defined(_MSC_VER) - #pragma warning(push) - #pragma warning(disable:4091) /* 'typedef ': ignored on left of '' when no variable is declared */ -#endif -#include -#include -#if defined(_MSC_VER) - #pragma warning(pop) -#endif -#endif /* 0 */ +static ma_bool32 ma_event_wait__posix(ma_event* pEvent) +{ + ((ma_pthread_mutex_lock_proc)pEvent->pContext->posix.pthread_mutex_lock)(&pEvent->posix.mutex); + { + while (pEvent->posix.value == 0) { + ((ma_pthread_cond_wait_proc)pEvent->pContext->posix.pthread_cond_wait)(&pEvent->posix.condition, &pEvent->posix.mutex); + } + pEvent->posix.value = 0; /* Auto-reset. */ + } + ((ma_pthread_mutex_unlock_proc)pEvent->pContext->posix.pthread_mutex_unlock)(&pEvent->posix.mutex); + return MA_TRUE; +} +static ma_bool32 ma_event_signal__posix(ma_event* pEvent) +{ + ((ma_pthread_mutex_lock_proc)pEvent->pContext->posix.pthread_mutex_lock)(&pEvent->posix.mutex); + { + pEvent->posix.value = 1; + ((ma_pthread_cond_signal_proc)pEvent->pContext->posix.pthread_cond_signal)(&pEvent->posix.condition); + } + ((ma_pthread_mutex_unlock_proc)pEvent->pContext->posix.pthread_mutex_unlock)(&pEvent->posix.mutex); + return MA_TRUE; +} -/* Some compilers don't define VerifyVersionInfoW. Need to write this ourselves. */ -#define MA_WIN32_WINNT_VISTA 0x0600 -#define MA_VER_MINORVERSION 0x01 -#define MA_VER_MAJORVERSION 0x02 -#define MA_VER_SERVICEPACKMAJOR 0x20 -#define MA_VER_GREATER_EQUAL 0x03 -typedef struct { - DWORD dwOSVersionInfoSize; - DWORD dwMajorVersion; - DWORD dwMinorVersion; - DWORD dwBuildNumber; - DWORD dwPlatformId; - WCHAR szCSDVersion[128]; - WORD wServicePackMajor; - WORD wServicePackMinor; - WORD wSuiteMask; - BYTE wProductType; - BYTE wReserved; -} ma_OSVERSIONINFOEXW; +static ma_result ma_semaphore_init__posix(ma_context* pContext, int initialValue, ma_semaphore* pSemaphore) +{ + (void)pContext; -typedef BOOL (WINAPI * ma_PFNVerifyVersionInfoW) (ma_OSVERSIONINFOEXW* lpVersionInfo, DWORD dwTypeMask, DWORDLONG dwlConditionMask); -typedef ULONGLONG (WINAPI * ma_PFNVerSetConditionMask)(ULONGLONG dwlConditionMask, DWORD dwTypeBitMask, BYTE dwConditionMask); +#if defined(MA_APPLE) + /* Not yet implemented for Apple platforms since sem_init() is deprecated. Need to use a named semaphore via sem_open() instead. */ + return MA_INVALID_OPERATION; +#else + if (sem_init(&pSemaphore->posix.semaphore, 0, (unsigned int)initialValue) == 0) { + return MA_FAILED_TO_CREATE_SEMAPHORE; + } +#endif + return MA_SUCCESS; +} -#ifndef PROPERTYKEY_DEFINED -#define PROPERTYKEY_DEFINED -typedef struct +static void ma_semaphore_uninit__posix(ma_semaphore* pSemaphore) { - GUID fmtid; - DWORD pid; -} PROPERTYKEY; -#endif + sem_close(&pSemaphore->posix.semaphore); +} -/* Some compilers don't define PropVariantInit(). We just do this ourselves since it's just a memset(). */ -static MA_INLINE void ma_PropVariantInit(PROPVARIANT* pProp) +static ma_bool32 ma_semaphore_wait__posix(ma_semaphore* pSemaphore) { - ma_zero_object(pProp); + return sem_wait(&pSemaphore->posix.semaphore) != -1; } +static ma_bool32 ma_semaphore_release__posix(ma_semaphore* pSemaphore) +{ + return sem_post(&pSemaphore->posix.semaphore) != -1; +} +#endif -const PROPERTYKEY MA_PKEY_Device_FriendlyName = {{0xA45C254E, 0xDF1C, 0x4EFD, {0x80, 0x20, 0x67, 0xD1, 0x46, 0xA8, 0x50, 0xE0}}, 14}; -const PROPERTYKEY MA_PKEY_AudioEngine_DeviceFormat = {{0xF19F064D, 0x82C, 0x4E27, {0xBC, 0x73, 0x68, 0x82, 0xA1, 0xBB, 0x8E, 0x4C}}, 0}; +static ma_result ma_thread_create(ma_context* pContext, ma_thread* pThread, ma_thread_entry_proc entryProc, void* pData) +{ + if (pContext == NULL || pThread == NULL || entryProc == NULL) { + return MA_FALSE; + } -const IID MA_IID_IUnknown = {0x00000000, 0x0000, 0x0000, {0xC0, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x46}}; /* 00000000-0000-0000-C000-000000000046 */ -const IID MA_IID_IAgileObject = {0x94EA2B94, 0xE9CC, 0x49E0, {0xC0, 0xFF, 0xEE, 0x64, 0xCA, 0x8F, 0x5B, 0x90}}; /* 94EA2B94-E9CC-49E0-C0FF-EE64CA8F5B90 */ + pThread->pContext = pContext; -const IID MA_IID_IAudioClient = {0x1CB9AD4C, 0xDBFA, 0x4C32, {0xB1, 0x78, 0xC2, 0xF5, 0x68, 0xA7, 0x03, 0xB2}}; /* 1CB9AD4C-DBFA-4C32-B178-C2F568A703B2 = __uuidof(IAudioClient) */ -const IID MA_IID_IAudioClient2 = {0x726778CD, 0xF60A, 0x4EDA, {0x82, 0xDE, 0xE4, 0x76, 0x10, 0xCD, 0x78, 0xAA}}; /* 726778CD-F60A-4EDA-82DE-E47610CD78AA = __uuidof(IAudioClient2) */ -const IID MA_IID_IAudioClient3 = {0x7ED4EE07, 0x8E67, 0x4CD4, {0x8C, 0x1A, 0x2B, 0x7A, 0x59, 0x87, 0xAD, 0x42}}; /* 7ED4EE07-8E67-4CD4-8C1A-2B7A5987AD42 = __uuidof(IAudioClient3) */ -const IID MA_IID_IAudioRenderClient = {0xF294ACFC, 0x3146, 0x4483, {0xA7, 0xBF, 0xAD, 0xDC, 0xA7, 0xC2, 0x60, 0xE2}}; /* F294ACFC-3146-4483-A7BF-ADDCA7C260E2 = __uuidof(IAudioRenderClient) */ -const IID MA_IID_IAudioCaptureClient = {0xC8ADBD64, 0xE71E, 0x48A0, {0xA4, 0xDE, 0x18, 0x5C, 0x39, 0x5C, 0xD3, 0x17}}; /* C8ADBD64-E71E-48A0-A4DE-185C395CD317 = __uuidof(IAudioCaptureClient) */ -const IID MA_IID_IMMNotificationClient = {0x7991EEC9, 0x7E89, 0x4D85, {0x83, 0x90, 0x6C, 0x70, 0x3C, 0xEC, 0x60, 0xC0}}; /* 7991EEC9-7E89-4D85-8390-6C703CEC60C0 = __uuidof(IMMNotificationClient) */ -#ifndef MA_WIN32_DESKTOP -const IID MA_IID_DEVINTERFACE_AUDIO_RENDER = {0xE6327CAD, 0xDCEC, 0x4949, {0xAE, 0x8A, 0x99, 0x1E, 0x97, 0x6A, 0x79, 0xD2}}; /* E6327CAD-DCEC-4949-AE8A-991E976A79D2 */ -const IID MA_IID_DEVINTERFACE_AUDIO_CAPTURE = {0x2EEF81BE, 0x33FA, 0x4800, {0x96, 0x70, 0x1C, 0xD4, 0x74, 0x97, 0x2C, 0x3F}}; /* 2EEF81BE-33FA-4800-9670-1CD474972C3F */ -const IID MA_IID_IActivateAudioInterfaceCompletionHandler = {0x41D949AB, 0x9862, 0x444A, {0x80, 0xF6, 0xC2, 0x61, 0x33, 0x4D, 0xA5, 0xEB}}; /* 41D949AB-9862-444A-80F6-C261334DA5EB */ +#ifdef MA_WIN32 + return ma_thread_create__win32(pContext, pThread, entryProc, pData); #endif - -const IID MA_CLSID_MMDeviceEnumerator_Instance = {0xBCDE0395, 0xE52F, 0x467C, {0x8E, 0x3D, 0xC4, 0x57, 0x92, 0x91, 0x69, 0x2E}}; /* BCDE0395-E52F-467C-8E3D-C4579291692E = __uuidof(MMDeviceEnumerator) */ -const IID MA_IID_IMMDeviceEnumerator_Instance = {0xA95664D2, 0x9614, 0x4F35, {0xA7, 0x46, 0xDE, 0x8D, 0xB6, 0x36, 0x17, 0xE6}}; /* A95664D2-9614-4F35-A746-DE8DB63617E6 = __uuidof(IMMDeviceEnumerator) */ -#ifdef __cplusplus -#define MA_CLSID_MMDeviceEnumerator MA_CLSID_MMDeviceEnumerator_Instance -#define MA_IID_IMMDeviceEnumerator MA_IID_IMMDeviceEnumerator_Instance -#else -#define MA_CLSID_MMDeviceEnumerator &MA_CLSID_MMDeviceEnumerator_Instance -#define MA_IID_IMMDeviceEnumerator &MA_IID_IMMDeviceEnumerator_Instance +#ifdef MA_POSIX + return ma_thread_create__posix(pContext, pThread, entryProc, pData); #endif +} -typedef struct ma_IUnknown ma_IUnknown; -#ifdef MA_WIN32_DESKTOP -#define MA_MM_DEVICE_STATE_ACTIVE 1 -#define MA_MM_DEVICE_STATE_DISABLED 2 -#define MA_MM_DEVICE_STATE_NOTPRESENT 4 -#define MA_MM_DEVICE_STATE_UNPLUGGED 8 +static void ma_thread_wait(ma_thread* pThread) +{ + if (pThread == NULL) { + return; + } + +#ifdef MA_WIN32 + ma_thread_wait__win32(pThread); +#endif +#ifdef MA_POSIX + ma_thread_wait__posix(pThread); +#endif +} -typedef struct ma_IMMDeviceEnumerator ma_IMMDeviceEnumerator; -typedef struct ma_IMMDeviceCollection ma_IMMDeviceCollection; -typedef struct ma_IMMDevice ma_IMMDevice; -#else -typedef struct ma_IActivateAudioInterfaceCompletionHandler ma_IActivateAudioInterfaceCompletionHandler; -typedef struct ma_IActivateAudioInterfaceAsyncOperation ma_IActivateAudioInterfaceAsyncOperation; +#if !defined(MA_EMSCRIPTEN) +static void ma_sleep(ma_uint32 milliseconds) +{ +#ifdef MA_WIN32 + ma_sleep__win32(milliseconds); +#endif +#ifdef MA_POSIX + ma_sleep__posix(milliseconds); +#endif +} #endif -typedef struct ma_IPropertyStore ma_IPropertyStore; -typedef struct ma_IAudioClient ma_IAudioClient; -typedef struct ma_IAudioClient2 ma_IAudioClient2; -typedef struct ma_IAudioClient3 ma_IAudioClient3; -typedef struct ma_IAudioRenderClient ma_IAudioRenderClient; -typedef struct ma_IAudioCaptureClient ma_IAudioCaptureClient; -typedef ma_int64 MA_REFERENCE_TIME; -#define MA_AUDCLNT_STREAMFLAGS_CROSSPROCESS 0x00010000 -#define MA_AUDCLNT_STREAMFLAGS_LOOPBACK 0x00020000 -#define MA_AUDCLNT_STREAMFLAGS_EVENTCALLBACK 0x00040000 -#define MA_AUDCLNT_STREAMFLAGS_NOPERSIST 0x00080000 -#define MA_AUDCLNT_STREAMFLAGS_RATEADJUST 0x00100000 -#define MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY 0x08000000 -#define MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM 0x80000000 -#define MA_AUDCLNT_SESSIONFLAGS_EXPIREWHENUNOWNED 0x10000000 -#define MA_AUDCLNT_SESSIONFLAGS_DISPLAY_HIDE 0x20000000 -#define MA_AUDCLNT_SESSIONFLAGS_DISPLAY_HIDEWHENEXPIRED 0x40000000 +ma_result ma_mutex_init(ma_context* pContext, ma_mutex* pMutex) +{ + if (pContext == NULL || pMutex == NULL) { + return MA_INVALID_ARGS; + } -/* We only care about a few error codes. */ -#define MA_AUDCLNT_E_INVALID_DEVICE_PERIOD (-2004287456) -#define MA_AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED (-2004287463) -#define MA_AUDCLNT_S_BUFFER_EMPTY (143196161) -#define MA_AUDCLNT_E_DEVICE_IN_USE (-2004287478) + pMutex->pContext = pContext; -/* Buffer flags. */ -#define MA_AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY 1 -#define MA_AUDCLNT_BUFFERFLAGS_SILENT 2 -#define MA_AUDCLNT_BUFFERFLAGS_TIMESTAMP_ERROR 4 +#ifdef MA_WIN32 + return ma_mutex_init__win32(pContext, pMutex); +#endif +#ifdef MA_POSIX + return ma_mutex_init__posix(pContext, pMutex); +#endif +} -typedef enum +void ma_mutex_uninit(ma_mutex* pMutex) { - ma_eRender = 0, - ma_eCapture = 1, - ma_eAll = 2 -} ma_EDataFlow; + if (pMutex == NULL || pMutex->pContext == NULL) { + return; + } -typedef enum -{ - ma_eConsole = 0, - ma_eMultimedia = 1, - ma_eCommunications = 2 -} ma_ERole; +#ifdef MA_WIN32 + ma_mutex_uninit__win32(pMutex); +#endif +#ifdef MA_POSIX + ma_mutex_uninit__posix(pMutex); +#endif +} -typedef enum +void ma_mutex_lock(ma_mutex* pMutex) { - MA_AUDCLNT_SHAREMODE_SHARED, - MA_AUDCLNT_SHAREMODE_EXCLUSIVE -} MA_AUDCLNT_SHAREMODE; + if (pMutex == NULL || pMutex->pContext == NULL) { + return; + } -typedef enum -{ - MA_AudioCategory_Other = 0 /* <-- miniaudio is only caring about Other. */ -} MA_AUDIO_STREAM_CATEGORY; +#ifdef MA_WIN32 + ma_mutex_lock__win32(pMutex); +#endif +#ifdef MA_POSIX + ma_mutex_lock__posix(pMutex); +#endif +} -typedef struct +void ma_mutex_unlock(ma_mutex* pMutex) { - UINT32 cbSize; - BOOL bIsOffload; - MA_AUDIO_STREAM_CATEGORY eCategory; -} ma_AudioClientProperties; + if (pMutex == NULL || pMutex->pContext == NULL) { + return; +} -/* IUnknown */ -typedef struct -{ - /* IUnknown */ - HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IUnknown* pThis, const IID* const riid, void** ppObject); - ULONG (STDMETHODCALLTYPE * AddRef) (ma_IUnknown* pThis); - ULONG (STDMETHODCALLTYPE * Release) (ma_IUnknown* pThis); -} ma_IUnknownVtbl; -struct ma_IUnknown -{ - ma_IUnknownVtbl* lpVtbl; -}; -HRESULT ma_IUnknown_QueryInterface(ma_IUnknown* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } -ULONG ma_IUnknown_AddRef(ma_IUnknown* pThis) { return pThis->lpVtbl->AddRef(pThis); } -ULONG ma_IUnknown_Release(ma_IUnknown* pThis) { return pThis->lpVtbl->Release(pThis); } +#ifdef MA_WIN32 + ma_mutex_unlock__win32(pMutex); +#endif +#ifdef MA_POSIX + ma_mutex_unlock__posix(pMutex); +#endif +} -#ifdef MA_WIN32_DESKTOP - /* IMMNotificationClient */ - typedef struct - { - /* IUnknown */ - HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMNotificationClient* pThis, const IID* const riid, void** ppObject); - ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMNotificationClient* pThis); - ULONG (STDMETHODCALLTYPE * Release) (ma_IMMNotificationClient* pThis); - /* IMMNotificationClient */ - HRESULT (STDMETHODCALLTYPE * OnDeviceStateChanged) (ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID, DWORD dwNewState); - HRESULT (STDMETHODCALLTYPE * OnDeviceAdded) (ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID); - HRESULT (STDMETHODCALLTYPE * OnDeviceRemoved) (ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID); - HRESULT (STDMETHODCALLTYPE * OnDefaultDeviceChanged)(ma_IMMNotificationClient* pThis, ma_EDataFlow dataFlow, ma_ERole role, LPCWSTR pDefaultDeviceID); - HRESULT (STDMETHODCALLTYPE * OnPropertyValueChanged)(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID, const PROPERTYKEY key); - } ma_IMMNotificationClientVtbl; +ma_result ma_event_init(ma_context* pContext, ma_event* pEvent) +{ + if (pContext == NULL || pEvent == NULL) { + return MA_FALSE; + } - /* IMMDeviceEnumerator */ - typedef struct - { - /* IUnknown */ - HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMDeviceEnumerator* pThis, const IID* const riid, void** ppObject); - ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMDeviceEnumerator* pThis); - ULONG (STDMETHODCALLTYPE * Release) (ma_IMMDeviceEnumerator* pThis); + pEvent->pContext = pContext; - /* IMMDeviceEnumerator */ - HRESULT (STDMETHODCALLTYPE * EnumAudioEndpoints) (ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, DWORD dwStateMask, ma_IMMDeviceCollection** ppDevices); - HRESULT (STDMETHODCALLTYPE * GetDefaultAudioEndpoint) (ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, ma_ERole role, ma_IMMDevice** ppEndpoint); - HRESULT (STDMETHODCALLTYPE * GetDevice) (ma_IMMDeviceEnumerator* pThis, LPCWSTR pID, ma_IMMDevice** ppDevice); - HRESULT (STDMETHODCALLTYPE * RegisterEndpointNotificationCallback) (ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient); - HRESULT (STDMETHODCALLTYPE * UnregisterEndpointNotificationCallback)(ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient); - } ma_IMMDeviceEnumeratorVtbl; - struct ma_IMMDeviceEnumerator - { - ma_IMMDeviceEnumeratorVtbl* lpVtbl; - }; - HRESULT ma_IMMDeviceEnumerator_QueryInterface(ma_IMMDeviceEnumerator* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } - ULONG ma_IMMDeviceEnumerator_AddRef(ma_IMMDeviceEnumerator* pThis) { return pThis->lpVtbl->AddRef(pThis); } - ULONG ma_IMMDeviceEnumerator_Release(ma_IMMDeviceEnumerator* pThis) { return pThis->lpVtbl->Release(pThis); } - HRESULT ma_IMMDeviceEnumerator_EnumAudioEndpoints(ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, DWORD dwStateMask, ma_IMMDeviceCollection** ppDevices) { return pThis->lpVtbl->EnumAudioEndpoints(pThis, dataFlow, dwStateMask, ppDevices); } - HRESULT ma_IMMDeviceEnumerator_GetDefaultAudioEndpoint(ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, ma_ERole role, ma_IMMDevice** ppEndpoint) { return pThis->lpVtbl->GetDefaultAudioEndpoint(pThis, dataFlow, role, ppEndpoint); } - HRESULT ma_IMMDeviceEnumerator_GetDevice(ma_IMMDeviceEnumerator* pThis, LPCWSTR pID, ma_IMMDevice** ppDevice) { return pThis->lpVtbl->GetDevice(pThis, pID, ppDevice); } - HRESULT ma_IMMDeviceEnumerator_RegisterEndpointNotificationCallback(ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient) { return pThis->lpVtbl->RegisterEndpointNotificationCallback(pThis, pClient); } - HRESULT ma_IMMDeviceEnumerator_UnregisterEndpointNotificationCallback(ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient) { return pThis->lpVtbl->UnregisterEndpointNotificationCallback(pThis, pClient); } +#ifdef MA_WIN32 + return ma_event_init__win32(pContext, pEvent); +#endif +#ifdef MA_POSIX + return ma_event_init__posix(pContext, pEvent); +#endif +} +void ma_event_uninit(ma_event* pEvent) +{ + if (pEvent == NULL || pEvent->pContext == NULL) { + return; + } - /* IMMDeviceCollection */ - typedef struct - { - /* IUnknown */ - HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMDeviceCollection* pThis, const IID* const riid, void** ppObject); - ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMDeviceCollection* pThis); - ULONG (STDMETHODCALLTYPE * Release) (ma_IMMDeviceCollection* pThis); +#ifdef MA_WIN32 + ma_event_uninit__win32(pEvent); +#endif +#ifdef MA_POSIX + ma_event_uninit__posix(pEvent); +#endif +} - /* IMMDeviceCollection */ - HRESULT (STDMETHODCALLTYPE * GetCount)(ma_IMMDeviceCollection* pThis, UINT* pDevices); - HRESULT (STDMETHODCALLTYPE * Item) (ma_IMMDeviceCollection* pThis, UINT nDevice, ma_IMMDevice** ppDevice); - } ma_IMMDeviceCollectionVtbl; - struct ma_IMMDeviceCollection - { - ma_IMMDeviceCollectionVtbl* lpVtbl; - }; - HRESULT ma_IMMDeviceCollection_QueryInterface(ma_IMMDeviceCollection* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } - ULONG ma_IMMDeviceCollection_AddRef(ma_IMMDeviceCollection* pThis) { return pThis->lpVtbl->AddRef(pThis); } - ULONG ma_IMMDeviceCollection_Release(ma_IMMDeviceCollection* pThis) { return pThis->lpVtbl->Release(pThis); } - HRESULT ma_IMMDeviceCollection_GetCount(ma_IMMDeviceCollection* pThis, UINT* pDevices) { return pThis->lpVtbl->GetCount(pThis, pDevices); } - HRESULT ma_IMMDeviceCollection_Item(ma_IMMDeviceCollection* pThis, UINT nDevice, ma_IMMDevice** ppDevice) { return pThis->lpVtbl->Item(pThis, nDevice, ppDevice); } +ma_bool32 ma_event_wait(ma_event* pEvent) +{ + if (pEvent == NULL || pEvent->pContext == NULL) { + return MA_FALSE; + } +#ifdef MA_WIN32 + return ma_event_wait__win32(pEvent); +#endif +#ifdef MA_POSIX + return ma_event_wait__posix(pEvent); +#endif +} - /* IMMDevice */ - typedef struct - { - /* IUnknown */ - HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMDevice* pThis, const IID* const riid, void** ppObject); - ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMDevice* pThis); - ULONG (STDMETHODCALLTYPE * Release) (ma_IMMDevice* pThis); +ma_bool32 ma_event_signal(ma_event* pEvent) +{ + if (pEvent == NULL || pEvent->pContext == NULL) { + return MA_FALSE; + } - /* IMMDevice */ - HRESULT (STDMETHODCALLTYPE * Activate) (ma_IMMDevice* pThis, const IID* const iid, DWORD dwClsCtx, PROPVARIANT* pActivationParams, void** ppInterface); - HRESULT (STDMETHODCALLTYPE * OpenPropertyStore)(ma_IMMDevice* pThis, DWORD stgmAccess, ma_IPropertyStore** ppProperties); - HRESULT (STDMETHODCALLTYPE * GetId) (ma_IMMDevice* pThis, LPWSTR *pID); - HRESULT (STDMETHODCALLTYPE * GetState) (ma_IMMDevice* pThis, DWORD *pState); - } ma_IMMDeviceVtbl; - struct ma_IMMDevice - { - ma_IMMDeviceVtbl* lpVtbl; - }; - HRESULT ma_IMMDevice_QueryInterface(ma_IMMDevice* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } - ULONG ma_IMMDevice_AddRef(ma_IMMDevice* pThis) { return pThis->lpVtbl->AddRef(pThis); } - ULONG ma_IMMDevice_Release(ma_IMMDevice* pThis) { return pThis->lpVtbl->Release(pThis); } - HRESULT ma_IMMDevice_Activate(ma_IMMDevice* pThis, const IID* const iid, DWORD dwClsCtx, PROPVARIANT* pActivationParams, void** ppInterface) { return pThis->lpVtbl->Activate(pThis, iid, dwClsCtx, pActivationParams, ppInterface); } - HRESULT ma_IMMDevice_OpenPropertyStore(ma_IMMDevice* pThis, DWORD stgmAccess, ma_IPropertyStore** ppProperties) { return pThis->lpVtbl->OpenPropertyStore(pThis, stgmAccess, ppProperties); } - HRESULT ma_IMMDevice_GetId(ma_IMMDevice* pThis, LPWSTR *pID) { return pThis->lpVtbl->GetId(pThis, pID); } - HRESULT ma_IMMDevice_GetState(ma_IMMDevice* pThis, DWORD *pState) { return pThis->lpVtbl->GetState(pThis, pState); } -#else - /* IActivateAudioInterfaceAsyncOperation */ - typedef struct - { - /* IUnknown */ - HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IActivateAudioInterfaceAsyncOperation* pThis, const IID* const riid, void** ppObject); - ULONG (STDMETHODCALLTYPE * AddRef) (ma_IActivateAudioInterfaceAsyncOperation* pThis); - ULONG (STDMETHODCALLTYPE * Release) (ma_IActivateAudioInterfaceAsyncOperation* pThis); +#ifdef MA_WIN32 + return ma_event_signal__win32(pEvent); +#endif +#ifdef MA_POSIX + return ma_event_signal__posix(pEvent); +#endif +} - /* IActivateAudioInterfaceAsyncOperation */ - HRESULT (STDMETHODCALLTYPE * GetActivateResult)(ma_IActivateAudioInterfaceAsyncOperation* pThis, HRESULT *pActivateResult, ma_IUnknown** ppActivatedInterface); - } ma_IActivateAudioInterfaceAsyncOperationVtbl; - struct ma_IActivateAudioInterfaceAsyncOperation - { - ma_IActivateAudioInterfaceAsyncOperationVtbl* lpVtbl; - }; - HRESULT ma_IActivateAudioInterfaceAsyncOperation_QueryInterface(ma_IActivateAudioInterfaceAsyncOperation* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } - ULONG ma_IActivateAudioInterfaceAsyncOperation_AddRef(ma_IActivateAudioInterfaceAsyncOperation* pThis) { return pThis->lpVtbl->AddRef(pThis); } - ULONG ma_IActivateAudioInterfaceAsyncOperation_Release(ma_IActivateAudioInterfaceAsyncOperation* pThis) { return pThis->lpVtbl->Release(pThis); } - HRESULT ma_IActivateAudioInterfaceAsyncOperation_GetActivateResult(ma_IActivateAudioInterfaceAsyncOperation* pThis, HRESULT *pActivateResult, ma_IUnknown** ppActivatedInterface) { return pThis->lpVtbl->GetActivateResult(pThis, pActivateResult, ppActivatedInterface); } + +ma_result ma_semaphore_init(ma_context* pContext, int initialValue, ma_semaphore* pSemaphore) +{ + if (pContext == NULL || pSemaphore == NULL) { + return MA_INVALID_ARGS; + } + +#ifdef MA_WIN32 + return ma_semaphore_init__win32(pContext, initialValue, pSemaphore); #endif +#ifdef MA_POSIX + return ma_semaphore_init__posix(pContext, initialValue, pSemaphore); +#endif +} -/* IPropertyStore */ -typedef struct +void ma_semaphore_uninit(ma_semaphore* pSemaphore) { - /* IUnknown */ - HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IPropertyStore* pThis, const IID* const riid, void** ppObject); - ULONG (STDMETHODCALLTYPE * AddRef) (ma_IPropertyStore* pThis); - ULONG (STDMETHODCALLTYPE * Release) (ma_IPropertyStore* pThis); + if (pSemaphore == NULL) { + return; + } - /* IPropertyStore */ - HRESULT (STDMETHODCALLTYPE * GetCount)(ma_IPropertyStore* pThis, DWORD* pPropCount); - HRESULT (STDMETHODCALLTYPE * GetAt) (ma_IPropertyStore* pThis, DWORD propIndex, PROPERTYKEY* pPropKey); - HRESULT (STDMETHODCALLTYPE * GetValue)(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, PROPVARIANT* pPropVar); - HRESULT (STDMETHODCALLTYPE * SetValue)(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, const PROPVARIANT* const pPropVar); - HRESULT (STDMETHODCALLTYPE * Commit) (ma_IPropertyStore* pThis); -} ma_IPropertyStoreVtbl; -struct ma_IPropertyStore +#ifdef MA_WIN32 + ma_semaphore_uninit__win32(pSemaphore); +#endif +#ifdef MA_POSIX + ma_semaphore_uninit__posix(pSemaphore); +#endif +} + +ma_bool32 ma_semaphore_wait(ma_semaphore* pSemaphore) { - ma_IPropertyStoreVtbl* lpVtbl; -}; -HRESULT ma_IPropertyStore_QueryInterface(ma_IPropertyStore* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } -ULONG ma_IPropertyStore_AddRef(ma_IPropertyStore* pThis) { return pThis->lpVtbl->AddRef(pThis); } -ULONG ma_IPropertyStore_Release(ma_IPropertyStore* pThis) { return pThis->lpVtbl->Release(pThis); } -HRESULT ma_IPropertyStore_GetCount(ma_IPropertyStore* pThis, DWORD* pPropCount) { return pThis->lpVtbl->GetCount(pThis, pPropCount); } -HRESULT ma_IPropertyStore_GetAt(ma_IPropertyStore* pThis, DWORD propIndex, PROPERTYKEY* pPropKey) { return pThis->lpVtbl->GetAt(pThis, propIndex, pPropKey); } -HRESULT ma_IPropertyStore_GetValue(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, PROPVARIANT* pPropVar) { return pThis->lpVtbl->GetValue(pThis, pKey, pPropVar); } -HRESULT ma_IPropertyStore_SetValue(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, const PROPVARIANT* const pPropVar) { return pThis->lpVtbl->SetValue(pThis, pKey, pPropVar); } -HRESULT ma_IPropertyStore_Commit(ma_IPropertyStore* pThis) { return pThis->lpVtbl->Commit(pThis); } + if (pSemaphore == NULL) { + return MA_FALSE; + } +#ifdef MA_WIN32 + return ma_semaphore_wait__win32(pSemaphore); +#endif +#ifdef MA_POSIX + return ma_semaphore_wait__posix(pSemaphore); +#endif +} -/* IAudioClient */ -typedef struct +ma_bool32 ma_semaphore_release(ma_semaphore* pSemaphore) { - /* IUnknown */ - HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioClient* pThis, const IID* const riid, void** ppObject); - ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioClient* pThis); - ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioClient* pThis); + if (pSemaphore == NULL) { + return MA_FALSE; + } - /* IAudioClient */ - HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid); - HRESULT (STDMETHODCALLTYPE * GetBufferSize) (ma_IAudioClient* pThis, ma_uint32* pNumBufferFrames); - HRESULT (STDMETHODCALLTYPE * GetStreamLatency) (ma_IAudioClient* pThis, MA_REFERENCE_TIME* pLatency); - HRESULT (STDMETHODCALLTYPE * GetCurrentPadding)(ma_IAudioClient* pThis, ma_uint32* pNumPaddingFrames); - HRESULT (STDMETHODCALLTYPE * IsFormatSupported)(ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch); - HRESULT (STDMETHODCALLTYPE * GetMixFormat) (ma_IAudioClient* pThis, WAVEFORMATEX** ppDeviceFormat); - HRESULT (STDMETHODCALLTYPE * GetDevicePeriod) (ma_IAudioClient* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod); - HRESULT (STDMETHODCALLTYPE * Start) (ma_IAudioClient* pThis); - HRESULT (STDMETHODCALLTYPE * Stop) (ma_IAudioClient* pThis); - HRESULT (STDMETHODCALLTYPE * Reset) (ma_IAudioClient* pThis); - HRESULT (STDMETHODCALLTYPE * SetEventHandle) (ma_IAudioClient* pThis, HANDLE eventHandle); - HRESULT (STDMETHODCALLTYPE * GetService) (ma_IAudioClient* pThis, const IID* const riid, void** pp); -} ma_IAudioClientVtbl; -struct ma_IAudioClient -{ - ma_IAudioClientVtbl* lpVtbl; -}; -HRESULT ma_IAudioClient_QueryInterface(ma_IAudioClient* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } -ULONG ma_IAudioClient_AddRef(ma_IAudioClient* pThis) { return pThis->lpVtbl->AddRef(pThis); } -ULONG ma_IAudioClient_Release(ma_IAudioClient* pThis) { return pThis->lpVtbl->Release(pThis); } -HRESULT ma_IAudioClient_Initialize(ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid) { return pThis->lpVtbl->Initialize(pThis, shareMode, streamFlags, bufferDuration, periodicity, pFormat, pAudioSessionGuid); } -HRESULT ma_IAudioClient_GetBufferSize(ma_IAudioClient* pThis, ma_uint32* pNumBufferFrames) { return pThis->lpVtbl->GetBufferSize(pThis, pNumBufferFrames); } -HRESULT ma_IAudioClient_GetStreamLatency(ma_IAudioClient* pThis, MA_REFERENCE_TIME* pLatency) { return pThis->lpVtbl->GetStreamLatency(pThis, pLatency); } -HRESULT ma_IAudioClient_GetCurrentPadding(ma_IAudioClient* pThis, ma_uint32* pNumPaddingFrames) { return pThis->lpVtbl->GetCurrentPadding(pThis, pNumPaddingFrames); } -HRESULT ma_IAudioClient_IsFormatSupported(ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch) { return pThis->lpVtbl->IsFormatSupported(pThis, shareMode, pFormat, ppClosestMatch); } -HRESULT ma_IAudioClient_GetMixFormat(ma_IAudioClient* pThis, WAVEFORMATEX** ppDeviceFormat) { return pThis->lpVtbl->GetMixFormat(pThis, ppDeviceFormat); } -HRESULT ma_IAudioClient_GetDevicePeriod(ma_IAudioClient* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod) { return pThis->lpVtbl->GetDevicePeriod(pThis, pDefaultDevicePeriod, pMinimumDevicePeriod); } -HRESULT ma_IAudioClient_Start(ma_IAudioClient* pThis) { return pThis->lpVtbl->Start(pThis); } -HRESULT ma_IAudioClient_Stop(ma_IAudioClient* pThis) { return pThis->lpVtbl->Stop(pThis); } -HRESULT ma_IAudioClient_Reset(ma_IAudioClient* pThis) { return pThis->lpVtbl->Reset(pThis); } -HRESULT ma_IAudioClient_SetEventHandle(ma_IAudioClient* pThis, HANDLE eventHandle) { return pThis->lpVtbl->SetEventHandle(pThis, eventHandle); } -HRESULT ma_IAudioClient_GetService(ma_IAudioClient* pThis, const IID* const riid, void** pp) { return pThis->lpVtbl->GetService(pThis, riid, pp); } +#ifdef MA_WIN32 + return ma_semaphore_release__win32(pSemaphore); +#endif +#ifdef MA_POSIX + return ma_semaphore_release__posix(pSemaphore); +#endif +} -/* IAudioClient2 */ -typedef struct + +#if 0 +ma_uint32 ma_get_closest_standard_sample_rate(ma_uint32 sampleRateIn) { - /* IUnknown */ - HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioClient2* pThis, const IID* const riid, void** ppObject); - ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioClient2* pThis); - ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioClient2* pThis); + ma_uint32 closestRate = 0; + ma_uint32 closestDiff = 0xFFFFFFFF; + size_t iStandardRate; - /* IAudioClient */ - HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid); - HRESULT (STDMETHODCALLTYPE * GetBufferSize) (ma_IAudioClient2* pThis, ma_uint32* pNumBufferFrames); - HRESULT (STDMETHODCALLTYPE * GetStreamLatency) (ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pLatency); - HRESULT (STDMETHODCALLTYPE * GetCurrentPadding)(ma_IAudioClient2* pThis, ma_uint32* pNumPaddingFrames); - HRESULT (STDMETHODCALLTYPE * IsFormatSupported)(ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch); - HRESULT (STDMETHODCALLTYPE * GetMixFormat) (ma_IAudioClient2* pThis, WAVEFORMATEX** ppDeviceFormat); - HRESULT (STDMETHODCALLTYPE * GetDevicePeriod) (ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod); - HRESULT (STDMETHODCALLTYPE * Start) (ma_IAudioClient2* pThis); - HRESULT (STDMETHODCALLTYPE * Stop) (ma_IAudioClient2* pThis); - HRESULT (STDMETHODCALLTYPE * Reset) (ma_IAudioClient2* pThis); - HRESULT (STDMETHODCALLTYPE * SetEventHandle) (ma_IAudioClient2* pThis, HANDLE eventHandle); - HRESULT (STDMETHODCALLTYPE * GetService) (ma_IAudioClient2* pThis, const IID* const riid, void** pp); + for (iStandardRate = 0; iStandardRate < ma_countof(g_maStandardSampleRatePriorities); ++iStandardRate) { + ma_uint32 standardRate = g_maStandardSampleRatePriorities[iStandardRate]; + ma_uint32 diff; - /* IAudioClient2 */ - HRESULT (STDMETHODCALLTYPE * IsOffloadCapable) (ma_IAudioClient2* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable); - HRESULT (STDMETHODCALLTYPE * SetClientProperties)(ma_IAudioClient2* pThis, const ma_AudioClientProperties* pProperties); - HRESULT (STDMETHODCALLTYPE * GetBufferSizeLimits)(ma_IAudioClient2* pThis, const WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration); -} ma_IAudioClient2Vtbl; -struct ma_IAudioClient2 -{ - ma_IAudioClient2Vtbl* lpVtbl; -}; -HRESULT ma_IAudioClient2_QueryInterface(ma_IAudioClient2* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } -ULONG ma_IAudioClient2_AddRef(ma_IAudioClient2* pThis) { return pThis->lpVtbl->AddRef(pThis); } -ULONG ma_IAudioClient2_Release(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Release(pThis); } -HRESULT ma_IAudioClient2_Initialize(ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid) { return pThis->lpVtbl->Initialize(pThis, shareMode, streamFlags, bufferDuration, periodicity, pFormat, pAudioSessionGuid); } -HRESULT ma_IAudioClient2_GetBufferSize(ma_IAudioClient2* pThis, ma_uint32* pNumBufferFrames) { return pThis->lpVtbl->GetBufferSize(pThis, pNumBufferFrames); } -HRESULT ma_IAudioClient2_GetStreamLatency(ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pLatency) { return pThis->lpVtbl->GetStreamLatency(pThis, pLatency); } -HRESULT ma_IAudioClient2_GetCurrentPadding(ma_IAudioClient2* pThis, ma_uint32* pNumPaddingFrames) { return pThis->lpVtbl->GetCurrentPadding(pThis, pNumPaddingFrames); } -HRESULT ma_IAudioClient2_IsFormatSupported(ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch) { return pThis->lpVtbl->IsFormatSupported(pThis, shareMode, pFormat, ppClosestMatch); } -HRESULT ma_IAudioClient2_GetMixFormat(ma_IAudioClient2* pThis, WAVEFORMATEX** ppDeviceFormat) { return pThis->lpVtbl->GetMixFormat(pThis, ppDeviceFormat); } -HRESULT ma_IAudioClient2_GetDevicePeriod(ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod) { return pThis->lpVtbl->GetDevicePeriod(pThis, pDefaultDevicePeriod, pMinimumDevicePeriod); } -HRESULT ma_IAudioClient2_Start(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Start(pThis); } -HRESULT ma_IAudioClient2_Stop(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Stop(pThis); } -HRESULT ma_IAudioClient2_Reset(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Reset(pThis); } -HRESULT ma_IAudioClient2_SetEventHandle(ma_IAudioClient2* pThis, HANDLE eventHandle) { return pThis->lpVtbl->SetEventHandle(pThis, eventHandle); } -HRESULT ma_IAudioClient2_GetService(ma_IAudioClient2* pThis, const IID* const riid, void** pp) { return pThis->lpVtbl->GetService(pThis, riid, pp); } -HRESULT ma_IAudioClient2_IsOffloadCapable(ma_IAudioClient2* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable) { return pThis->lpVtbl->IsOffloadCapable(pThis, category, pOffloadCapable); } -HRESULT ma_IAudioClient2_SetClientProperties(ma_IAudioClient2* pThis, const ma_AudioClientProperties* pProperties) { return pThis->lpVtbl->SetClientProperties(pThis, pProperties); } -HRESULT ma_IAudioClient2_GetBufferSizeLimits(ma_IAudioClient2* pThis, const WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration) { return pThis->lpVtbl->GetBufferSizeLimits(pThis, pFormat, eventDriven, pMinBufferDuration, pMaxBufferDuration); } + if (sampleRateIn > standardRate) { + diff = sampleRateIn - standardRate; + } else { + diff = standardRate - sampleRateIn; + } + if (diff == 0) { + return standardRate; /* The input sample rate is a standard rate. */ + } -/* IAudioClient3 */ -typedef struct -{ - /* IUnknown */ - HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioClient3* pThis, const IID* const riid, void** ppObject); - ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioClient3* pThis); - ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioClient3* pThis); + if (closestDiff > diff) { + closestDiff = diff; + closestRate = standardRate; + } + } - /* IAudioClient */ - HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid); - HRESULT (STDMETHODCALLTYPE * GetBufferSize) (ma_IAudioClient3* pThis, ma_uint32* pNumBufferFrames); - HRESULT (STDMETHODCALLTYPE * GetStreamLatency) (ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pLatency); - HRESULT (STDMETHODCALLTYPE * GetCurrentPadding)(ma_IAudioClient3* pThis, ma_uint32* pNumPaddingFrames); - HRESULT (STDMETHODCALLTYPE * IsFormatSupported)(ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch); - HRESULT (STDMETHODCALLTYPE * GetMixFormat) (ma_IAudioClient3* pThis, WAVEFORMATEX** ppDeviceFormat); - HRESULT (STDMETHODCALLTYPE * GetDevicePeriod) (ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod); - HRESULT (STDMETHODCALLTYPE * Start) (ma_IAudioClient3* pThis); - HRESULT (STDMETHODCALLTYPE * Stop) (ma_IAudioClient3* pThis); - HRESULT (STDMETHODCALLTYPE * Reset) (ma_IAudioClient3* pThis); - HRESULT (STDMETHODCALLTYPE * SetEventHandle) (ma_IAudioClient3* pThis, HANDLE eventHandle); - HRESULT (STDMETHODCALLTYPE * GetService) (ma_IAudioClient3* pThis, const IID* const riid, void** pp); + return closestRate; +} +#endif - /* IAudioClient2 */ - HRESULT (STDMETHODCALLTYPE * IsOffloadCapable) (ma_IAudioClient3* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable); - HRESULT (STDMETHODCALLTYPE * SetClientProperties)(ma_IAudioClient3* pThis, const ma_AudioClientProperties* pProperties); - HRESULT (STDMETHODCALLTYPE * GetBufferSizeLimits)(ma_IAudioClient3* pThis, const WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration); +ma_uint32 ma_scale_buffer_size(ma_uint32 baseBufferSize, float scale) +{ + return ma_max(1, (ma_uint32)(baseBufferSize*scale)); +} - /* IAudioClient3 */ - HRESULT (STDMETHODCALLTYPE * GetSharedModeEnginePeriod) (ma_IAudioClient3* pThis, const WAVEFORMATEX* pFormat, UINT32* pDefaultPeriodInFrames, UINT32* pFundamentalPeriodInFrames, UINT32* pMinPeriodInFrames, UINT32* pMaxPeriodInFrames); - HRESULT (STDMETHODCALLTYPE * GetCurrentSharedModeEnginePeriod)(ma_IAudioClient3* pThis, WAVEFORMATEX** ppFormat, UINT32* pCurrentPeriodInFrames); - HRESULT (STDMETHODCALLTYPE * InitializeSharedAudioStream) (ma_IAudioClient3* pThis, DWORD streamFlags, UINT32 periodInFrames, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid); -} ma_IAudioClient3Vtbl; -struct ma_IAudioClient3 +ma_uint32 ma_calculate_buffer_size_in_milliseconds_from_frames(ma_uint32 bufferSizeInFrames, ma_uint32 sampleRate) { - ma_IAudioClient3Vtbl* lpVtbl; -}; -HRESULT ma_IAudioClient3_QueryInterface(ma_IAudioClient3* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } -ULONG ma_IAudioClient3_AddRef(ma_IAudioClient3* pThis) { return pThis->lpVtbl->AddRef(pThis); } -ULONG ma_IAudioClient3_Release(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Release(pThis); } -HRESULT ma_IAudioClient3_Initialize(ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid) { return pThis->lpVtbl->Initialize(pThis, shareMode, streamFlags, bufferDuration, periodicity, pFormat, pAudioSessionGuid); } -HRESULT ma_IAudioClient3_GetBufferSize(ma_IAudioClient3* pThis, ma_uint32* pNumBufferFrames) { return pThis->lpVtbl->GetBufferSize(pThis, pNumBufferFrames); } -HRESULT ma_IAudioClient3_GetStreamLatency(ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pLatency) { return pThis->lpVtbl->GetStreamLatency(pThis, pLatency); } -HRESULT ma_IAudioClient3_GetCurrentPadding(ma_IAudioClient3* pThis, ma_uint32* pNumPaddingFrames) { return pThis->lpVtbl->GetCurrentPadding(pThis, pNumPaddingFrames); } -HRESULT ma_IAudioClient3_IsFormatSupported(ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch) { return pThis->lpVtbl->IsFormatSupported(pThis, shareMode, pFormat, ppClosestMatch); } -HRESULT ma_IAudioClient3_GetMixFormat(ma_IAudioClient3* pThis, WAVEFORMATEX** ppDeviceFormat) { return pThis->lpVtbl->GetMixFormat(pThis, ppDeviceFormat); } -HRESULT ma_IAudioClient3_GetDevicePeriod(ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod) { return pThis->lpVtbl->GetDevicePeriod(pThis, pDefaultDevicePeriod, pMinimumDevicePeriod); } -HRESULT ma_IAudioClient3_Start(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Start(pThis); } -HRESULT ma_IAudioClient3_Stop(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Stop(pThis); } -HRESULT ma_IAudioClient3_Reset(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Reset(pThis); } -HRESULT ma_IAudioClient3_SetEventHandle(ma_IAudioClient3* pThis, HANDLE eventHandle) { return pThis->lpVtbl->SetEventHandle(pThis, eventHandle); } -HRESULT ma_IAudioClient3_GetService(ma_IAudioClient3* pThis, const IID* const riid, void** pp) { return pThis->lpVtbl->GetService(pThis, riid, pp); } -HRESULT ma_IAudioClient3_IsOffloadCapable(ma_IAudioClient3* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable) { return pThis->lpVtbl->IsOffloadCapable(pThis, category, pOffloadCapable); } -HRESULT ma_IAudioClient3_SetClientProperties(ma_IAudioClient3* pThis, const ma_AudioClientProperties* pProperties) { return pThis->lpVtbl->SetClientProperties(pThis, pProperties); } -HRESULT ma_IAudioClient3_GetBufferSizeLimits(ma_IAudioClient3* pThis, const WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration) { return pThis->lpVtbl->GetBufferSizeLimits(pThis, pFormat, eventDriven, pMinBufferDuration, pMaxBufferDuration); } -HRESULT ma_IAudioClient3_GetSharedModeEnginePeriod(ma_IAudioClient3* pThis, const WAVEFORMATEX* pFormat, UINT32* pDefaultPeriodInFrames, UINT32* pFundamentalPeriodInFrames, UINT32* pMinPeriodInFrames, UINT32* pMaxPeriodInFrames) { return pThis->lpVtbl->GetSharedModeEnginePeriod(pThis, pFormat, pDefaultPeriodInFrames, pFundamentalPeriodInFrames, pMinPeriodInFrames, pMaxPeriodInFrames); } -HRESULT ma_IAudioClient3_GetCurrentSharedModeEnginePeriod(ma_IAudioClient3* pThis, WAVEFORMATEX** ppFormat, UINT32* pCurrentPeriodInFrames) { return pThis->lpVtbl->GetCurrentSharedModeEnginePeriod(pThis, ppFormat, pCurrentPeriodInFrames); } -HRESULT ma_IAudioClient3_InitializeSharedAudioStream(ma_IAudioClient3* pThis, DWORD streamFlags, UINT32 periodInFrames, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGUID) { return pThis->lpVtbl->InitializeSharedAudioStream(pThis, streamFlags, periodInFrames, pFormat, pAudioSessionGUID); } + return bufferSizeInFrames / (sampleRate/1000); +} +ma_uint32 ma_calculate_buffer_size_in_frames_from_milliseconds(ma_uint32 bufferSizeInMilliseconds, ma_uint32 sampleRate) +{ + return bufferSizeInMilliseconds * (sampleRate/1000); +} -/* IAudioRenderClient */ -typedef struct +void ma_zero_pcm_frames(void* p, ma_uint32 frameCount, ma_format format, ma_uint32 channels) { - /* IUnknown */ - HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioRenderClient* pThis, const IID* const riid, void** ppObject); - ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioRenderClient* pThis); - ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioRenderClient* pThis); + MA_ZERO_MEMORY(p, frameCount * ma_get_bytes_per_frame(format, channels)); +} - /* IAudioRenderClient */ - HRESULT (STDMETHODCALLTYPE * GetBuffer) (ma_IAudioRenderClient* pThis, ma_uint32 numFramesRequested, BYTE** ppData); - HRESULT (STDMETHODCALLTYPE * ReleaseBuffer)(ma_IAudioRenderClient* pThis, ma_uint32 numFramesWritten, DWORD dwFlags); -} ma_IAudioRenderClientVtbl; -struct ma_IAudioRenderClient +void ma_clip_samples_f32(float* p, ma_uint32 sampleCount) { - ma_IAudioRenderClientVtbl* lpVtbl; -}; -HRESULT ma_IAudioRenderClient_QueryInterface(ma_IAudioRenderClient* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } -ULONG ma_IAudioRenderClient_AddRef(ma_IAudioRenderClient* pThis) { return pThis->lpVtbl->AddRef(pThis); } -ULONG ma_IAudioRenderClient_Release(ma_IAudioRenderClient* pThis) { return pThis->lpVtbl->Release(pThis); } -HRESULT ma_IAudioRenderClient_GetBuffer(ma_IAudioRenderClient* pThis, ma_uint32 numFramesRequested, BYTE** ppData) { return pThis->lpVtbl->GetBuffer(pThis, numFramesRequested, ppData); } -HRESULT ma_IAudioRenderClient_ReleaseBuffer(ma_IAudioRenderClient* pThis, ma_uint32 numFramesWritten, DWORD dwFlags) { return pThis->lpVtbl->ReleaseBuffer(pThis, numFramesWritten, dwFlags); } + ma_uint32 iSample; + /* TODO: Research a branchless SSE implementation. */ + for (iSample = 0; iSample < sampleCount; iSample += 1) { + p[iSample] = ma_clip_f32(p[iSample]); + } +} -/* IAudioCaptureClient */ -typedef struct -{ - /* IUnknown */ - HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioCaptureClient* pThis, const IID* const riid, void** ppObject); - ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioCaptureClient* pThis); - ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioCaptureClient* pThis); - /* IAudioRenderClient */ - HRESULT (STDMETHODCALLTYPE * GetBuffer) (ma_IAudioCaptureClient* pThis, BYTE** ppData, ma_uint32* pNumFramesToRead, DWORD* pFlags, ma_uint64* pDevicePosition, ma_uint64* pQPCPosition); - HRESULT (STDMETHODCALLTYPE * ReleaseBuffer) (ma_IAudioCaptureClient* pThis, ma_uint32 numFramesRead); - HRESULT (STDMETHODCALLTYPE * GetNextPacketSize)(ma_IAudioCaptureClient* pThis, ma_uint32* pNumFramesInNextPacket); -} ma_IAudioCaptureClientVtbl; -struct ma_IAudioCaptureClient +void ma_copy_and_apply_volume_factor_u8(ma_uint8* pSamplesOut, const ma_uint8* pSamplesIn, ma_uint32 sampleCount, float factor) { - ma_IAudioCaptureClientVtbl* lpVtbl; -}; -HRESULT ma_IAudioCaptureClient_QueryInterface(ma_IAudioCaptureClient* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } -ULONG ma_IAudioCaptureClient_AddRef(ma_IAudioCaptureClient* pThis) { return pThis->lpVtbl->AddRef(pThis); } -ULONG ma_IAudioCaptureClient_Release(ma_IAudioCaptureClient* pThis) { return pThis->lpVtbl->Release(pThis); } -HRESULT ma_IAudioCaptureClient_GetBuffer(ma_IAudioCaptureClient* pThis, BYTE** ppData, ma_uint32* pNumFramesToRead, DWORD* pFlags, ma_uint64* pDevicePosition, ma_uint64* pQPCPosition) { return pThis->lpVtbl->GetBuffer(pThis, ppData, pNumFramesToRead, pFlags, pDevicePosition, pQPCPosition); } -HRESULT ma_IAudioCaptureClient_ReleaseBuffer(ma_IAudioCaptureClient* pThis, ma_uint32 numFramesRead) { return pThis->lpVtbl->ReleaseBuffer(pThis, numFramesRead); } -HRESULT ma_IAudioCaptureClient_GetNextPacketSize(ma_IAudioCaptureClient* pThis, ma_uint32* pNumFramesInNextPacket) { return pThis->lpVtbl->GetNextPacketSize(pThis, pNumFramesInNextPacket); } + ma_uint32 iSample; -#ifndef MA_WIN32_DESKTOP -#include -typedef struct ma_completion_handler_uwp ma_completion_handler_uwp; + if (pSamplesOut == NULL || pSamplesIn == NULL) { + return; + } -typedef struct + for (iSample = 0; iSample < sampleCount; iSample += 1) { + pSamplesOut[iSample] = (ma_uint8)(pSamplesIn[iSample] * factor); + } +} + +void ma_copy_and_apply_volume_factor_s16(ma_int16* pSamplesOut, const ma_int16* pSamplesIn, ma_uint32 sampleCount, float factor) { - /* IUnknown */ - HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_completion_handler_uwp* pThis, const IID* const riid, void** ppObject); - ULONG (STDMETHODCALLTYPE * AddRef) (ma_completion_handler_uwp* pThis); - ULONG (STDMETHODCALLTYPE * Release) (ma_completion_handler_uwp* pThis); + ma_uint32 iSample; - /* IActivateAudioInterfaceCompletionHandler */ - HRESULT (STDMETHODCALLTYPE * ActivateCompleted)(ma_completion_handler_uwp* pThis, ma_IActivateAudioInterfaceAsyncOperation* pActivateOperation); -} ma_completion_handler_uwp_vtbl; -struct ma_completion_handler_uwp + if (pSamplesOut == NULL || pSamplesIn == NULL) { + return; + } + + for (iSample = 0; iSample < sampleCount; iSample += 1) { + pSamplesOut[iSample] = (ma_int16)(pSamplesIn[iSample] * factor); + } +} + +void ma_copy_and_apply_volume_factor_s24(void* pSamplesOut, const void* pSamplesIn, ma_uint32 sampleCount, float factor) { - ma_completion_handler_uwp_vtbl* lpVtbl; - ma_uint32 counter; - HANDLE hEvent; -}; + ma_uint32 iSample; + ma_uint8* pSamplesOut8; + ma_uint8* pSamplesIn8; + + if (pSamplesOut == NULL || pSamplesIn == NULL) { + return; + } + + pSamplesOut8 = (ma_uint8*)pSamplesOut; + pSamplesIn8 = (ma_uint8*)pSamplesIn; + + for (iSample = 0; iSample < sampleCount; iSample += 1) { + ma_int32 sampleS32; + + sampleS32 = (ma_int32)(((ma_uint32)(pSamplesIn8[iSample*3+0]) << 8) | ((ma_uint32)(pSamplesIn8[iSample*3+1]) << 16) | ((ma_uint32)(pSamplesIn8[iSample*3+2])) << 24); + sampleS32 = (ma_int32)(sampleS32 * factor); -HRESULT STDMETHODCALLTYPE ma_completion_handler_uwp_QueryInterface(ma_completion_handler_uwp* pThis, const IID* const riid, void** ppObject) -{ - /* - We need to "implement" IAgileObject which is just an indicator that's used internally by WASAPI for some multithreading management. To - "implement" this, we just make sure we return pThis when the IAgileObject is requested. - */ - if (!ma_is_guid_equal(riid, &MA_IID_IUnknown) && !ma_is_guid_equal(riid, &MA_IID_IActivateAudioInterfaceCompletionHandler) && !ma_is_guid_equal(riid, &MA_IID_IAgileObject)) { - *ppObject = NULL; - return E_NOINTERFACE; + pSamplesOut8[iSample*3+0] = (ma_uint8)(((ma_uint32)sampleS32 & 0x0000FF00) >> 8); + pSamplesOut8[iSample*3+1] = (ma_uint8)(((ma_uint32)sampleS32 & 0x00FF0000) >> 16); + pSamplesOut8[iSample*3+2] = (ma_uint8)(((ma_uint32)sampleS32 & 0xFF000000) >> 24); } - - /* Getting here means the IID is IUnknown or IMMNotificationClient. */ - *ppObject = (void*)pThis; - ((ma_completion_handler_uwp_vtbl*)pThis->lpVtbl)->AddRef(pThis); - return S_OK; } -ULONG STDMETHODCALLTYPE ma_completion_handler_uwp_AddRef(ma_completion_handler_uwp* pThis) +void ma_copy_and_apply_volume_factor_s32(ma_int32* pSamplesOut, const ma_int32* pSamplesIn, ma_uint32 sampleCount, float factor) { - return (ULONG)ma_atomic_increment_32(&pThis->counter); + ma_uint32 iSample; + + if (pSamplesOut == NULL || pSamplesIn == NULL) { + return; + } + + for (iSample = 0; iSample < sampleCount; iSample += 1) { + pSamplesOut[iSample] = (ma_int32)(pSamplesIn[iSample] * factor); + } } -ULONG STDMETHODCALLTYPE ma_completion_handler_uwp_Release(ma_completion_handler_uwp* pThis) +void ma_copy_and_apply_volume_factor_f32(float* pSamplesOut, const float* pSamplesIn, ma_uint32 sampleCount, float factor) { - ma_uint32 newRefCount = ma_atomic_decrement_32(&pThis->counter); - if (newRefCount == 0) { - return 0; /* We don't free anything here because we never allocate the object on the heap. */ + ma_uint32 iSample; + + if (pSamplesOut == NULL || pSamplesIn == NULL) { + return; } - return (ULONG)newRefCount; + for (iSample = 0; iSample < sampleCount; iSample += 1) { + pSamplesOut[iSample] = pSamplesIn[iSample] * factor; + } } -HRESULT STDMETHODCALLTYPE ma_completion_handler_uwp_ActivateCompleted(ma_completion_handler_uwp* pThis, ma_IActivateAudioInterfaceAsyncOperation* pActivateOperation) +void ma_apply_volume_factor_u8(ma_uint8* pSamples, ma_uint32 sampleCount, float factor) { - (void)pActivateOperation; - SetEvent(pThis->hEvent); - return S_OK; + ma_copy_and_apply_volume_factor_u8(pSamples, pSamples, sampleCount, factor); } +void ma_apply_volume_factor_s16(ma_int16* pSamples, ma_uint32 sampleCount, float factor) +{ + ma_copy_and_apply_volume_factor_s16(pSamples, pSamples, sampleCount, factor); +} -static ma_completion_handler_uwp_vtbl g_maCompletionHandlerVtblInstance = { - ma_completion_handler_uwp_QueryInterface, - ma_completion_handler_uwp_AddRef, - ma_completion_handler_uwp_Release, - ma_completion_handler_uwp_ActivateCompleted -}; - -ma_result ma_completion_handler_uwp_init(ma_completion_handler_uwp* pHandler) +void ma_apply_volume_factor_s24(void* pSamples, ma_uint32 sampleCount, float factor) { - ma_assert(pHandler != NULL); - ma_zero_object(pHandler); + ma_copy_and_apply_volume_factor_s24(pSamples, pSamples, sampleCount, factor); +} - pHandler->lpVtbl = &g_maCompletionHandlerVtblInstance; - pHandler->counter = 1; - pHandler->hEvent = CreateEventA(NULL, FALSE, FALSE, NULL); - if (pHandler->hEvent == NULL) { - return MA_ERROR; - } +void ma_apply_volume_factor_s32(ma_int32* pSamples, ma_uint32 sampleCount, float factor) +{ + ma_copy_and_apply_volume_factor_s32(pSamples, pSamples, sampleCount, factor); +} - return MA_SUCCESS; +void ma_apply_volume_factor_f32(float* pSamples, ma_uint32 sampleCount, float factor) +{ + ma_copy_and_apply_volume_factor_f32(pSamples, pSamples, sampleCount, factor); } -void ma_completion_handler_uwp_uninit(ma_completion_handler_uwp* pHandler) +void ma_copy_and_apply_volume_factor_pcm_frames_u8(ma_uint8* pPCMFramesOut, const ma_uint8* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor) { - if (pHandler->hEvent != NULL) { - CloseHandle(pHandler->hEvent); - } + ma_copy_and_apply_volume_factor_u8(pPCMFramesOut, pPCMFramesIn, frameCount*channels, factor); } -void ma_completion_handler_uwp_wait(ma_completion_handler_uwp* pHandler) +void ma_copy_and_apply_volume_factor_pcm_frames_s16(ma_int16* pPCMFramesOut, const ma_int16* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor) { - WaitForSingleObject(pHandler->hEvent, INFINITE); + ma_copy_and_apply_volume_factor_s16(pPCMFramesOut, pPCMFramesIn, frameCount*channels, factor); } -#endif /* !MA_WIN32_DESKTOP */ -/* We need a virtual table for our notification client object that's used for detecting changes to the default device. */ -#ifdef MA_WIN32_DESKTOP -HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_QueryInterface(ma_IMMNotificationClient* pThis, const IID* const riid, void** ppObject) +void ma_copy_and_apply_volume_factor_pcm_frames_s24(void* pPCMFramesOut, const void* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor) { - /* - We care about two interfaces - IUnknown and IMMNotificationClient. If the requested IID is something else - we just return E_NOINTERFACE. Otherwise we need to increment the reference counter and return S_OK. - */ - if (!ma_is_guid_equal(riid, &MA_IID_IUnknown) && !ma_is_guid_equal(riid, &MA_IID_IMMNotificationClient)) { - *ppObject = NULL; - return E_NOINTERFACE; - } + ma_copy_and_apply_volume_factor_s24(pPCMFramesOut, pPCMFramesIn, frameCount*channels, factor); +} - /* Getting here means the IID is IUnknown or IMMNotificationClient. */ - *ppObject = (void*)pThis; - ((ma_IMMNotificationClientVtbl*)pThis->lpVtbl)->AddRef(pThis); - return S_OK; +void ma_copy_and_apply_volume_factor_pcm_frames_s32(ma_int32* pPCMFramesOut, const ma_int32* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor) +{ + ma_copy_and_apply_volume_factor_s32(pPCMFramesOut, pPCMFramesIn, frameCount*channels, factor); } -ULONG STDMETHODCALLTYPE ma_IMMNotificationClient_AddRef(ma_IMMNotificationClient* pThis) +void ma_copy_and_apply_volume_factor_pcm_frames_f32(float* pPCMFramesOut, const float* pPCMFramesIn, ma_uint32 frameCount, ma_uint32 channels, float factor) { - return (ULONG)ma_atomic_increment_32(&pThis->counter); + ma_copy_and_apply_volume_factor_f32(pPCMFramesOut, pPCMFramesIn, frameCount*channels, factor); } -ULONG STDMETHODCALLTYPE ma_IMMNotificationClient_Release(ma_IMMNotificationClient* pThis) +void ma_copy_and_apply_volume_factor_pcm_frames(void* pPCMFramesOut, const void* pPCMFramesIn, ma_uint32 frameCount, ma_format format, ma_uint32 channels, float factor) { - ma_uint32 newRefCount = ma_atomic_decrement_32(&pThis->counter); - if (newRefCount == 0) { - return 0; /* We don't free anything here because we never allocate the object on the heap. */ + switch (format) + { + case ma_format_u8: ma_copy_and_apply_volume_factor_pcm_frames_u8 ((ma_uint8*)pPCMFramesOut, (const ma_uint8*)pPCMFramesIn, frameCount, channels, factor); return; + case ma_format_s16: ma_copy_and_apply_volume_factor_pcm_frames_s16((ma_int16*)pPCMFramesOut, (const ma_int16*)pPCMFramesIn, frameCount, channels, factor); return; + case ma_format_s24: ma_copy_and_apply_volume_factor_pcm_frames_s24( pPCMFramesOut, pPCMFramesIn, frameCount, channels, factor); return; + case ma_format_s32: ma_copy_and_apply_volume_factor_pcm_frames_s32((ma_int32*)pPCMFramesOut, (const ma_int32*)pPCMFramesIn, frameCount, channels, factor); return; + case ma_format_f32: ma_copy_and_apply_volume_factor_pcm_frames_f32( (float*)pPCMFramesOut, (const float*)pPCMFramesIn, frameCount, channels, factor); return; + default: return; /* Do nothing. */ } +} - return (ULONG)newRefCount; +void ma_apply_volume_factor_pcm_frames_u8(ma_uint8* pPCMFrames, ma_uint32 frameCount, ma_uint32 channels, float factor) +{ + ma_copy_and_apply_volume_factor_pcm_frames_u8(pPCMFrames, pPCMFrames, frameCount, channels, factor); } +void ma_apply_volume_factor_pcm_frames_s16(ma_int16* pPCMFrames, ma_uint32 frameCount, ma_uint32 channels, float factor) +{ + ma_copy_and_apply_volume_factor_pcm_frames_s16(pPCMFrames, pPCMFrames, frameCount, channels, factor); +} -HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDeviceStateChanged(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID, DWORD dwNewState) +void ma_apply_volume_factor_pcm_frames_s24(void* pPCMFrames, ma_uint32 frameCount, ma_uint32 channels, float factor) { -#ifdef MA_DEBUG_OUTPUT - printf("IMMNotificationClient_OnDeviceStateChanged(pDeviceID=%S, dwNewState=%u)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)", (unsigned int)dwNewState); -#endif + ma_copy_and_apply_volume_factor_pcm_frames_s24(pPCMFrames, pPCMFrames, frameCount, channels, factor); +} - (void)pThis; - (void)pDeviceID; - (void)dwNewState; - return S_OK; +void ma_apply_volume_factor_pcm_frames_s32(ma_int32* pPCMFrames, ma_uint32 frameCount, ma_uint32 channels, float factor) +{ + ma_copy_and_apply_volume_factor_pcm_frames_s32(pPCMFrames, pPCMFrames, frameCount, channels, factor); } -HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDeviceAdded(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID) +void ma_apply_volume_factor_pcm_frames_f32(float* pPCMFrames, ma_uint32 frameCount, ma_uint32 channels, float factor) { -#ifdef MA_DEBUG_OUTPUT - printf("IMMNotificationClient_OnDeviceAdded(pDeviceID=%S)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)"); -#endif + ma_copy_and_apply_volume_factor_pcm_frames_f32(pPCMFrames, pPCMFrames, frameCount, channels, factor); +} - /* We don't need to worry about this event for our purposes. */ - (void)pThis; - (void)pDeviceID; - return S_OK; +void ma_apply_volume_factor_pcm_frames(void* pPCMFrames, ma_uint32 frameCount, ma_format format, ma_uint32 channels, float factor) +{ + ma_copy_and_apply_volume_factor_pcm_frames(pPCMFrames, pPCMFrames, frameCount, format, channels, factor); } -HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDeviceRemoved(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID) + +float ma_factor_to_gain_db(float factor) { -#ifdef MA_DEBUG_OUTPUT - printf("IMMNotificationClient_OnDeviceRemoved(pDeviceID=%S)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)"); -#endif + return (float)(20*ma_log10f(factor)); +} - /* We don't need to worry about this event for our purposes. */ - (void)pThis; - (void)pDeviceID; - return S_OK; +float ma_gain_db_to_factor(float gain) +{ + return (float)ma_powf(10, gain/20.0f); } -HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDefaultDeviceChanged(ma_IMMNotificationClient* pThis, ma_EDataFlow dataFlow, ma_ERole role, LPCWSTR pDefaultDeviceID) + +static void ma_device__on_data(ma_device* pDevice, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount) { -#ifdef MA_DEBUG_OUTPUT - printf("IMMNotificationClient_OnDefaultDeviceChanged(dataFlow=%d, role=%d, pDefaultDeviceID=%S)\n", dataFlow, role, (pDefaultDeviceID != NULL) ? pDefaultDeviceID : L"(NULL)"); -#endif + float masterVolumeFactor; + + masterVolumeFactor = pDevice->masterVolumeFactor; - /* We only ever use the eConsole role in miniaudio. */ - if (role != ma_eConsole) { - return S_OK; - } + if (pDevice->onData) { + if (!pDevice->noPreZeroedOutputBuffer && pFramesOut != NULL) { + ma_zero_pcm_frames(pFramesOut, frameCount, pDevice->playback.format, pDevice->playback.channels); + } - /* We only care about devices with the same data flow and role as the current device. */ - if ((pThis->pDevice->type == ma_device_type_playback && dataFlow != ma_eRender) || - (pThis->pDevice->type == ma_device_type_capture && dataFlow != ma_eCapture)) { - return S_OK; - } + /* Volume control of input makes things a bit awkward because the input buffer is read-only. We'll need to use a temp buffer and loop in this case. */ + if (pFramesIn != NULL && masterVolumeFactor < 1) { + ma_uint8 tempFramesIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 bpfCapture = ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); + ma_uint32 bpfPlayback = ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); + ma_uint32 totalFramesProcessed = 0; + while (totalFramesProcessed < frameCount) { + ma_uint32 framesToProcessThisIteration = frameCount - totalFramesProcessed; + if (framesToProcessThisIteration > sizeof(tempFramesIn)/bpfCapture) { + framesToProcessThisIteration = sizeof(tempFramesIn)/bpfCapture; + } - /* - Not currently supporting automatic stream routing in exclusive mode. This is not working correctly on my machine due to - AUDCLNT_E_DEVICE_IN_USE errors when reinitializing the device. If this is a bug in miniaudio, we can try re-enabling this once - it's fixed. - */ - if ((dataFlow == ma_eRender && pThis->pDevice->playback.shareMode == ma_share_mode_exclusive) || - (dataFlow == ma_eCapture && pThis->pDevice->capture.shareMode == ma_share_mode_exclusive)) { - return S_OK; - } + ma_copy_and_apply_volume_factor_pcm_frames(tempFramesIn, ma_offset_ptr(pFramesIn, totalFramesProcessed*bpfCapture), framesToProcessThisIteration, pDevice->capture.format, pDevice->capture.channels, masterVolumeFactor); - /* - We don't change the device here - we change it in the worker thread to keep synchronization simple. To do this I'm just setting a flag to - indicate that the default device has changed. Loopback devices are treated as capture devices so we need to do a bit of a dance to handle - that properly. - */ - if (dataFlow == ma_eRender && pThis->pDevice->type != ma_device_type_loopback) { - ma_atomic_exchange_32(&pThis->pDevice->wasapi.hasDefaultPlaybackDeviceChanged, MA_TRUE); - } - if (dataFlow == ma_eCapture || pThis->pDevice->type == ma_device_type_loopback) { - ma_atomic_exchange_32(&pThis->pDevice->wasapi.hasDefaultCaptureDeviceChanged, MA_TRUE); - } + pDevice->onData(pDevice, ma_offset_ptr(pFramesOut, totalFramesProcessed*bpfPlayback), tempFramesIn, framesToProcessThisIteration); - (void)pDefaultDeviceID; - return S_OK; + totalFramesProcessed += framesToProcessThisIteration; + } + } else { + pDevice->onData(pDevice, pFramesOut, pFramesIn, frameCount); + } + + /* Volume control and clipping for playback devices. */ + if (pFramesOut != NULL) { + if (masterVolumeFactor < 1) { + if (pFramesIn == NULL) { /* <-- In full-duplex situations, the volume will have been applied to the input samples before the data callback. Applying it again post-callback will incorrectly compound it. */ + ma_apply_volume_factor_pcm_frames(pFramesOut, frameCount, pDevice->playback.format, pDevice->playback.channels, masterVolumeFactor); + } + } + + if (!pDevice->noClip && pDevice->playback.format == ma_format_f32) { + ma_clip_pcm_frames_f32((float*)pFramesOut, frameCount, pDevice->playback.channels); + } + } + } } -HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnPropertyValueChanged(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID, const PROPERTYKEY key) + + +/* A helper function for reading sample data from the client. */ +static void ma_device__read_frames_from_client(ma_device* pDevice, ma_uint32 frameCount, void* pFramesOut) { -#ifdef MA_DEBUG_OUTPUT - printf("IMMNotificationClient_OnPropertyValueChanged(pDeviceID=%S)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)"); -#endif + MA_ASSERT(pDevice != NULL); + MA_ASSERT(frameCount > 0); + MA_ASSERT(pFramesOut != NULL); - (void)pThis; - (void)pDeviceID; - (void)key; - return S_OK; -} + if (pDevice->playback.converter.isPassthrough) { + ma_device__on_data(pDevice, pFramesOut, NULL, frameCount); + } else { + ma_result result; + ma_uint64 totalFramesReadOut; + ma_uint64 totalFramesReadIn; + void* pRunningFramesOut; -static ma_IMMNotificationClientVtbl g_maNotificationCientVtbl = { - ma_IMMNotificationClient_QueryInterface, - ma_IMMNotificationClient_AddRef, - ma_IMMNotificationClient_Release, - ma_IMMNotificationClient_OnDeviceStateChanged, - ma_IMMNotificationClient_OnDeviceAdded, - ma_IMMNotificationClient_OnDeviceRemoved, - ma_IMMNotificationClient_OnDefaultDeviceChanged, - ma_IMMNotificationClient_OnPropertyValueChanged -}; -#endif /* MA_WIN32_DESKTOP */ + totalFramesReadOut = 0; + totalFramesReadIn = 0; + pRunningFramesOut = pFramesOut; + + while (totalFramesReadOut < frameCount) { + ma_uint8 pIntermediaryBuffer[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In client format. */ + ma_uint64 intermediaryBufferCap = sizeof(pIntermediaryBuffer) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); + ma_uint64 framesToReadThisIterationIn; + ma_uint64 framesReadThisIterationIn; + ma_uint64 framesToReadThisIterationOut; + ma_uint64 framesReadThisIterationOut; + ma_uint64 requiredInputFrameCount; + + framesToReadThisIterationOut = (frameCount - totalFramesReadOut); + framesToReadThisIterationIn = framesToReadThisIterationOut; + if (framesToReadThisIterationIn > intermediaryBufferCap) { + framesToReadThisIterationIn = intermediaryBufferCap; + } -#ifdef MA_WIN32_DESKTOP -typedef ma_IMMDevice ma_WASAPIDeviceInterface; -#else -typedef ma_IUnknown ma_WASAPIDeviceInterface; -#endif + requiredInputFrameCount = ma_data_converter_get_required_input_frame_count(&pDevice->playback.converter, frameCount); + if (framesToReadThisIterationIn > requiredInputFrameCount) { + framesToReadThisIterationIn = requiredInputFrameCount; + } + if (framesToReadThisIterationIn > 0) { + ma_device__on_data(pDevice, pIntermediaryBuffer, NULL, (ma_uint32)framesToReadThisIterationIn); + totalFramesReadIn += framesToReadThisIterationIn; + } + /* + At this point we have our decoded data in input format and now we need to convert to output format. Note that even if we didn't read any + input frames, we still want to try processing frames because there may some output frames generated from cached input data. + */ + framesReadThisIterationIn = framesToReadThisIterationIn; + framesReadThisIterationOut = framesToReadThisIterationOut; + result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, pIntermediaryBuffer, &framesReadThisIterationIn, pRunningFramesOut, &framesReadThisIterationOut); + if (result != MA_SUCCESS) { + break; + } -ma_bool32 ma_context_is_device_id_equal__wasapi(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) -{ - ma_assert(pContext != NULL); - ma_assert(pID0 != NULL); - ma_assert(pID1 != NULL); - (void)pContext; + totalFramesReadOut += framesReadThisIterationOut; + pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesReadThisIterationOut * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); - return memcmp(pID0->wasapi, pID1->wasapi, sizeof(pID0->wasapi)) == 0; + if (framesReadThisIterationIn == 0 && framesReadThisIterationOut == 0) { + break; /* We're done. */ + } + } + } } -void ma_set_device_info_from_WAVEFORMATEX(const WAVEFORMATEX* pWF, ma_device_info* pInfo) +/* A helper for sending sample data to the client. */ +static void ma_device__send_frames_to_client(ma_device* pDevice, ma_uint32 frameCountInDeviceFormat, const void* pFramesInDeviceFormat) { - ma_assert(pWF != NULL); - ma_assert(pInfo != NULL); + MA_ASSERT(pDevice != NULL); + MA_ASSERT(frameCountInDeviceFormat > 0); + MA_ASSERT(pFramesInDeviceFormat != NULL); - pInfo->formatCount = 1; - pInfo->formats[0] = ma_format_from_WAVEFORMATEX(pWF); - pInfo->minChannels = pWF->nChannels; - pInfo->maxChannels = pWF->nChannels; - pInfo->minSampleRate = pWF->nSamplesPerSec; - pInfo->maxSampleRate = pWF->nSamplesPerSec; + if (pDevice->capture.converter.isPassthrough) { + ma_device__on_data(pDevice, NULL, pFramesInDeviceFormat, frameCountInDeviceFormat); + } else { + ma_result result; + ma_uint8 pFramesInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint64 framesInClientFormatCap = sizeof(pFramesInClientFormat) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); + ma_uint64 totalDeviceFramesProcessed = 0; + ma_uint64 totalClientFramesProcessed = 0; + const void* pRunningFramesInDeviceFormat = pFramesInDeviceFormat; + + /* We just keep going until we've exhaused all of our input frames and cannot generate any more output frames. */ + for (;;) { + ma_uint64 deviceFramesProcessedThisIteration; + ma_uint64 clientFramesProcessedThisIteration; + + deviceFramesProcessedThisIteration = (frameCountInDeviceFormat - totalDeviceFramesProcessed); + clientFramesProcessedThisIteration = framesInClientFormatCap; + + result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningFramesInDeviceFormat, &deviceFramesProcessedThisIteration, pFramesInClientFormat, &clientFramesProcessedThisIteration); + if (result != MA_SUCCESS) { + break; + } + + if (clientFramesProcessedThisIteration > 0) { + ma_device__on_data(pDevice, NULL, pFramesInClientFormat, (ma_uint32)clientFramesProcessedThisIteration); /* Safe cast. */ + } + + pRunningFramesInDeviceFormat = ma_offset_ptr(pRunningFramesInDeviceFormat, deviceFramesProcessedThisIteration * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); + totalDeviceFramesProcessed += deviceFramesProcessedThisIteration; + totalClientFramesProcessed += clientFramesProcessedThisIteration; + + if (deviceFramesProcessedThisIteration == 0 && clientFramesProcessedThisIteration == 0) { + break; /* We're done. */ + } + } + } } -ma_result ma_context_get_device_info_from_IAudioClient__wasapi(ma_context* pContext, /*ma_IMMDevice**/void* pMMDevice, ma_IAudioClient* pAudioClient, ma_share_mode shareMode, ma_device_info* pInfo) + +/* We only want to expose ma_device__handle_duplex_callback_capture() and ma_device__handle_duplex_callback_playback() if we have an asynchronous backend enabled. */ +#if defined(MA_HAS_JACK) || \ + defined(MA_HAS_COREAUDIO) || \ + defined(MA_HAS_AAUDIO) || \ + defined(MA_HAS_OPENSL) || \ + defined(MA_HAS_WEBAUDIO) +static ma_result ma_device__handle_duplex_callback_capture(ma_device* pDevice, ma_uint32 frameCountInDeviceFormat, const void* pFramesInDeviceFormat, ma_pcm_rb* pRB) { - ma_assert(pAudioClient != NULL); - ma_assert(pInfo != NULL); + ma_result result; + ma_uint32 totalDeviceFramesProcessed = 0; + const void* pRunningFramesInDeviceFormat = pFramesInDeviceFormat; - /* We use a different technique to retrieve the device information depending on whether or not we are using shared or exclusive mode. */ - if (shareMode == ma_share_mode_shared) { - /* Shared Mode. We use GetMixFormat() here. */ - WAVEFORMATEX* pWF = NULL; - HRESULT hr = ma_IAudioClient_GetMixFormat((ma_IAudioClient*)pAudioClient, (WAVEFORMATEX**)&pWF); - if (SUCCEEDED(hr)) { - ma_set_device_info_from_WAVEFORMATEX(pWF, pInfo); - return MA_SUCCESS; - } else { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve mix format for device info retrieval.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + MA_ASSERT(pDevice != NULL); + MA_ASSERT(frameCountInDeviceFormat > 0); + MA_ASSERT(pFramesInDeviceFormat != NULL); + MA_ASSERT(pRB != NULL); + + /* Write to the ring buffer. The ring buffer is in the client format which means we need to convert. */ + for (;;) { + ma_uint32 framesToProcessInDeviceFormat = (frameCountInDeviceFormat - totalDeviceFramesProcessed); + ma_uint32 framesToProcessInClientFormat = MA_DATA_CONVERTER_STACK_BUFFER_SIZE / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); + ma_uint64 framesProcessedInDeviceFormat; + ma_uint64 framesProcessedInClientFormat; + void* pFramesInClientFormat; + + result = ma_pcm_rb_acquire_write(pRB, &framesToProcessInClientFormat, &pFramesInClientFormat); + if (result != MA_SUCCESS) { + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "Failed to acquire capture PCM frames from ring buffer.", result); + break; } - } else { - /* Exlcusive Mode. We repeatedly call IsFormatSupported() here. This is not currently support on UWP. */ -#ifdef MA_WIN32_DESKTOP - /* - The first thing to do is get the format from PKEY_AudioEngine_DeviceFormat. This should give us a channel count we assume is - correct which will simplify our searching. - */ - ma_IPropertyStore *pProperties; - HRESULT hr = ma_IMMDevice_OpenPropertyStore((ma_IMMDevice*)pMMDevice, STGM_READ, &pProperties); - if (SUCCEEDED(hr)) { - PROPVARIANT var; - ma_PropVariantInit(&var); - hr = ma_IPropertyStore_GetValue(pProperties, &MA_PKEY_AudioEngine_DeviceFormat, &var); - if (SUCCEEDED(hr)) { - WAVEFORMATEX* pWF = (WAVEFORMATEX*)var.blob.pBlobData; - ma_set_device_info_from_WAVEFORMATEX(pWF, pInfo); + if (framesToProcessInClientFormat == 0) { + if (ma_pcm_rb_pointer_disance(pRB) == (ma_int32)ma_pcm_rb_get_subbuffer_size(pRB)) { + break; /* Overrun. Not enough room in the ring buffer for input frame. Excess frames are dropped. */ + } + } - /* - In my testing, the format returned by PKEY_AudioEngine_DeviceFormat is suitable for exclusive mode so we check this format - first. If this fails, fall back to a search. - */ - hr = ma_IAudioClient_IsFormatSupported((ma_IAudioClient*)pAudioClient, MA_AUDCLNT_SHAREMODE_EXCLUSIVE, pWF, NULL); - ma_PropVariantClear(pContext, &var); + /* Convert. */ + framesProcessedInDeviceFormat = framesToProcessInDeviceFormat; + framesProcessedInClientFormat = framesToProcessInClientFormat; + result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningFramesInDeviceFormat, &framesProcessedInDeviceFormat, pFramesInClientFormat, &framesProcessedInClientFormat); + if (result != MA_SUCCESS) { + break; + } - if (FAILED(hr)) { - /* - The format returned by PKEY_AudioEngine_DeviceFormat is not supported, so fall back to a search. We assume the channel - count returned by MA_PKEY_AudioEngine_DeviceFormat is valid and correct. For simplicity we're only returning one format. - */ - ma_uint32 channels = pInfo->minChannels; - ma_format formatsToSearch[] = { - ma_format_s16, - ma_format_s24, - /*ma_format_s24_32,*/ - ma_format_f32, - ma_format_s32, - ma_format_u8 - }; - ma_channel defaultChannelMap[MA_MAX_CHANNELS]; - WAVEFORMATEXTENSIBLE wf; - ma_bool32 found; - ma_uint32 iFormat; + result = ma_pcm_rb_commit_write(pRB, (ma_uint32)framesProcessedInDeviceFormat, pFramesInClientFormat); /* Safe cast. */ + if (result != MA_SUCCESS) { + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "Failed to commit capture PCM frames to ring buffer.", result); + break; + } - ma_get_standard_channel_map(ma_standard_channel_map_microsoft, channels, defaultChannelMap); + pRunningFramesInDeviceFormat = ma_offset_ptr(pRunningFramesInDeviceFormat, framesProcessedInDeviceFormat * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); + totalDeviceFramesProcessed += (ma_uint32)framesProcessedInDeviceFormat; /* Safe cast. */ - ma_zero_object(&wf); - wf.Format.cbSize = sizeof(wf); - wf.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE; - wf.Format.nChannels = (WORD)channels; - wf.dwChannelMask = ma_channel_map_to_channel_mask__win32(defaultChannelMap, channels); + /* We're done when we're unable to process any client nor device frames. */ + if (framesProcessedInClientFormat == 0 && framesProcessedInDeviceFormat == 0) { + break; /* Done. */ + } + } - found = MA_FALSE; - for (iFormat = 0; iFormat < ma_countof(formatsToSearch); ++iFormat) { - ma_format format = formatsToSearch[iFormat]; - ma_uint32 iSampleRate; + return MA_SUCCESS; +} - wf.Format.wBitsPerSample = (WORD)ma_get_bytes_per_sample(format)*8; - wf.Format.nBlockAlign = (wf.Format.nChannels * wf.Format.wBitsPerSample) / 8; - wf.Format.nAvgBytesPerSec = wf.Format.nBlockAlign * wf.Format.nSamplesPerSec; - wf.Samples.wValidBitsPerSample = /*(format == ma_format_s24_32) ? 24 :*/ wf.Format.wBitsPerSample; - if (format == ma_format_f32) { - wf.SubFormat = MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT; - } else { - wf.SubFormat = MA_GUID_KSDATAFORMAT_SUBTYPE_PCM; - } +static ma_result ma_device__handle_duplex_callback_playback(ma_device* pDevice, ma_uint32 frameCount, void* pFramesInInternalFormat, ma_pcm_rb* pRB) +{ + ma_result result; + ma_uint8 playbackFramesInExternalFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint8 silentInputFrames[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 totalFramesToReadFromClient; + ma_uint32 totalFramesReadFromClient; + ma_uint32 totalFramesReadOut = 0; - for (iSampleRate = 0; iSampleRate < ma_countof(g_maStandardSampleRatePriorities); ++iSampleRate) { - wf.Format.nSamplesPerSec = g_maStandardSampleRatePriorities[iSampleRate]; + MA_ASSERT(pDevice != NULL); + MA_ASSERT(frameCount > 0); + MA_ASSERT(pFramesInInternalFormat != NULL); + MA_ASSERT(pRB != NULL); + + /* + Sitting in the ring buffer should be captured data from the capture callback in external format. If there's not enough data in there for + the whole frameCount frames we just use silence instead for the input data. + */ + MA_ZERO_MEMORY(silentInputFrames, sizeof(silentInputFrames)); - hr = ma_IAudioClient_IsFormatSupported((ma_IAudioClient*)pAudioClient, MA_AUDCLNT_SHAREMODE_EXCLUSIVE, (WAVEFORMATEX*)&wf, NULL); - if (SUCCEEDED(hr)) { - ma_set_device_info_from_WAVEFORMATEX((WAVEFORMATEX*)&wf, pInfo); - found = MA_TRUE; - break; - } - } + /* We need to calculate how many output frames are required to be read from the client to completely fill frameCount internal frames. */ + totalFramesToReadFromClient = (ma_uint32)ma_data_converter_get_required_input_frame_count(&pDevice->playback.converter, frameCount); + totalFramesReadFromClient = 0; + while (totalFramesReadFromClient < totalFramesToReadFromClient && ma_device_is_started(pDevice)) { + ma_uint32 framesRemainingFromClient; + ma_uint32 framesToProcessFromClient; + ma_uint32 inputFrameCount; + void* pInputFrames; - if (found) { - break; - } - } + framesRemainingFromClient = (totalFramesToReadFromClient - totalFramesReadFromClient); + framesToProcessFromClient = sizeof(playbackFramesInExternalFormat) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); + if (framesToProcessFromClient > framesRemainingFromClient) { + framesToProcessFromClient = framesRemainingFromClient; + } - if (!found) { - ma_IPropertyStore_Release(pProperties); - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to find suitable device format for device info retrieval.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } - } + /* We need to grab captured samples before firing the callback. If there's not enough input samples we just pass silence. */ + inputFrameCount = framesToProcessFromClient; + result = ma_pcm_rb_acquire_read(pRB, &inputFrameCount, &pInputFrames); + if (result == MA_SUCCESS) { + if (inputFrameCount > 0) { + /* Use actual input frames. */ + ma_device__on_data(pDevice, playbackFramesInExternalFormat, pInputFrames, inputFrameCount); } else { - ma_IPropertyStore_Release(pProperties); - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve device format for device info retrieval.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + if (ma_pcm_rb_pointer_disance(pRB) == 0) { + break; /* Underrun. */ + } + } + + /* We're done with the captured samples. */ + result = ma_pcm_rb_commit_read(pRB, inputFrameCount, pInputFrames); + if (result != MA_SUCCESS) { + break; /* Don't know what to do here... Just abandon ship. */ } - - ma_IPropertyStore_Release(pProperties); } else { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to open property store for device info retrieval.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + /* Use silent input frames. */ + inputFrameCount = ma_min( + sizeof(playbackFramesInExternalFormat) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels), + sizeof(silentInputFrames) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels) + ); + + ma_device__on_data(pDevice, playbackFramesInExternalFormat, silentInputFrames, inputFrameCount); } - return MA_SUCCESS; -#else - /* Exclusive mode not fully supported in UWP right now. */ - return MA_ERROR; + /* We have samples in external format so now we need to convert to internal format and output to the device. */ + { + ma_uint64 framesConvertedIn = inputFrameCount; + ma_uint64 framesConvertedOut = (frameCount - totalFramesReadOut); + ma_data_converter_process_pcm_frames(&pDevice->playback.converter, playbackFramesInExternalFormat, &framesConvertedIn, pFramesInInternalFormat, &framesConvertedOut); + + totalFramesReadFromClient += (ma_uint32)framesConvertedIn; /* Safe cast. */ + totalFramesReadOut += (ma_uint32)framesConvertedOut; /* Safe cast. */ + pFramesInInternalFormat = ma_offset_ptr(pFramesInInternalFormat, framesConvertedOut * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); + } + } + + return MA_SUCCESS; +} +#endif /* Asynchronous backends. */ + +/* A helper for changing the state of the device. */ +static MA_INLINE void ma_device__set_state(ma_device* pDevice, ma_uint32 newState) +{ + ma_atomic_exchange_32(&pDevice->state, newState); +} + +/* A helper for getting the state of the device. */ +static MA_INLINE ma_uint32 ma_device__get_state(ma_device* pDevice) +{ + return pDevice->state; +} + + +#ifdef MA_WIN32 + GUID MA_GUID_KSDATAFORMAT_SUBTYPE_PCM = {0x00000001, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}}; + GUID MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = {0x00000003, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}}; + /*GUID MA_GUID_KSDATAFORMAT_SUBTYPE_ALAW = {0x00000006, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}};*/ + /*GUID MA_GUID_KSDATAFORMAT_SUBTYPE_MULAW = {0x00000007, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}};*/ #endif + + +typedef struct +{ + ma_device_type deviceType; + const ma_device_id* pDeviceID; + char* pName; + size_t nameBufferSize; + ma_bool32 foundDevice; +} ma_context__try_get_device_name_by_id__enum_callback_data; + +static ma_bool32 ma_context__try_get_device_name_by_id__enum_callback(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pDeviceInfo, void* pUserData) +{ + ma_context__try_get_device_name_by_id__enum_callback_data* pData = (ma_context__try_get_device_name_by_id__enum_callback_data*)pUserData; + MA_ASSERT(pData != NULL); + + if (pData->deviceType == deviceType) { + if (pContext->onDeviceIDEqual(pContext, pData->pDeviceID, &pDeviceInfo->id)) { + ma_strncpy_s(pData->pName, pData->nameBufferSize, pDeviceInfo->name, (size_t)-1); + pData->foundDevice = MA_TRUE; + } } + + return !pData->foundDevice; } -#ifdef MA_WIN32_DESKTOP -ma_result ma_context_get_MMDevice__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_IMMDevice** ppMMDevice) +/* +Generic function for retrieving the name of a device by it's ID. + +This function simply enumerates every device and then retrieves the name of the first device that has the same ID. +*/ +static ma_result ma_context__try_get_device_name_by_id(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, char* pName, size_t nameBufferSize) { - ma_IMMDeviceEnumerator* pDeviceEnumerator; - HRESULT hr; - - ma_assert(pContext != NULL); - ma_assert(ppMMDevice != NULL); + ma_result result; + ma_context__try_get_device_name_by_id__enum_callback_data data; - hr = ma_CoCreateInstance(pContext, MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator); - if (FAILED(hr)) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create IMMDeviceEnumerator.", MA_FAILED_TO_INIT_BACKEND); - } + MA_ASSERT(pContext != NULL); + MA_ASSERT(pName != NULL); if (pDeviceID == NULL) { - hr = ma_IMMDeviceEnumerator_GetDefaultAudioEndpoint(pDeviceEnumerator, (deviceType == ma_device_type_capture) ? ma_eCapture : ma_eRender, ma_eConsole, ppMMDevice); - } else { - hr = ma_IMMDeviceEnumerator_GetDevice(pDeviceEnumerator, pDeviceID->wasapi, ppMMDevice); + return MA_NO_DEVICE; } - ma_IMMDeviceEnumerator_Release(pDeviceEnumerator); - if (FAILED(hr)) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve IMMDevice.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + data.deviceType = deviceType; + data.pDeviceID = pDeviceID; + data.pName = pName; + data.nameBufferSize = nameBufferSize; + data.foundDevice = MA_FALSE; + result = ma_context_enumerate_devices(pContext, ma_context__try_get_device_name_by_id__enum_callback, &data); + if (result != MA_SUCCESS) { + return result; } - return MA_SUCCESS; + if (!data.foundDevice) { + return MA_NO_DEVICE; + } else { + return MA_SUCCESS; + } } -ma_result ma_context_get_device_info_from_MMDevice__wasapi(ma_context* pContext, ma_IMMDevice* pMMDevice, ma_share_mode shareMode, ma_bool32 onlySimpleInfo, ma_device_info* pInfo) -{ - LPWSTR id; - HRESULT hr; - - ma_assert(pContext != NULL); - ma_assert(pMMDevice != NULL); - ma_assert(pInfo != NULL); - /* ID. */ - hr = ma_IMMDevice_GetId(pMMDevice, &id); - if (SUCCEEDED(hr)) { - size_t idlen = wcslen(id); - if (idlen+1 > ma_countof(pInfo->id.wasapi)) { - ma_CoTaskMemFree(pContext, id); - ma_assert(MA_FALSE); /* NOTE: If this is triggered, please report it. It means the format of the ID must haved change and is too long to fit in our fixed sized buffer. */ - return MA_ERROR; +ma_uint32 ma_get_format_priority_index(ma_format format) /* Lower = better. */ +{ + ma_uint32 i; + for (i = 0; i < ma_countof(g_maFormatPriorities); ++i) { + if (g_maFormatPriorities[i] == format) { + return i; } + } - ma_copy_memory(pInfo->id.wasapi, id, idlen * sizeof(wchar_t)); - pInfo->id.wasapi[idlen] = '\0'; + /* Getting here means the format could not be found or is equal to ma_format_unknown. */ + return (ma_uint32)-1; +} - ma_CoTaskMemFree(pContext, id); - } +static ma_result ma_device__post_init_setup(ma_device* pDevice, ma_device_type deviceType); - { - ma_IPropertyStore *pProperties; - hr = ma_IMMDevice_OpenPropertyStore(pMMDevice, STGM_READ, &pProperties); - if (SUCCEEDED(hr)) { - PROPVARIANT var; - /* Description / Friendly Name */ - ma_PropVariantInit(&var); - hr = ma_IPropertyStore_GetValue(pProperties, &MA_PKEY_Device_FriendlyName, &var); - if (SUCCEEDED(hr)) { - WideCharToMultiByte(CP_UTF8, 0, var.pwszVal, -1, pInfo->name, sizeof(pInfo->name), 0, FALSE); - ma_PropVariantClear(pContext, &var); - } +/******************************************************************************* - ma_IPropertyStore_Release(pProperties); - } - } +Null Backend - /* Format */ - if (!onlySimpleInfo) { - ma_IAudioClient* pAudioClient; - hr = ma_IMMDevice_Activate(pMMDevice, &MA_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pAudioClient); - if (SUCCEEDED(hr)) { - ma_result result = ma_context_get_device_info_from_IAudioClient__wasapi(pContext, pMMDevice, pAudioClient, shareMode, pInfo); - - ma_IAudioClient_Release(pAudioClient); - return result; - } else { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to activate audio client for device info retrieval.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } - } +*******************************************************************************/ +#ifdef MA_HAS_NULL - return MA_SUCCESS; -} +#define MA_DEVICE_OP_NONE__NULL 0 +#define MA_DEVICE_OP_START__NULL 1 +#define MA_DEVICE_OP_SUSPEND__NULL 2 +#define MA_DEVICE_OP_KILL__NULL 3 -ma_result ma_context_enumerate_device_collection__wasapi(ma_context* pContext, ma_IMMDeviceCollection* pDeviceCollection, ma_device_type deviceType, ma_enum_devices_callback_proc callback, void* pUserData) +static ma_thread_result MA_THREADCALL ma_device_thread__null(void* pData) { - UINT deviceCount; - HRESULT hr; - ma_uint32 iDevice; + ma_device* pDevice = (ma_device*)pData; + MA_ASSERT(pDevice != NULL); - ma_assert(pContext != NULL); - ma_assert(callback != NULL); + for (;;) { /* Keep the thread alive until the device is uninitialized. */ + /* Wait for an operation to be requested. */ + ma_event_wait(&pDevice->null_device.operationEvent); - hr = ma_IMMDeviceCollection_GetCount(pDeviceCollection, &deviceCount); - if (FAILED(hr)) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to get playback device count.", MA_NO_DEVICE); - } + /* At this point an event should have been triggered. */ - for (iDevice = 0; iDevice < deviceCount; ++iDevice) { - ma_device_info deviceInfo; - ma_IMMDevice* pMMDevice; - - ma_zero_object(&deviceInfo); + /* Starting the device needs to put the thread into a loop. */ + if (pDevice->null_device.operation == MA_DEVICE_OP_START__NULL) { + ma_atomic_exchange_32(&pDevice->null_device.operation, MA_DEVICE_OP_NONE__NULL); - hr = ma_IMMDeviceCollection_Item(pDeviceCollection, iDevice, &pMMDevice); - if (SUCCEEDED(hr)) { - ma_result result = ma_context_get_device_info_from_MMDevice__wasapi(pContext, pMMDevice, ma_share_mode_shared, MA_TRUE, &deviceInfo); /* MA_TRUE = onlySimpleInfo. */ + /* Reset the timer just in case. */ + ma_timer_init(&pDevice->null_device.timer); - ma_IMMDevice_Release(pMMDevice); - if (result == MA_SUCCESS) { - ma_bool32 cbResult = callback(pContext, deviceType, &deviceInfo, pUserData); - if (cbResult == MA_FALSE) { - break; - } + /* Keep looping until an operation has been requested. */ + while (pDevice->null_device.operation != MA_DEVICE_OP_NONE__NULL && pDevice->null_device.operation != MA_DEVICE_OP_START__NULL) { + ma_sleep(10); /* Don't hog the CPU. */ } + + /* Getting here means a suspend or kill operation has been requested. */ + ma_atomic_exchange_32(&pDevice->null_device.operationResult, MA_SUCCESS); + ma_event_signal(&pDevice->null_device.operationCompletionEvent); + continue; + } + + /* Suspending the device means we need to stop the timer and just continue the loop. */ + if (pDevice->null_device.operation == MA_DEVICE_OP_SUSPEND__NULL) { + ma_atomic_exchange_32(&pDevice->null_device.operation, MA_DEVICE_OP_NONE__NULL); + + /* We need to add the current run time to the prior run time, then reset the timer. */ + pDevice->null_device.priorRunTime += ma_timer_get_time_in_seconds(&pDevice->null_device.timer); + ma_timer_init(&pDevice->null_device.timer); + + /* We're done. */ + ma_atomic_exchange_32(&pDevice->null_device.operationResult, MA_SUCCESS); + ma_event_signal(&pDevice->null_device.operationCompletionEvent); + continue; + } + + /* Killing the device means we need to get out of this loop so that this thread can terminate. */ + if (pDevice->null_device.operation == MA_DEVICE_OP_KILL__NULL) { + ma_atomic_exchange_32(&pDevice->null_device.operation, MA_DEVICE_OP_NONE__NULL); + ma_atomic_exchange_32(&pDevice->null_device.operationResult, MA_SUCCESS); + ma_event_signal(&pDevice->null_device.operationCompletionEvent); + break; + } + + /* Getting a signal on a "none" operation probably means an error. Return invalid operation. */ + if (pDevice->null_device.operation == MA_DEVICE_OP_NONE__NULL) { + MA_ASSERT(MA_FALSE); /* <-- Trigger this in debug mode to ensure developers are aware they're doing something wrong (or there's a bug in a miniaudio). */ + ma_atomic_exchange_32(&pDevice->null_device.operationResult, MA_INVALID_OPERATION); + ma_event_signal(&pDevice->null_device.operationCompletionEvent); + continue; /* Continue the loop. Don't terminate. */ } } - return MA_SUCCESS; + return (ma_thread_result)0; } -#endif -#ifdef MA_WIN32_DESKTOP -ma_result ma_context_get_IAudioClient_Desktop__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_IAudioClient** ppAudioClient, ma_IMMDevice** ppMMDevice) +static ma_result ma_device_do_operation__null(ma_device* pDevice, ma_uint32 operation) { - ma_result result; - HRESULT hr; - - ma_assert(pContext != NULL); - ma_assert(ppAudioClient != NULL); - ma_assert(ppMMDevice != NULL); + ma_atomic_exchange_32(&pDevice->null_device.operation, operation); + if (!ma_event_signal(&pDevice->null_device.operationEvent)) { + return MA_ERROR; + } - result = ma_context_get_MMDevice__wasapi(pContext, deviceType, pDeviceID, ppMMDevice); - if (result != MA_SUCCESS) { - return result; + if (!ma_event_wait(&pDevice->null_device.operationCompletionEvent)) { + return MA_ERROR; } - hr = ma_IMMDevice_Activate(*ppMMDevice, &MA_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)ppAudioClient); - if (FAILED(hr)) { - return MA_FAILED_TO_OPEN_BACKEND_DEVICE; + return pDevice->null_device.operationResult; +} + +static ma_uint64 ma_device_get_total_run_time_in_frames__null(ma_device* pDevice) +{ + ma_uint32 internalSampleRate; + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + internalSampleRate = pDevice->capture.internalSampleRate; + } else { + internalSampleRate = pDevice->playback.internalSampleRate; } - return MA_SUCCESS; + + return (ma_uint64)((pDevice->null_device.priorRunTime + ma_timer_get_time_in_seconds(&pDevice->null_device.timer)) * internalSampleRate); } -#else -ma_result ma_context_get_IAudioClient_UWP__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_IAudioClient** ppAudioClient, ma_IUnknown** ppActivatedInterface) + +static ma_bool32 ma_context_is_device_id_equal__null(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) { - ma_IActivateAudioInterfaceAsyncOperation *pAsyncOp = NULL; - ma_completion_handler_uwp completionHandler; - IID iid; - LPOLESTR iidStr; - HRESULT hr; - ma_result result; - HRESULT activateResult; - ma_IUnknown* pActivatedInterface; + MA_ASSERT(pContext != NULL); + MA_ASSERT(pID0 != NULL); + MA_ASSERT(pID1 != NULL); + (void)pContext; - ma_assert(pContext != NULL); - ma_assert(ppAudioClient != NULL); + return pID0->nullbackend == pID1->nullbackend; +} - if (pDeviceID != NULL) { - ma_copy_memory(&iid, pDeviceID->wasapi, sizeof(iid)); - } else { - if (deviceType == ma_device_type_playback) { - iid = MA_IID_DEVINTERFACE_AUDIO_RENDER; - } else { - iid = MA_IID_DEVINTERFACE_AUDIO_CAPTURE; - } - } +static ma_result ma_context_enumerate_devices__null(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +{ + ma_bool32 cbResult = MA_TRUE; -#if defined(__cplusplus) - hr = StringFromIID(iid, &iidStr); -#else - hr = StringFromIID(&iid, &iidStr); -#endif - if (FAILED(hr)) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to convert device IID to string for ActivateAudioInterfaceAsync(). Out of memory.", MA_OUT_OF_MEMORY); - } + MA_ASSERT(pContext != NULL); + MA_ASSERT(callback != NULL); - result = ma_completion_handler_uwp_init(&completionHandler); - if (result != MA_SUCCESS) { - ma_CoTaskMemFree(pContext, iidStr); - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create event for waiting for ActivateAudioInterfaceAsync().", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + /* Playback. */ + if (cbResult) { + ma_device_info deviceInfo; + MA_ZERO_OBJECT(&deviceInfo); + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), "NULL Playback Device", (size_t)-1); + cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); } -#if defined(__cplusplus) - hr = ActivateAudioInterfaceAsync(iidStr, MA_IID_IAudioClient, NULL, (IActivateAudioInterfaceCompletionHandler*)&completionHandler, (IActivateAudioInterfaceAsyncOperation**)&pAsyncOp); -#else - hr = ActivateAudioInterfaceAsync(iidStr, &MA_IID_IAudioClient, NULL, (IActivateAudioInterfaceCompletionHandler*)&completionHandler, (IActivateAudioInterfaceAsyncOperation**)&pAsyncOp); -#endif - if (FAILED(hr)) { - ma_completion_handler_uwp_uninit(&completionHandler); - ma_CoTaskMemFree(pContext, iidStr); - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] ActivateAudioInterfaceAsync() failed.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + /* Capture. */ + if (cbResult) { + ma_device_info deviceInfo; + MA_ZERO_OBJECT(&deviceInfo); + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), "NULL Capture Device", (size_t)-1); + cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); } - ma_CoTaskMemFree(pContext, iidStr); - - /* Wait for the async operation for finish. */ - ma_completion_handler_uwp_wait(&completionHandler); - ma_completion_handler_uwp_uninit(&completionHandler); + return MA_SUCCESS; +} - hr = ma_IActivateAudioInterfaceAsyncOperation_GetActivateResult(pAsyncOp, &activateResult, &pActivatedInterface); - ma_IActivateAudioInterfaceAsyncOperation_Release(pAsyncOp); +static ma_result ma_context_get_device_info__null(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) +{ + ma_uint32 iFormat; - if (FAILED(hr) || FAILED(activateResult)) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to activate device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } + MA_ASSERT(pContext != NULL); - /* Here is where we grab the IAudioClient interface. */ - hr = ma_IUnknown_QueryInterface(pActivatedInterface, &MA_IID_IAudioClient, (void**)ppAudioClient); - if (FAILED(hr)) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to query IAudioClient interface.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + if (pDeviceID != NULL && pDeviceID->nullbackend != 0) { + return MA_NO_DEVICE; /* Don't know the device. */ } - if (ppActivatedInterface) { - *ppActivatedInterface = pActivatedInterface; + /* Name / Description */ + if (deviceType == ma_device_type_playback) { + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), "NULL Playback Device", (size_t)-1); } else { - ma_IUnknown_Release(pActivatedInterface); + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), "NULL Capture Device", (size_t)-1); + } + + /* Support everything on the null backend. */ + pDeviceInfo->formatCount = ma_format_count - 1; /* Minus one because we don't want to include ma_format_unknown. */ + for (iFormat = 0; iFormat < pDeviceInfo->formatCount; ++iFormat) { + pDeviceInfo->formats[iFormat] = (ma_format)(iFormat + 1); /* +1 to skip over ma_format_unknown. */ } + pDeviceInfo->minChannels = 1; + pDeviceInfo->maxChannels = MA_MAX_CHANNELS; + pDeviceInfo->minSampleRate = MA_SAMPLE_RATE_8000; + pDeviceInfo->maxSampleRate = MA_SAMPLE_RATE_384000; + + (void)pContext; + (void)shareMode; return MA_SUCCESS; } -#endif -ma_result ma_context_get_IAudioClient__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_IAudioClient** ppAudioClient, ma_WASAPIDeviceInterface** ppDeviceInterface) + +static void ma_device_uninit__null(ma_device* pDevice) { -#ifdef MA_WIN32_DESKTOP - return ma_context_get_IAudioClient_Desktop__wasapi(pContext, deviceType, pDeviceID, ppAudioClient, ppDeviceInterface); -#else - return ma_context_get_IAudioClient_UWP__wasapi(pContext, deviceType, pDeviceID, ppAudioClient, ppDeviceInterface); -#endif -} + MA_ASSERT(pDevice != NULL); + /* Keep it clean and wait for the device thread to finish before returning. */ + ma_device_do_operation__null(pDevice, MA_DEVICE_OP_KILL__NULL); + + /* At this point the loop in the device thread is as good as terminated so we can uninitialize our events. */ + ma_event_uninit(&pDevice->null_device.operationCompletionEvent); + ma_event_uninit(&pDevice->null_device.operationEvent); +} -ma_result ma_context_enumerate_devices__wasapi(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +static ma_result ma_device_init__null(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) { - /* Different enumeration for desktop and UWP. */ -#ifdef MA_WIN32_DESKTOP - /* Desktop */ - HRESULT hr; - ma_IMMDeviceEnumerator* pDeviceEnumerator; - ma_IMMDeviceCollection* pDeviceCollection; + ma_result result; + ma_uint32 periodSizeInFrames; - hr = ma_CoCreateInstance(pContext, MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator); - if (FAILED(hr)) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create device enumerator.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + MA_ASSERT(pDevice != NULL); + + MA_ZERO_OBJECT(&pDevice->null_device); + + if (pConfig->deviceType == ma_device_type_loopback) { + return MA_DEVICE_TYPE_NOT_SUPPORTED; } - /* Playback. */ - hr = ma_IMMDeviceEnumerator_EnumAudioEndpoints(pDeviceEnumerator, ma_eRender, MA_MM_DEVICE_STATE_ACTIVE, &pDeviceCollection); - if (SUCCEEDED(hr)) { - ma_context_enumerate_device_collection__wasapi(pContext, pDeviceCollection, ma_device_type_playback, callback, pUserData); - ma_IMMDeviceCollection_Release(pDeviceCollection); + periodSizeInFrames = pConfig->periodSizeInFrames; + if (periodSizeInFrames == 0) { + periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->periodSizeInMilliseconds, pConfig->sampleRate); } - /* Capture. */ - hr = ma_IMMDeviceEnumerator_EnumAudioEndpoints(pDeviceEnumerator, ma_eCapture, MA_MM_DEVICE_STATE_ACTIVE, &pDeviceCollection); - if (SUCCEEDED(hr)) { - ma_context_enumerate_device_collection__wasapi(pContext, pDeviceCollection, ma_device_type_capture, callback, pUserData); - ma_IMMDeviceCollection_Release(pDeviceCollection); + if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { + ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), "NULL Capture Device", (size_t)-1); + pDevice->capture.internalFormat = pConfig->capture.format; + pDevice->capture.internalChannels = pConfig->capture.channels; + ma_channel_map_copy(pDevice->capture.internalChannelMap, pConfig->capture.channelMap, pConfig->capture.channels); + pDevice->capture.internalPeriodSizeInFrames = periodSizeInFrames; + pDevice->capture.internalPeriods = pConfig->periods; + } + if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { + ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), "NULL Playback Device", (size_t)-1); + pDevice->playback.internalFormat = pConfig->playback.format; + pDevice->playback.internalChannels = pConfig->playback.channels; + ma_channel_map_copy(pDevice->playback.internalChannelMap, pConfig->playback.channelMap, pConfig->playback.channels); + pDevice->playback.internalPeriodSizeInFrames = periodSizeInFrames; + pDevice->playback.internalPeriods = pConfig->periods; } - ma_IMMDeviceEnumerator_Release(pDeviceEnumerator); -#else /* - UWP - - The MMDevice API is only supported on desktop applications. For now, while I'm still figuring out how to properly enumerate - over devices without using MMDevice, I'm restricting devices to defaults. - - Hint: DeviceInformation::FindAllAsync() with DeviceClass.AudioCapture/AudioRender. https://blogs.windows.com/buildingapps/2014/05/15/real-time-audio-in-windows-store-and-windows-phone-apps/ + In order to get timing right, we need to create a thread that does nothing but keeps track of the timer. This timer is started when the + first period is "written" to it, and then stopped in ma_device_stop__null(). */ - if (callback) { - ma_bool32 cbResult = MA_TRUE; - - /* Playback. */ - if (cbResult) { - ma_device_info deviceInfo; - ma_zero_object(&deviceInfo); - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); - cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); - } - - /* Capture. */ - if (cbResult) { - ma_device_info deviceInfo; - ma_zero_object(&deviceInfo); - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); - cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); - } + result = ma_event_init(pContext, &pDevice->null_device.operationEvent); + if (result != MA_SUCCESS) { + return result; } -#endif - return MA_SUCCESS; -} + result = ma_event_init(pContext, &pDevice->null_device.operationCompletionEvent); + if (result != MA_SUCCESS) { + return result; + } -ma_result ma_context_get_device_info__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) -{ -#ifdef MA_WIN32_DESKTOP - ma_IMMDevice* pMMDevice = NULL; - ma_result result; - - result = ma_context_get_MMDevice__wasapi(pContext, deviceType, pDeviceID, &pMMDevice); + result = ma_thread_create(pContext, &pDevice->thread, ma_device_thread__null, pDevice); if (result != MA_SUCCESS) { return result; } - result = ma_context_get_device_info_from_MMDevice__wasapi(pContext, pMMDevice, shareMode, MA_FALSE, pDeviceInfo); /* MA_FALSE = !onlySimpleInfo. */ + return MA_SUCCESS; +} - ma_IMMDevice_Release(pMMDevice); - return result; -#else - ma_IAudioClient* pAudioClient; - ma_result result; +static ma_result ma_device_start__null(ma_device* pDevice) +{ + MA_ASSERT(pDevice != NULL); - /* UWP currently only uses default devices. */ - if (deviceType == ma_device_type_playback) { - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); - } else { - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); - } + ma_device_do_operation__null(pDevice, MA_DEVICE_OP_START__NULL); - /* Not currently supporting exclusive mode on UWP. */ - if (shareMode == ma_share_mode_exclusive) { - return MA_ERROR; - } + ma_atomic_exchange_32(&pDevice->null_device.isStarted, MA_TRUE); + return MA_SUCCESS; +} - result = ma_context_get_IAudioClient_UWP__wasapi(pContext, deviceType, pDeviceID, &pAudioClient, NULL); - if (result != MA_SUCCESS) { - return result; - } +static ma_result ma_device_stop__null(ma_device* pDevice) +{ + MA_ASSERT(pDevice != NULL); - result = ma_context_get_device_info_from_IAudioClient__wasapi(pContext, NULL, pAudioClient, shareMode, pDeviceInfo); + ma_device_do_operation__null(pDevice, MA_DEVICE_OP_SUSPEND__NULL); - ma_IAudioClient_Release(pAudioClient); - return result; -#endif + ma_atomic_exchange_32(&pDevice->null_device.isStarted, MA_FALSE); + return MA_SUCCESS; } -void ma_device_uninit__wasapi(ma_device* pDevice) +static ma_result ma_device_write__null(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) { - ma_assert(pDevice != NULL); + ma_result result = MA_SUCCESS; + ma_uint32 totalPCMFramesProcessed; + ma_bool32 wasStartedOnEntry; -#ifdef MA_WIN32_DESKTOP - if (pDevice->wasapi.pDeviceEnumerator) { - ((ma_IMMDeviceEnumerator*)pDevice->wasapi.pDeviceEnumerator)->lpVtbl->UnregisterEndpointNotificationCallback((ma_IMMDeviceEnumerator*)pDevice->wasapi.pDeviceEnumerator, &pDevice->wasapi.notificationClient); - ma_IMMDeviceEnumerator_Release((ma_IMMDeviceEnumerator*)pDevice->wasapi.pDeviceEnumerator); + if (pFramesWritten != NULL) { + *pFramesWritten = 0; } -#endif - if (pDevice->wasapi.pRenderClient) { - ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient); - } - if (pDevice->wasapi.pCaptureClient) { - ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient); - } + wasStartedOnEntry = pDevice->null_device.isStarted; - if (pDevice->wasapi.pAudioClientPlayback) { - ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); - } - if (pDevice->wasapi.pAudioClientCapture) { - ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); - } + /* Keep going until everything has been read. */ + totalPCMFramesProcessed = 0; + while (totalPCMFramesProcessed < frameCount) { + ma_uint64 targetFrame; - if (pDevice->wasapi.hEventPlayback) { - CloseHandle(pDevice->wasapi.hEventPlayback); - } - if (pDevice->wasapi.hEventCapture) { - CloseHandle(pDevice->wasapi.hEventCapture); - } -} + /* If there are any frames remaining in the current period, consume those first. */ + if (pDevice->null_device.currentPeriodFramesRemainingPlayback > 0) { + ma_uint32 framesRemaining = (frameCount - totalPCMFramesProcessed); + ma_uint32 framesToProcess = pDevice->null_device.currentPeriodFramesRemainingPlayback; + if (framesToProcess > framesRemaining) { + framesToProcess = framesRemaining; + } + /* We don't actually do anything with pPCMFrames, so just mark it as unused to prevent a warning. */ + (void)pPCMFrames; -typedef struct -{ - /* Input. */ - ma_format formatIn; - ma_uint32 channelsIn; - ma_uint32 sampleRateIn; - ma_channel channelMapIn[MA_MAX_CHANNELS]; - ma_uint32 bufferSizeInFramesIn; - ma_uint32 bufferSizeInMillisecondsIn; - ma_uint32 periodsIn; - ma_bool32 usingDefaultFormat; - ma_bool32 usingDefaultChannels; - ma_bool32 usingDefaultSampleRate; - ma_bool32 usingDefaultChannelMap; - ma_share_mode shareMode; - ma_bool32 noAutoConvertSRC; - ma_bool32 noDefaultQualitySRC; + pDevice->null_device.currentPeriodFramesRemainingPlayback -= framesToProcess; + totalPCMFramesProcessed += framesToProcess; + } - /* Output. */ - ma_IAudioClient* pAudioClient; - ma_IAudioRenderClient* pRenderClient; - ma_IAudioCaptureClient* pCaptureClient; - ma_format formatOut; - ma_uint32 channelsOut; - ma_uint32 sampleRateOut; - ma_channel channelMapOut[MA_MAX_CHANNELS]; - ma_uint32 bufferSizeInFramesOut; - ma_uint32 periodSizeInFramesOut; - ma_uint32 periodsOut; - ma_bool32 usingAudioClient3; - char deviceName[256]; -} ma_device_init_internal_data__wasapi; + /* If we've consumed the current period we'll need to mark it as such an ensure the device is started if it's not already. */ + if (pDevice->null_device.currentPeriodFramesRemainingPlayback == 0) { + pDevice->null_device.currentPeriodFramesRemainingPlayback = 0; -ma_result ma_device_init_internal__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_init_internal_data__wasapi* pData) -{ - HRESULT hr; - ma_result result = MA_SUCCESS; - const char* errorMsg = ""; - MA_AUDCLNT_SHAREMODE shareMode = MA_AUDCLNT_SHAREMODE_SHARED; - DWORD streamFlags = 0; - MA_REFERENCE_TIME bufferDurationInMicroseconds; - ma_bool32 wasInitializedUsingIAudioClient3 = MA_FALSE; - WAVEFORMATEXTENSIBLE wf = {0}; - ma_WASAPIDeviceInterface* pDeviceInterface = NULL; - ma_IAudioClient2* pAudioClient2; - ma_uint32 nativeSampleRate; + if (!pDevice->null_device.isStarted && !wasStartedOnEntry) { + result = ma_device_start__null(pDevice); + if (result != MA_SUCCESS) { + break; + } + } + } + + /* If we've consumed the whole buffer we can return now. */ + MA_ASSERT(totalPCMFramesProcessed <= frameCount); + if (totalPCMFramesProcessed == frameCount) { + break; + } - ma_assert(pContext != NULL); - ma_assert(pData != NULL); + /* Getting here means we've still got more frames to consume, we but need to wait for it to become available. */ + targetFrame = pDevice->null_device.lastProcessedFramePlayback; + for (;;) { + ma_uint64 currentFrame; - /* This function is only used to initialize one device type: either playback, capture or loopback. Never full-duplex. */ - if (deviceType == ma_device_type_duplex) { - return MA_INVALID_ARGS; - } + /* Stop waiting if the device has been stopped. */ + if (!pDevice->null_device.isStarted) { + break; + } - pData->pAudioClient = NULL; - pData->pRenderClient = NULL; - pData->pCaptureClient = NULL; + currentFrame = ma_device_get_total_run_time_in_frames__null(pDevice); + if (currentFrame >= targetFrame) { + break; + } - streamFlags = MA_AUDCLNT_STREAMFLAGS_EVENTCALLBACK; - if (!pData->noAutoConvertSRC && !pData->usingDefaultSampleRate && pData->shareMode != ma_share_mode_exclusive) { /* <-- Exclusive streams must use the native sample rate. */ - streamFlags |= MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM; - } - if (!pData->noDefaultQualitySRC && !pData->usingDefaultSampleRate && (streamFlags & MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM) != 0) { - streamFlags |= MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY; - } - if (deviceType == ma_device_type_loopback) { - streamFlags |= MA_AUDCLNT_STREAMFLAGS_LOOPBACK; + /* Getting here means we haven't yet reached the target sample, so continue waiting. */ + ma_sleep(10); + } + + pDevice->null_device.lastProcessedFramePlayback += pDevice->playback.internalPeriodSizeInFrames; + pDevice->null_device.currentPeriodFramesRemainingPlayback = pDevice->playback.internalPeriodSizeInFrames; } - result = ma_context_get_IAudioClient__wasapi(pContext, deviceType, pDeviceID, &pData->pAudioClient, &pDeviceInterface); - if (result != MA_SUCCESS) { - goto done; + if (pFramesWritten != NULL) { + *pFramesWritten = totalPCMFramesProcessed; } + return result; +} - /* Try enabling hardware offloading. */ - hr = ma_IAudioClient_QueryInterface(pData->pAudioClient, &MA_IID_IAudioClient2, (void**)&pAudioClient2); - if (SUCCEEDED(hr)) { - BOOL isHardwareOffloadingSupported = 0; - hr = ma_IAudioClient2_IsOffloadCapable(pAudioClient2, MA_AudioCategory_Other, &isHardwareOffloadingSupported); - if (SUCCEEDED(hr) && isHardwareOffloadingSupported) { - ma_AudioClientProperties clientProperties; - ma_zero_object(&clientProperties); - clientProperties.cbSize = sizeof(clientProperties); - clientProperties.bIsOffload = 1; - clientProperties.eCategory = MA_AudioCategory_Other; - ma_IAudioClient2_SetClientProperties(pAudioClient2, &clientProperties); - } +static ma_result ma_device_read__null(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead) +{ + ma_result result = MA_SUCCESS; + ma_uint32 totalPCMFramesProcessed; - pAudioClient2->lpVtbl->Release(pAudioClient2); + if (pFramesRead != NULL) { + *pFramesRead = 0; } - /* Here is where we try to determine the best format to use with the device. If the client if wanting exclusive mode, first try finding the best format for that. If this fails, fall back to shared mode. */ - result = MA_FORMAT_NOT_SUPPORTED; - if (pData->shareMode == ma_share_mode_exclusive) { - #ifdef MA_WIN32_DESKTOP - /* In exclusive mode on desktop we always use the backend's native format. */ - ma_IPropertyStore* pStore = NULL; - hr = ma_IMMDevice_OpenPropertyStore(pDeviceInterface, STGM_READ, &pStore); - if (SUCCEEDED(hr)) { - PROPVARIANT prop; - ma_PropVariantInit(&prop); - hr = ma_IPropertyStore_GetValue(pStore, &MA_PKEY_AudioEngine_DeviceFormat, &prop); - if (SUCCEEDED(hr)) { - WAVEFORMATEX* pActualFormat = (WAVEFORMATEX*)prop.blob.pBlobData; - hr = ma_IAudioClient_IsFormatSupported((ma_IAudioClient*)pData->pAudioClient, MA_AUDCLNT_SHAREMODE_EXCLUSIVE, pActualFormat, NULL); - if (SUCCEEDED(hr)) { - ma_copy_memory(&wf, pActualFormat, sizeof(WAVEFORMATEXTENSIBLE)); - } + /* Keep going until everything has been read. */ + totalPCMFramesProcessed = 0; + while (totalPCMFramesProcessed < frameCount) { + ma_uint64 targetFrame; - ma_PropVariantClear(pContext, &prop); + /* If there are any frames remaining in the current period, consume those first. */ + if (pDevice->null_device.currentPeriodFramesRemainingCapture > 0) { + ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + ma_uint32 framesRemaining = (frameCount - totalPCMFramesProcessed); + ma_uint32 framesToProcess = pDevice->null_device.currentPeriodFramesRemainingCapture; + if (framesToProcess > framesRemaining) { + framesToProcess = framesRemaining; } - ma_IPropertyStore_Release(pStore); + /* We need to ensured the output buffer is zeroed. */ + MA_ZERO_MEMORY(ma_offset_ptr(pPCMFrames, totalPCMFramesProcessed*bpf), framesToProcess*bpf); + + pDevice->null_device.currentPeriodFramesRemainingCapture -= framesToProcess; + totalPCMFramesProcessed += framesToProcess; } - #else - /* - I do not know how to query the device's native format on UWP so for now I'm just disabling support for - exclusive mode. The alternative is to enumerate over different formats and check IsFormatSupported() - until you find one that works. - - TODO: Add support for exclusive mode to UWP. - */ - hr = S_FALSE; - #endif - if (hr == S_OK) { - shareMode = MA_AUDCLNT_SHAREMODE_EXCLUSIVE; - result = MA_SUCCESS; - } else { - result = MA_SHARE_MODE_NOT_SUPPORTED; + /* If we've consumed the current period we'll need to mark it as such an ensure the device is started if it's not already. */ + if (pDevice->null_device.currentPeriodFramesRemainingCapture == 0) { + pDevice->null_device.currentPeriodFramesRemainingCapture = 0; } - } else { - /* In shared mode we are always using the format reported by the operating system. */ - WAVEFORMATEXTENSIBLE* pNativeFormat = NULL; - hr = ma_IAudioClient_GetMixFormat((ma_IAudioClient*)pData->pAudioClient, (WAVEFORMATEX**)&pNativeFormat); - if (hr != S_OK) { - result = MA_FORMAT_NOT_SUPPORTED; - } else { - ma_copy_memory(&wf, pNativeFormat, sizeof(wf)); - result = MA_SUCCESS; + + /* If we've consumed the whole buffer we can return now. */ + MA_ASSERT(totalPCMFramesProcessed <= frameCount); + if (totalPCMFramesProcessed == frameCount) { + break; } - ma_CoTaskMemFree(pContext, pNativeFormat); + /* Getting here means we've still got more frames to consume, we but need to wait for it to become available. */ + targetFrame = pDevice->null_device.lastProcessedFrameCapture + pDevice->capture.internalPeriodSizeInFrames; + for (;;) { + ma_uint64 currentFrame; - shareMode = MA_AUDCLNT_SHAREMODE_SHARED; - } + /* Stop waiting if the device has been stopped. */ + if (!pDevice->null_device.isStarted) { + break; + } - /* Return an error if we still haven't found a format. */ - if (result != MA_SUCCESS) { - errorMsg = "[WASAPI] Failed to find best device mix format."; - goto done; + currentFrame = ma_device_get_total_run_time_in_frames__null(pDevice); + if (currentFrame >= targetFrame) { + break; + } + + /* Getting here means we haven't yet reached the target sample, so continue waiting. */ + ma_sleep(10); + } + + pDevice->null_device.lastProcessedFrameCapture += pDevice->capture.internalPeriodSizeInFrames; + pDevice->null_device.currentPeriodFramesRemainingCapture = pDevice->capture.internalPeriodSizeInFrames; } - /* - Override the native sample rate with the one requested by the caller, but only if we're not using the default sample rate. We'll use - WASAPI to perform the sample rate conversion. - */ - nativeSampleRate = wf.Format.nSamplesPerSec; - if (streamFlags & MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM) { - wf.Format.nSamplesPerSec = pData->sampleRateIn; - wf.Format.nAvgBytesPerSec = wf.Format.nSamplesPerSec * wf.Format.nBlockAlign; + if (pFramesRead != NULL) { + *pFramesRead = totalPCMFramesProcessed; } - pData->formatOut = ma_format_from_WAVEFORMATEX((WAVEFORMATEX*)&wf); - pData->channelsOut = wf.Format.nChannels; - pData->sampleRateOut = wf.Format.nSamplesPerSec; + return result; +} - /* Get the internal channel map based on the channel mask. */ - ma_channel_mask_to_channel_map__win32(wf.dwChannelMask, pData->channelsOut, pData->channelMapOut); +static ma_result ma_device_main_loop__null(ma_device* pDevice) +{ + ma_result result = MA_SUCCESS; + ma_bool32 exitLoop = MA_FALSE; + + MA_ASSERT(pDevice != NULL); - /* If we're using a default buffer size we need to calculate it based on the efficiency of the system. */ - pData->periodsOut = pData->periodsIn; - pData->bufferSizeInFramesOut = pData->bufferSizeInFramesIn; - if (pData->bufferSizeInFramesOut == 0) { - pData->bufferSizeInFramesOut = ma_calculate_buffer_size_in_frames_from_milliseconds(pData->bufferSizeInMillisecondsIn, wf.Format.nSamplesPerSec); + /* The capture device needs to be started immediately. */ + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + result = ma_device_start__null(pDevice); + if (result != MA_SUCCESS) { + return result; + } } - bufferDurationInMicroseconds = ((ma_uint64)pData->bufferSizeInFramesOut * 1000 * 1000) / wf.Format.nSamplesPerSec; + while (ma_device__get_state(pDevice) == MA_STATE_STARTED && !exitLoop) { + switch (pDevice->type) + { + case ma_device_type_duplex: + { + /* The process is: device_read -> convert -> callback -> convert -> device_write */ + ma_uint32 totalCapturedDeviceFramesProcessed = 0; + ma_uint32 capturedDevicePeriodSizeInFrames = ma_min(pDevice->capture.internalPeriodSizeInFrames, pDevice->playback.internalPeriodSizeInFrames); + + while (totalCapturedDeviceFramesProcessed < capturedDevicePeriodSizeInFrames) { + ma_uint8 capturedDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint8 playbackDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 capturedDeviceDataCapInFrames = sizeof(capturedDeviceData) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + ma_uint32 playbackDeviceDataCapInFrames = sizeof(playbackDeviceData) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + ma_uint32 capturedDeviceFramesRemaining; + ma_uint32 capturedDeviceFramesProcessed; + ma_uint32 capturedDeviceFramesToProcess; + ma_uint32 capturedDeviceFramesToTryProcessing = capturedDevicePeriodSizeInFrames - totalCapturedDeviceFramesProcessed; + if (capturedDeviceFramesToTryProcessing > capturedDeviceDataCapInFrames) { + capturedDeviceFramesToTryProcessing = capturedDeviceDataCapInFrames; + } + + result = ma_device_read__null(pDevice, capturedDeviceData, capturedDeviceFramesToTryProcessing, &capturedDeviceFramesToProcess); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } + capturedDeviceFramesRemaining = capturedDeviceFramesToProcess; + capturedDeviceFramesProcessed = 0; - /* Slightly different initialization for shared and exclusive modes. We try exclusive mode first, and if it fails, fall back to shared mode. */ - if (shareMode == MA_AUDCLNT_SHAREMODE_EXCLUSIVE) { - MA_REFERENCE_TIME bufferDuration = (bufferDurationInMicroseconds / pData->periodsOut) * 10; + /* At this point we have our captured data in device format and we now need to convert it to client format. */ + for (;;) { + ma_uint8 capturedClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint8 playbackClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 capturedClientDataCapInFrames = sizeof(capturedClientData) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); + ma_uint32 playbackClientDataCapInFrames = sizeof(playbackClientData) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); + ma_uint64 capturedClientFramesToProcessThisIteration = ma_min(capturedClientDataCapInFrames, playbackClientDataCapInFrames); + ma_uint64 capturedDeviceFramesToProcessThisIteration = capturedDeviceFramesRemaining; + ma_uint8* pRunningCapturedDeviceFrames = ma_offset_ptr(capturedDeviceData, capturedDeviceFramesProcessed * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); + + /* Convert capture data from device format to client format. */ + result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningCapturedDeviceFrames, &capturedDeviceFramesToProcessThisIteration, capturedClientData, &capturedClientFramesToProcessThisIteration); + if (result != MA_SUCCESS) { + break; + } - /* - If the periodicy is too small, Initialize() will fail with AUDCLNT_E_INVALID_DEVICE_PERIOD. In this case we should just keep increasing - it and trying it again. - */ - hr = E_FAIL; - for (;;) { - hr = ma_IAudioClient_Initialize((ma_IAudioClient*)pData->pAudioClient, shareMode, streamFlags, bufferDuration, bufferDuration, (WAVEFORMATEX*)&wf, NULL); - if (hr == MA_AUDCLNT_E_INVALID_DEVICE_PERIOD) { - if (bufferDuration > 500*10000) { - break; - } else { - if (bufferDuration == 0) { /* <-- Just a sanity check to prevent an infinit loop. Should never happen, but it makes me feel better. */ + /* + If we weren't able to generate any output frames it must mean we've exhaused all of our input. The only time this would not be the case is if capturedClientData was too small + which should never be the case when it's of the size MA_DATA_CONVERTER_STACK_BUFFER_SIZE. + */ + if (capturedClientFramesToProcessThisIteration == 0) { + break; + } + + ma_device__on_data(pDevice, playbackClientData, capturedClientData, (ma_uint32)capturedClientFramesToProcessThisIteration); /* Safe cast .*/ + + capturedDeviceFramesProcessed += (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */ + capturedDeviceFramesRemaining -= (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */ + + /* At this point the playbackClientData buffer should be holding data that needs to be written to the device. */ + for (;;) { + ma_uint64 convertedClientFrameCount = capturedClientFramesToProcessThisIteration; + ma_uint64 convertedDeviceFrameCount = playbackDeviceDataCapInFrames; + result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, playbackClientData, &convertedClientFrameCount, playbackDeviceData, &convertedDeviceFrameCount); + if (result != MA_SUCCESS) { + break; + } + + result = ma_device_write__null(pDevice, playbackDeviceData, (ma_uint32)convertedDeviceFrameCount, NULL); /* Safe cast. */ + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } + + capturedClientFramesToProcessThisIteration -= (ma_uint32)convertedClientFrameCount; /* Safe cast. */ + if (capturedClientFramesToProcessThisIteration == 0) { + break; + } + } + + /* In case an error happened from ma_device_write__null()... */ + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } + } + + totalCapturedDeviceFramesProcessed += capturedDeviceFramesProcessed; + } + } break; + + case ma_device_type_capture: + { + /* We read in chunks of the period size, but use a stack allocated buffer for the intermediary. */ + ma_uint8 intermediaryBuffer[8192]; + ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + ma_uint32 periodSizeInFrames = pDevice->capture.internalPeriodSizeInFrames; + ma_uint32 framesReadThisPeriod = 0; + while (framesReadThisPeriod < periodSizeInFrames) { + ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesReadThisPeriod; + ma_uint32 framesProcessed; + ma_uint32 framesToReadThisIteration = framesRemainingInPeriod; + if (framesToReadThisIteration > intermediaryBufferSizeInFrames) { + framesToReadThisIteration = intermediaryBufferSizeInFrames; + } + + result = ma_device_read__null(pDevice, intermediaryBuffer, framesToReadThisIteration, &framesProcessed); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } + + ma_device__send_frames_to_client(pDevice, framesProcessed, intermediaryBuffer); + + framesReadThisPeriod += framesProcessed; + } + } break; + + case ma_device_type_playback: + { + /* We write in chunks of the period size, but use a stack allocated buffer for the intermediary. */ + ma_uint8 intermediaryBuffer[8192]; + ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + ma_uint32 periodSizeInFrames = pDevice->playback.internalPeriodSizeInFrames; + ma_uint32 framesWrittenThisPeriod = 0; + while (framesWrittenThisPeriod < periodSizeInFrames) { + ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesWrittenThisPeriod; + ma_uint32 framesProcessed; + ma_uint32 framesToWriteThisIteration = framesRemainingInPeriod; + if (framesToWriteThisIteration > intermediaryBufferSizeInFrames) { + framesToWriteThisIteration = intermediaryBufferSizeInFrames; + } + + ma_device__read_frames_from_client(pDevice, framesToWriteThisIteration, intermediaryBuffer); + + result = ma_device_write__null(pDevice, intermediaryBuffer, framesToWriteThisIteration, &framesProcessed); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; break; } - bufferDuration = bufferDuration * 2; - continue; + framesWrittenThisPeriod += framesProcessed; } - } else { - break; - } + } break; + + /* To silence a warning. Will never hit this. */ + case ma_device_type_loopback: + default: break; } - - if (hr == MA_AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED) { - ma_uint32 bufferSizeInFrames; - hr = ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pData->pAudioClient, &bufferSizeInFrames); - if (SUCCEEDED(hr)) { - bufferDuration = (MA_REFERENCE_TIME)((10000.0 * 1000 / wf.Format.nSamplesPerSec * bufferSizeInFrames) + 0.5); + } - /* Unfortunately we need to release and re-acquire the audio client according to MSDN. Seems silly - why not just call IAudioClient_Initialize() again?! */ - ma_IAudioClient_Release((ma_IAudioClient*)pData->pAudioClient); - #ifdef MA_WIN32_DESKTOP - hr = ma_IMMDevice_Activate(pDeviceInterface, &MA_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pData->pAudioClient); - #else - hr = ma_IUnknown_QueryInterface(pDeviceInterface, &MA_IID_IAudioClient, (void**)&pData->pAudioClient); - #endif + /* Here is where the device is started. */ + ma_device_stop__null(pDevice); - if (SUCCEEDED(hr)) { - hr = ma_IAudioClient_Initialize((ma_IAudioClient*)pData->pAudioClient, shareMode, streamFlags, bufferDuration, bufferDuration, (WAVEFORMATEX*)&wf, NULL); - } - } - } + return result; +} - if (FAILED(hr)) { - /* Failed to initialize in exclusive mode. Don't fall back to shared mode - instead tell the client about it. They can reinitialize in shared mode if they want. */ - if (hr == E_ACCESSDENIED) { - errorMsg = "[WASAPI] Failed to initialize device in exclusive mode. Access denied.", result = MA_ACCESS_DENIED; - } else if (hr == MA_AUDCLNT_E_DEVICE_IN_USE) { - errorMsg = "[WASAPI] Failed to initialize device in exclusive mode. Device in use.", result = MA_DEVICE_BUSY; - } else { - errorMsg = "[WASAPI] Failed to initialize device in exclusive mode."; result = MA_SHARE_MODE_NOT_SUPPORTED; - } - goto done; - } - } +static ma_result ma_context_uninit__null(ma_context* pContext) +{ + MA_ASSERT(pContext != NULL); + MA_ASSERT(pContext->backend == ma_backend_null); - if (shareMode == MA_AUDCLNT_SHAREMODE_SHARED) { - /* - Low latency shared mode via IAudioClient3. + (void)pContext; + return MA_SUCCESS; +} - NOTE - ==== - Contrary to the documentation on MSDN (https://docs.microsoft.com/en-us/windows/win32/api/audioclient/nf-audioclient-iaudioclient3-initializesharedaudiostream), the - use of AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM and AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY with IAudioClient3_InitializeSharedAudioStream() absolutely does not work. Using - any of these flags will result in HRESULT code 0x88890021. The other problem is that calling IAudioClient3_GetSharedModeEnginePeriod() with a sample rate different to - that returned by IAudioClient_GetMixFormat() also results in an error. I'm therefore disabling low-latency shared mode with AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM. - */ -#ifndef MA_WASAPI_NO_LOW_LATENCY_SHARED_MODE - if ((streamFlags & MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM) == 0 || nativeSampleRate == wf.Format.nSamplesPerSec) { - ma_IAudioClient3* pAudioClient3 = NULL; - hr = ma_IAudioClient_QueryInterface(pData->pAudioClient, &MA_IID_IAudioClient3, (void**)&pAudioClient3); - if (SUCCEEDED(hr)) { - UINT32 defaultPeriodInFrames; - UINT32 fundamentalPeriodInFrames; - UINT32 minPeriodInFrames; - UINT32 maxPeriodInFrames; - hr = ma_IAudioClient3_GetSharedModeEnginePeriod(pAudioClient3, (WAVEFORMATEX*)&wf, &defaultPeriodInFrames, &fundamentalPeriodInFrames, &minPeriodInFrames, &maxPeriodInFrames); - if (SUCCEEDED(hr)) { - UINT32 desiredPeriodInFrames = pData->bufferSizeInFramesOut / pData->periodsOut; - UINT32 actualPeriodInFrames = desiredPeriodInFrames; +static ma_result ma_context_init__null(const ma_context_config* pConfig, ma_context* pContext) +{ + MA_ASSERT(pContext != NULL); - /* Make sure the period size is a multiple of fundamentalPeriodInFrames. */ - actualPeriodInFrames = actualPeriodInFrames / fundamentalPeriodInFrames; - actualPeriodInFrames = actualPeriodInFrames * fundamentalPeriodInFrames; + (void)pConfig; - /* The period needs to be clamped between minPeriodInFrames and maxPeriodInFrames. */ - actualPeriodInFrames = ma_clamp(actualPeriodInFrames, minPeriodInFrames, maxPeriodInFrames); + pContext->onUninit = ma_context_uninit__null; + pContext->onDeviceIDEqual = ma_context_is_device_id_equal__null; + pContext->onEnumDevices = ma_context_enumerate_devices__null; + pContext->onGetDeviceInfo = ma_context_get_device_info__null; + pContext->onDeviceInit = ma_device_init__null; + pContext->onDeviceUninit = ma_device_uninit__null; + pContext->onDeviceStart = NULL; /* Not required for synchronous backends. */ + pContext->onDeviceStop = NULL; /* Not required for synchronous backends. */ + pContext->onDeviceMainLoop = ma_device_main_loop__null; - #if defined(MA_DEBUG_OUTPUT) - printf("[WASAPI] Trying IAudioClient3_InitializeSharedAudioStream(actualPeriodInFrames=%d)\n", actualPeriodInFrames); - printf(" defaultPeriodInFrames=%d\n", defaultPeriodInFrames); - printf(" fundamentalPeriodInFrames=%d\n", fundamentalPeriodInFrames); - printf(" minPeriodInFrames=%d\n", minPeriodInFrames); - printf(" maxPeriodInFrames=%d\n", maxPeriodInFrames); - #endif + /* The null backend always works. */ + return MA_SUCCESS; +} +#endif - /* If the client requested a largish buffer than we don't actually want to use low latency shared mode because it forces small buffers. */ - if (actualPeriodInFrames >= desiredPeriodInFrames) { - /* - MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM | MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY must not be in the stream flags. If either of these are specified, - IAudioClient3_InitializeSharedAudioStream() will fail. - */ - hr = ma_IAudioClient3_InitializeSharedAudioStream(pAudioClient3, streamFlags & ~(MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM | MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY), actualPeriodInFrames, (WAVEFORMATEX*)&wf, NULL); - if (SUCCEEDED(hr)) { - wasInitializedUsingIAudioClient3 = MA_TRUE; - pData->periodSizeInFramesOut = actualPeriodInFrames; - pData->bufferSizeInFramesOut = actualPeriodInFrames * pData->periodsOut; - #if defined(MA_DEBUG_OUTPUT) - printf("[WASAPI] Using IAudioClient3\n"); - printf(" periodSizeInFramesOut=%d\n", pData->periodSizeInFramesOut); - printf(" bufferSizeInFramesOut=%d\n", pData->bufferSizeInFramesOut); - #endif - } else { - #if defined(MA_DEBUG_OUTPUT) - printf("[WASAPI] IAudioClient3_InitializeSharedAudioStream failed. Falling back to IAudioClient.\n"); - #endif - } - } else { - #if defined(MA_DEBUG_OUTPUT) - printf("[WASAPI] Not using IAudioClient3 because the desired period size is larger than the maximum supported by IAudioClient3.\n"); - #endif - } - } else { - #if defined(MA_DEBUG_OUTPUT) - printf("[WASAPI] IAudioClient3_GetSharedModeEnginePeriod failed. Falling back to IAudioClient.\n"); - #endif - } - ma_IAudioClient3_Release(pAudioClient3); - pAudioClient3 = NULL; - } - } +/******************************************************************************* + +WIN32 COMMON + +*******************************************************************************/ +#if defined(MA_WIN32) +#if defined(MA_WIN32_DESKTOP) + #define ma_CoInitializeEx(pContext, pvReserved, dwCoInit) ((MA_PFN_CoInitializeEx)pContext->win32.CoInitializeEx)(pvReserved, dwCoInit) + #define ma_CoUninitialize(pContext) ((MA_PFN_CoUninitialize)pContext->win32.CoUninitialize)() + #define ma_CoCreateInstance(pContext, rclsid, pUnkOuter, dwClsContext, riid, ppv) ((MA_PFN_CoCreateInstance)pContext->win32.CoCreateInstance)(rclsid, pUnkOuter, dwClsContext, riid, ppv) + #define ma_CoTaskMemFree(pContext, pv) ((MA_PFN_CoTaskMemFree)pContext->win32.CoTaskMemFree)(pv) + #define ma_PropVariantClear(pContext, pvar) ((MA_PFN_PropVariantClear)pContext->win32.PropVariantClear)(pvar) #else - #if defined(MA_DEBUG_OUTPUT) - printf("[WASAPI] Not using IAudioClient3 because MA_WASAPI_NO_LOW_LATENCY_SHARED_MODE is enabled.\n"); - #endif + #define ma_CoInitializeEx(pContext, pvReserved, dwCoInit) CoInitializeEx(pvReserved, dwCoInit) + #define ma_CoUninitialize(pContext) CoUninitialize() + #define ma_CoCreateInstance(pContext, rclsid, pUnkOuter, dwClsContext, riid, ppv) CoCreateInstance(rclsid, pUnkOuter, dwClsContext, riid, ppv) + #define ma_CoTaskMemFree(pContext, pv) CoTaskMemFree(pv) + #define ma_PropVariantClear(pContext, pvar) PropVariantClear(pvar) #endif - /* If we don't have an IAudioClient3 then we need to use the normal initialization routine. */ - if (!wasInitializedUsingIAudioClient3) { - MA_REFERENCE_TIME bufferDuration = bufferDurationInMicroseconds*10; - hr = ma_IAudioClient_Initialize((ma_IAudioClient*)pData->pAudioClient, shareMode, streamFlags, bufferDuration, 0, (WAVEFORMATEX*)&wf, NULL); - if (FAILED(hr)) { - if (hr == E_ACCESSDENIED) { - errorMsg = "[WASAPI] Failed to initialize device. Access denied.", result = MA_ACCESS_DENIED; - } else if (hr == MA_AUDCLNT_E_DEVICE_IN_USE) { - errorMsg = "[WASAPI] Failed to initialize device. Device in use.", result = MA_DEVICE_BUSY; - } else { - errorMsg = "[WASAPI] Failed to initialize device.", result = MA_FAILED_TO_OPEN_BACKEND_DEVICE; - } +#if !defined(MAXULONG_PTR) +typedef size_t DWORD_PTR; +#endif - goto done; - } - } +#if !defined(WAVE_FORMAT_44M08) +#define WAVE_FORMAT_44M08 0x00000100 +#define WAVE_FORMAT_44S08 0x00000200 +#define WAVE_FORMAT_44M16 0x00000400 +#define WAVE_FORMAT_44S16 0x00000800 +#define WAVE_FORMAT_48M08 0x00001000 +#define WAVE_FORMAT_48S08 0x00002000 +#define WAVE_FORMAT_48M16 0x00004000 +#define WAVE_FORMAT_48S16 0x00008000 +#define WAVE_FORMAT_96M08 0x00010000 +#define WAVE_FORMAT_96S08 0x00020000 +#define WAVE_FORMAT_96M16 0x00040000 +#define WAVE_FORMAT_96S16 0x00080000 +#endif + +#ifndef SPEAKER_FRONT_LEFT +#define SPEAKER_FRONT_LEFT 0x1 +#define SPEAKER_FRONT_RIGHT 0x2 +#define SPEAKER_FRONT_CENTER 0x4 +#define SPEAKER_LOW_FREQUENCY 0x8 +#define SPEAKER_BACK_LEFT 0x10 +#define SPEAKER_BACK_RIGHT 0x20 +#define SPEAKER_FRONT_LEFT_OF_CENTER 0x40 +#define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80 +#define SPEAKER_BACK_CENTER 0x100 +#define SPEAKER_SIDE_LEFT 0x200 +#define SPEAKER_SIDE_RIGHT 0x400 +#define SPEAKER_TOP_CENTER 0x800 +#define SPEAKER_TOP_FRONT_LEFT 0x1000 +#define SPEAKER_TOP_FRONT_CENTER 0x2000 +#define SPEAKER_TOP_FRONT_RIGHT 0x4000 +#define SPEAKER_TOP_BACK_LEFT 0x8000 +#define SPEAKER_TOP_BACK_CENTER 0x10000 +#define SPEAKER_TOP_BACK_RIGHT 0x20000 +#endif + +/* +The SDK that comes with old versions of MSVC (VC6, for example) does not appear to define WAVEFORMATEXTENSIBLE. We +define our own implementation in this case. +*/ +#if (defined(_MSC_VER) && !defined(_WAVEFORMATEXTENSIBLE_)) || defined(__DMC__) +typedef struct +{ + WAVEFORMATEX Format; + union + { + WORD wValidBitsPerSample; + WORD wSamplesPerBlock; + WORD wReserved; + } Samples; + DWORD dwChannelMask; + GUID SubFormat; +} WAVEFORMATEXTENSIBLE; +#endif + +#ifndef WAVE_FORMAT_EXTENSIBLE +#define WAVE_FORMAT_EXTENSIBLE 0xFFFE +#endif + +#ifndef WAVE_FORMAT_IEEE_FLOAT +#define WAVE_FORMAT_IEEE_FLOAT 0x0003 +#endif + +static GUID MA_GUID_NULL = {0x00000000, 0x0000, 0x0000, {0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00}}; + +/* Converts an individual Win32-style channel identifier (SPEAKER_FRONT_LEFT, etc.) to miniaudio. */ +static ma_uint8 ma_channel_id_to_ma__win32(DWORD id) +{ + switch (id) + { + case SPEAKER_FRONT_LEFT: return MA_CHANNEL_FRONT_LEFT; + case SPEAKER_FRONT_RIGHT: return MA_CHANNEL_FRONT_RIGHT; + case SPEAKER_FRONT_CENTER: return MA_CHANNEL_FRONT_CENTER; + case SPEAKER_LOW_FREQUENCY: return MA_CHANNEL_LFE; + case SPEAKER_BACK_LEFT: return MA_CHANNEL_BACK_LEFT; + case SPEAKER_BACK_RIGHT: return MA_CHANNEL_BACK_RIGHT; + case SPEAKER_FRONT_LEFT_OF_CENTER: return MA_CHANNEL_FRONT_LEFT_CENTER; + case SPEAKER_FRONT_RIGHT_OF_CENTER: return MA_CHANNEL_FRONT_RIGHT_CENTER; + case SPEAKER_BACK_CENTER: return MA_CHANNEL_BACK_CENTER; + case SPEAKER_SIDE_LEFT: return MA_CHANNEL_SIDE_LEFT; + case SPEAKER_SIDE_RIGHT: return MA_CHANNEL_SIDE_RIGHT; + case SPEAKER_TOP_CENTER: return MA_CHANNEL_TOP_CENTER; + case SPEAKER_TOP_FRONT_LEFT: return MA_CHANNEL_TOP_FRONT_LEFT; + case SPEAKER_TOP_FRONT_CENTER: return MA_CHANNEL_TOP_FRONT_CENTER; + case SPEAKER_TOP_FRONT_RIGHT: return MA_CHANNEL_TOP_FRONT_RIGHT; + case SPEAKER_TOP_BACK_LEFT: return MA_CHANNEL_TOP_BACK_LEFT; + case SPEAKER_TOP_BACK_CENTER: return MA_CHANNEL_TOP_BACK_CENTER; + case SPEAKER_TOP_BACK_RIGHT: return MA_CHANNEL_TOP_BACK_RIGHT; + default: return 0; } +} - if (!wasInitializedUsingIAudioClient3) { - hr = ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pData->pAudioClient, &pData->bufferSizeInFramesOut); - if (FAILED(hr)) { - errorMsg = "[WASAPI] Failed to get audio client's actual buffer size.", result = MA_FAILED_TO_OPEN_BACKEND_DEVICE; - goto done; - } - - pData->periodSizeInFramesOut = pData->bufferSizeInFramesOut / pData->periodsOut; +/* Converts an individual miniaudio channel identifier (MA_CHANNEL_FRONT_LEFT, etc.) to Win32-style. */ +static DWORD ma_channel_id_to_win32(DWORD id) +{ + switch (id) + { + case MA_CHANNEL_MONO: return SPEAKER_FRONT_CENTER; + case MA_CHANNEL_FRONT_LEFT: return SPEAKER_FRONT_LEFT; + case MA_CHANNEL_FRONT_RIGHT: return SPEAKER_FRONT_RIGHT; + case MA_CHANNEL_FRONT_CENTER: return SPEAKER_FRONT_CENTER; + case MA_CHANNEL_LFE: return SPEAKER_LOW_FREQUENCY; + case MA_CHANNEL_BACK_LEFT: return SPEAKER_BACK_LEFT; + case MA_CHANNEL_BACK_RIGHT: return SPEAKER_BACK_RIGHT; + case MA_CHANNEL_FRONT_LEFT_CENTER: return SPEAKER_FRONT_LEFT_OF_CENTER; + case MA_CHANNEL_FRONT_RIGHT_CENTER: return SPEAKER_FRONT_RIGHT_OF_CENTER; + case MA_CHANNEL_BACK_CENTER: return SPEAKER_BACK_CENTER; + case MA_CHANNEL_SIDE_LEFT: return SPEAKER_SIDE_LEFT; + case MA_CHANNEL_SIDE_RIGHT: return SPEAKER_SIDE_RIGHT; + case MA_CHANNEL_TOP_CENTER: return SPEAKER_TOP_CENTER; + case MA_CHANNEL_TOP_FRONT_LEFT: return SPEAKER_TOP_FRONT_LEFT; + case MA_CHANNEL_TOP_FRONT_CENTER: return SPEAKER_TOP_FRONT_CENTER; + case MA_CHANNEL_TOP_FRONT_RIGHT: return SPEAKER_TOP_FRONT_RIGHT; + case MA_CHANNEL_TOP_BACK_LEFT: return SPEAKER_TOP_BACK_LEFT; + case MA_CHANNEL_TOP_BACK_CENTER: return SPEAKER_TOP_BACK_CENTER; + case MA_CHANNEL_TOP_BACK_RIGHT: return SPEAKER_TOP_BACK_RIGHT; + default: return 0; } +} - pData->usingAudioClient3 = wasInitializedUsingIAudioClient3; +/* Converts a channel mapping to a Win32-style channel mask. */ +static DWORD ma_channel_map_to_channel_mask__win32(const ma_channel channelMap[MA_MAX_CHANNELS], ma_uint32 channels) +{ + DWORD dwChannelMask = 0; + ma_uint32 iChannel; - if (deviceType == ma_device_type_playback) { - hr = ma_IAudioClient_GetService((ma_IAudioClient*)pData->pAudioClient, &MA_IID_IAudioRenderClient, (void**)&pData->pRenderClient); - } else { - hr = ma_IAudioClient_GetService((ma_IAudioClient*)pData->pAudioClient, &MA_IID_IAudioCaptureClient, (void**)&pData->pCaptureClient); + for (iChannel = 0; iChannel < channels; ++iChannel) { + dwChannelMask |= ma_channel_id_to_win32(channelMap[iChannel]); } - if (FAILED(hr)) { - errorMsg = "[WASAPI] Failed to get audio client service.", result = MA_API_NOT_FOUND; - goto done; - } + return dwChannelMask; +} +/* Converts a Win32-style channel mask to a miniaudio channel map. */ +static void ma_channel_mask_to_channel_map__win32(DWORD dwChannelMask, ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) +{ + if (channels == 1 && dwChannelMask == 0) { + channelMap[0] = MA_CHANNEL_MONO; + } else if (channels == 2 && dwChannelMask == 0) { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + } else { + if (channels == 1 && (dwChannelMask & SPEAKER_FRONT_CENTER) != 0) { + channelMap[0] = MA_CHANNEL_MONO; + } else { + /* Just iterate over each bit. */ + ma_uint32 iChannel = 0; + ma_uint32 iBit; - /* Grab the name of the device. */ -#ifdef MA_WIN32_DESKTOP - { - ma_IPropertyStore *pProperties; - hr = ma_IMMDevice_OpenPropertyStore(pDeviceInterface, STGM_READ, &pProperties); - if (SUCCEEDED(hr)) { - PROPVARIANT varName; - ma_PropVariantInit(&varName); - hr = ma_IPropertyStore_GetValue(pProperties, &MA_PKEY_Device_FriendlyName, &varName); - if (SUCCEEDED(hr)) { - WideCharToMultiByte(CP_UTF8, 0, varName.pwszVal, -1, pData->deviceName, sizeof(pData->deviceName), 0, FALSE); - ma_PropVariantClear(pContext, &varName); + for (iBit = 0; iBit < 32; ++iBit) { + DWORD bitValue = (dwChannelMask & (1UL << iBit)); + if (bitValue != 0) { + /* The bit is set. */ + channelMap[iChannel] = ma_channel_id_to_ma__win32(bitValue); + iChannel += 1; + } } - - ma_IPropertyStore_Release(pProperties); } } -#endif +} -done: - /* Clean up. */ -#ifdef MA_WIN32_DESKTOP - if (pDeviceInterface != NULL) { - ma_IMMDevice_Release(pDeviceInterface); - } +#ifdef __cplusplus +static ma_bool32 ma_is_guid_equal(const void* a, const void* b) +{ + return IsEqualGUID(*(const GUID*)a, *(const GUID*)b); +} #else - if (pDeviceInterface != NULL) { - ma_IUnknown_Release(pDeviceInterface); - } +#define ma_is_guid_equal(a, b) IsEqualGUID((const GUID*)a, (const GUID*)b) #endif - if (result != MA_SUCCESS) { - if (pData->pRenderClient) { - ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pData->pRenderClient); - pData->pRenderClient = NULL; - } - if (pData->pCaptureClient) { - ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pData->pCaptureClient); - pData->pCaptureClient = NULL; +static ma_format ma_format_from_WAVEFORMATEX(const WAVEFORMATEX* pWF) +{ + MA_ASSERT(pWF != NULL); + + if (pWF->wFormatTag == WAVE_FORMAT_EXTENSIBLE) { + const WAVEFORMATEXTENSIBLE* pWFEX = (const WAVEFORMATEXTENSIBLE*)pWF; + if (ma_is_guid_equal(&pWFEX->SubFormat, &MA_GUID_KSDATAFORMAT_SUBTYPE_PCM)) { + if (pWFEX->Samples.wValidBitsPerSample == 32) { + return ma_format_s32; + } + if (pWFEX->Samples.wValidBitsPerSample == 24) { + if (pWFEX->Format.wBitsPerSample == 32) { + /*return ma_format_s24_32;*/ + } + if (pWFEX->Format.wBitsPerSample == 24) { + return ma_format_s24; + } + } + if (pWFEX->Samples.wValidBitsPerSample == 16) { + return ma_format_s16; + } + if (pWFEX->Samples.wValidBitsPerSample == 8) { + return ma_format_u8; + } } - if (pData->pAudioClient) { - ma_IAudioClient_Release((ma_IAudioClient*)pData->pAudioClient); - pData->pAudioClient = NULL; + if (ma_is_guid_equal(&pWFEX->SubFormat, &MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)) { + if (pWFEX->Samples.wValidBitsPerSample == 32) { + return ma_format_f32; + } + /* + if (pWFEX->Samples.wValidBitsPerSample == 64) { + return ma_format_f64; + } + */ } - - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, errorMsg, result); } else { - return MA_SUCCESS; + if (pWF->wFormatTag == WAVE_FORMAT_PCM) { + if (pWF->wBitsPerSample == 32) { + return ma_format_s32; + } + if (pWF->wBitsPerSample == 24) { + return ma_format_s24; + } + if (pWF->wBitsPerSample == 16) { + return ma_format_s16; + } + if (pWF->wBitsPerSample == 8) { + return ma_format_u8; + } + } + if (pWF->wFormatTag == WAVE_FORMAT_IEEE_FLOAT) { + if (pWF->wBitsPerSample == 32) { + return ma_format_f32; + } + if (pWF->wBitsPerSample == 64) { + /*return ma_format_f64;*/ + } + } } + + return ma_format_unknown; } +#endif -ma_result ma_device_reinit__wasapi(ma_device* pDevice, ma_device_type deviceType) -{ - ma_device_init_internal_data__wasapi data; - ma_result result; - ma_assert(pDevice != NULL); +/******************************************************************************* - /* We only re-initialize the playback or capture device. Never a full-duplex device. */ - if (deviceType == ma_device_type_duplex) { - return MA_INVALID_ARGS; - } +WASAPI Backend - if (deviceType == ma_device_type_playback) { - data.formatIn = pDevice->playback.format; - data.channelsIn = pDevice->playback.channels; - ma_copy_memory(data.channelMapIn, pDevice->playback.channelMap, sizeof(pDevice->playback.channelMap)); - data.shareMode = pDevice->playback.shareMode; - data.usingDefaultFormat = pDevice->playback.usingDefaultFormat; - data.usingDefaultChannels = pDevice->playback.usingDefaultChannels; - data.usingDefaultChannelMap = pDevice->playback.usingDefaultChannelMap; - } else { - data.formatIn = pDevice->capture.format; - data.channelsIn = pDevice->capture.channels; - ma_copy_memory(data.channelMapIn, pDevice->capture.channelMap, sizeof(pDevice->capture.channelMap)); - data.shareMode = pDevice->capture.shareMode; - data.usingDefaultFormat = pDevice->capture.usingDefaultFormat; - data.usingDefaultChannels = pDevice->capture.usingDefaultChannels; - data.usingDefaultChannelMap = pDevice->capture.usingDefaultChannelMap; - } - - data.sampleRateIn = pDevice->sampleRate; - data.usingDefaultSampleRate = pDevice->usingDefaultSampleRate; - data.bufferSizeInFramesIn = pDevice->wasapi.originalBufferSizeInFrames; - data.bufferSizeInMillisecondsIn = pDevice->wasapi.originalBufferSizeInMilliseconds; - data.periodsIn = pDevice->wasapi.originalPeriods; - data.noAutoConvertSRC = pDevice->wasapi.noAutoConvertSRC; - data.noDefaultQualitySRC = pDevice->wasapi.noDefaultQualitySRC; - result = ma_device_init_internal__wasapi(pDevice->pContext, deviceType, NULL, &data); - if (result != MA_SUCCESS) { - return result; - } +*******************************************************************************/ +#ifdef MA_HAS_WASAPI +#if 0 +#if defined(_MSC_VER) + #pragma warning(push) + #pragma warning(disable:4091) /* 'typedef ': ignored on left of '' when no variable is declared */ +#endif +#include +#include +#if defined(_MSC_VER) + #pragma warning(pop) +#endif +#endif /* 0 */ + +/* Some compilers don't define VerifyVersionInfoW. Need to write this ourselves. */ +#define MA_WIN32_WINNT_VISTA 0x0600 +#define MA_VER_MINORVERSION 0x01 +#define MA_VER_MAJORVERSION 0x02 +#define MA_VER_SERVICEPACKMAJOR 0x20 +#define MA_VER_GREATER_EQUAL 0x03 + +typedef struct { + DWORD dwOSVersionInfoSize; + DWORD dwMajorVersion; + DWORD dwMinorVersion; + DWORD dwBuildNumber; + DWORD dwPlatformId; + WCHAR szCSDVersion[128]; + WORD wServicePackMajor; + WORD wServicePackMinor; + WORD wSuiteMask; + BYTE wProductType; + BYTE wReserved; +} ma_OSVERSIONINFOEXW; + +typedef BOOL (WINAPI * ma_PFNVerifyVersionInfoW) (ma_OSVERSIONINFOEXW* lpVersionInfo, DWORD dwTypeMask, DWORDLONG dwlConditionMask); +typedef ULONGLONG (WINAPI * ma_PFNVerSetConditionMask)(ULONGLONG dwlConditionMask, DWORD dwTypeBitMask, BYTE dwConditionMask); + + +#ifndef PROPERTYKEY_DEFINED +#define PROPERTYKEY_DEFINED +typedef struct +{ + GUID fmtid; + DWORD pid; +} PROPERTYKEY; +#endif + +/* Some compilers don't define PropVariantInit(). We just do this ourselves since it's just a memset(). */ +static MA_INLINE void ma_PropVariantInit(PROPVARIANT* pProp) +{ + MA_ZERO_OBJECT(pProp); +} - /* At this point we have some new objects ready to go. We need to uninitialize the previous ones and then set the new ones. */ - if (deviceType == ma_device_type_capture || deviceType == ma_device_type_loopback) { - if (pDevice->wasapi.pCaptureClient) { - ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient); - pDevice->wasapi.pCaptureClient = NULL; - } - if (pDevice->wasapi.pAudioClientCapture) { - ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); - pDevice->wasapi.pAudioClientCapture = NULL; - } +static const PROPERTYKEY MA_PKEY_Device_FriendlyName = {{0xA45C254E, 0xDF1C, 0x4EFD, {0x80, 0x20, 0x67, 0xD1, 0x46, 0xA8, 0x50, 0xE0}}, 14}; +static const PROPERTYKEY MA_PKEY_AudioEngine_DeviceFormat = {{0xF19F064D, 0x82C, 0x4E27, {0xBC, 0x73, 0x68, 0x82, 0xA1, 0xBB, 0x8E, 0x4C}}, 0}; - pDevice->wasapi.pAudioClientCapture = data.pAudioClient; - pDevice->wasapi.pCaptureClient = data.pCaptureClient; +static const IID MA_IID_IUnknown = {0x00000000, 0x0000, 0x0000, {0xC0, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x46}}; /* 00000000-0000-0000-C000-000000000046 */ +static const IID MA_IID_IAgileObject = {0x94EA2B94, 0xE9CC, 0x49E0, {0xC0, 0xFF, 0xEE, 0x64, 0xCA, 0x8F, 0x5B, 0x90}}; /* 94EA2B94-E9CC-49E0-C0FF-EE64CA8F5B90 */ - pDevice->capture.internalFormat = data.formatOut; - pDevice->capture.internalChannels = data.channelsOut; - pDevice->capture.internalSampleRate = data.sampleRateOut; - ma_copy_memory(pDevice->capture.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); - pDevice->capture.internalBufferSizeInFrames = data.bufferSizeInFramesOut; - pDevice->capture.internalPeriods = data.periodsOut; - ma_strcpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), data.deviceName); +static const IID MA_IID_IAudioClient = {0x1CB9AD4C, 0xDBFA, 0x4C32, {0xB1, 0x78, 0xC2, 0xF5, 0x68, 0xA7, 0x03, 0xB2}}; /* 1CB9AD4C-DBFA-4C32-B178-C2F568A703B2 = __uuidof(IAudioClient) */ +static const IID MA_IID_IAudioClient2 = {0x726778CD, 0xF60A, 0x4EDA, {0x82, 0xDE, 0xE4, 0x76, 0x10, 0xCD, 0x78, 0xAA}}; /* 726778CD-F60A-4EDA-82DE-E47610CD78AA = __uuidof(IAudioClient2) */ +static const IID MA_IID_IAudioClient3 = {0x7ED4EE07, 0x8E67, 0x4CD4, {0x8C, 0x1A, 0x2B, 0x7A, 0x59, 0x87, 0xAD, 0x42}}; /* 7ED4EE07-8E67-4CD4-8C1A-2B7A5987AD42 = __uuidof(IAudioClient3) */ +static const IID MA_IID_IAudioRenderClient = {0xF294ACFC, 0x3146, 0x4483, {0xA7, 0xBF, 0xAD, 0xDC, 0xA7, 0xC2, 0x60, 0xE2}}; /* F294ACFC-3146-4483-A7BF-ADDCA7C260E2 = __uuidof(IAudioRenderClient) */ +static const IID MA_IID_IAudioCaptureClient = {0xC8ADBD64, 0xE71E, 0x48A0, {0xA4, 0xDE, 0x18, 0x5C, 0x39, 0x5C, 0xD3, 0x17}}; /* C8ADBD64-E71E-48A0-A4DE-185C395CD317 = __uuidof(IAudioCaptureClient) */ +static const IID MA_IID_IMMNotificationClient = {0x7991EEC9, 0x7E89, 0x4D85, {0x83, 0x90, 0x6C, 0x70, 0x3C, 0xEC, 0x60, 0xC0}}; /* 7991EEC9-7E89-4D85-8390-6C703CEC60C0 = __uuidof(IMMNotificationClient) */ +#ifndef MA_WIN32_DESKTOP +static const IID MA_IID_DEVINTERFACE_AUDIO_RENDER = {0xE6327CAD, 0xDCEC, 0x4949, {0xAE, 0x8A, 0x99, 0x1E, 0x97, 0x6A, 0x79, 0xD2}}; /* E6327CAD-DCEC-4949-AE8A-991E976A79D2 */ +static const IID MA_IID_DEVINTERFACE_AUDIO_CAPTURE = {0x2EEF81BE, 0x33FA, 0x4800, {0x96, 0x70, 0x1C, 0xD4, 0x74, 0x97, 0x2C, 0x3F}}; /* 2EEF81BE-33FA-4800-9670-1CD474972C3F */ +static const IID MA_IID_IActivateAudioInterfaceCompletionHandler = {0x41D949AB, 0x9862, 0x444A, {0x80, 0xF6, 0xC2, 0x61, 0x33, 0x4D, 0xA5, 0xEB}}; /* 41D949AB-9862-444A-80F6-C261334DA5EB */ +#endif - ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, pDevice->wasapi.hEventCapture); +static const IID MA_CLSID_MMDeviceEnumerator_Instance = {0xBCDE0395, 0xE52F, 0x467C, {0x8E, 0x3D, 0xC4, 0x57, 0x92, 0x91, 0x69, 0x2E}}; /* BCDE0395-E52F-467C-8E3D-C4579291692E = __uuidof(MMDeviceEnumerator) */ +static const IID MA_IID_IMMDeviceEnumerator_Instance = {0xA95664D2, 0x9614, 0x4F35, {0xA7, 0x46, 0xDE, 0x8D, 0xB6, 0x36, 0x17, 0xE6}}; /* A95664D2-9614-4F35-A746-DE8DB63617E6 = __uuidof(IMMDeviceEnumerator) */ +#ifdef __cplusplus +#define MA_CLSID_MMDeviceEnumerator MA_CLSID_MMDeviceEnumerator_Instance +#define MA_IID_IMMDeviceEnumerator MA_IID_IMMDeviceEnumerator_Instance +#else +#define MA_CLSID_MMDeviceEnumerator &MA_CLSID_MMDeviceEnumerator_Instance +#define MA_IID_IMMDeviceEnumerator &MA_IID_IMMDeviceEnumerator_Instance +#endif - pDevice->wasapi.periodSizeInFramesCapture = data.periodSizeInFramesOut; - ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, &pDevice->wasapi.actualBufferSizeInFramesCapture); +typedef struct ma_IUnknown ma_IUnknown; +#ifdef MA_WIN32_DESKTOP +#define MA_MM_DEVICE_STATE_ACTIVE 1 +#define MA_MM_DEVICE_STATE_DISABLED 2 +#define MA_MM_DEVICE_STATE_NOTPRESENT 4 +#define MA_MM_DEVICE_STATE_UNPLUGGED 8 - /* The device may be in a started state. If so we need to immediately restart it. */ - if (pDevice->wasapi.isStartedCapture) { - HRESULT hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); - if (FAILED(hr)) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal capture device after reinitialization.", MA_FAILED_TO_START_BACKEND_DEVICE); - } - } - } +typedef struct ma_IMMDeviceEnumerator ma_IMMDeviceEnumerator; +typedef struct ma_IMMDeviceCollection ma_IMMDeviceCollection; +typedef struct ma_IMMDevice ma_IMMDevice; +#else +typedef struct ma_IActivateAudioInterfaceCompletionHandler ma_IActivateAudioInterfaceCompletionHandler; +typedef struct ma_IActivateAudioInterfaceAsyncOperation ma_IActivateAudioInterfaceAsyncOperation; +#endif +typedef struct ma_IPropertyStore ma_IPropertyStore; +typedef struct ma_IAudioClient ma_IAudioClient; +typedef struct ma_IAudioClient2 ma_IAudioClient2; +typedef struct ma_IAudioClient3 ma_IAudioClient3; +typedef struct ma_IAudioRenderClient ma_IAudioRenderClient; +typedef struct ma_IAudioCaptureClient ma_IAudioCaptureClient; - if (deviceType == ma_device_type_playback) { - if (pDevice->wasapi.pRenderClient) { - ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient); - pDevice->wasapi.pRenderClient = NULL; - } +typedef ma_int64 MA_REFERENCE_TIME; - if (pDevice->wasapi.pAudioClientPlayback) { - ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); - pDevice->wasapi.pAudioClientPlayback = NULL; - } +#define MA_AUDCLNT_STREAMFLAGS_CROSSPROCESS 0x00010000 +#define MA_AUDCLNT_STREAMFLAGS_LOOPBACK 0x00020000 +#define MA_AUDCLNT_STREAMFLAGS_EVENTCALLBACK 0x00040000 +#define MA_AUDCLNT_STREAMFLAGS_NOPERSIST 0x00080000 +#define MA_AUDCLNT_STREAMFLAGS_RATEADJUST 0x00100000 +#define MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY 0x08000000 +#define MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM 0x80000000 +#define MA_AUDCLNT_SESSIONFLAGS_EXPIREWHENUNOWNED 0x10000000 +#define MA_AUDCLNT_SESSIONFLAGS_DISPLAY_HIDE 0x20000000 +#define MA_AUDCLNT_SESSIONFLAGS_DISPLAY_HIDEWHENEXPIRED 0x40000000 - pDevice->wasapi.pAudioClientPlayback = data.pAudioClient; - pDevice->wasapi.pRenderClient = data.pRenderClient; +/* We only care about a few error codes. */ +#define MA_AUDCLNT_E_INVALID_DEVICE_PERIOD (-2004287456) +#define MA_AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED (-2004287463) +#define MA_AUDCLNT_S_BUFFER_EMPTY (143196161) +#define MA_AUDCLNT_E_DEVICE_IN_USE (-2004287478) - pDevice->playback.internalFormat = data.formatOut; - pDevice->playback.internalChannels = data.channelsOut; - pDevice->playback.internalSampleRate = data.sampleRateOut; - ma_copy_memory(pDevice->playback.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); - pDevice->playback.internalBufferSizeInFrames = data.bufferSizeInFramesOut; - pDevice->playback.internalPeriods = data.periodsOut; - ma_strcpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), data.deviceName); +/* Buffer flags. */ +#define MA_AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY 1 +#define MA_AUDCLNT_BUFFERFLAGS_SILENT 2 +#define MA_AUDCLNT_BUFFERFLAGS_TIMESTAMP_ERROR 4 - ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, pDevice->wasapi.hEventPlayback); +typedef enum +{ + ma_eRender = 0, + ma_eCapture = 1, + ma_eAll = 2 +} ma_EDataFlow; - pDevice->wasapi.periodSizeInFramesPlayback = data.periodSizeInFramesOut; - ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &pDevice->wasapi.actualBufferSizeInFramesPlayback); +typedef enum +{ + ma_eConsole = 0, + ma_eMultimedia = 1, + ma_eCommunications = 2 +} ma_ERole; - /* The device may be in a started state. If so we need to immediately restart it. */ - if (pDevice->wasapi.isStartedPlayback) { - HRESULT hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); - if (FAILED(hr)) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal playback device after reinitialization.", MA_FAILED_TO_START_BACKEND_DEVICE); - } - } - } +typedef enum +{ + MA_AUDCLNT_SHAREMODE_SHARED, + MA_AUDCLNT_SHAREMODE_EXCLUSIVE +} MA_AUDCLNT_SHAREMODE; - return MA_SUCCESS; -} +typedef enum +{ + MA_AudioCategory_Other = 0 /* <-- miniaudio is only caring about Other. */ +} MA_AUDIO_STREAM_CATEGORY; -ma_result ma_device_init__wasapi(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) +typedef struct { - ma_result result = MA_SUCCESS; + UINT32 cbSize; + BOOL bIsOffload; + MA_AUDIO_STREAM_CATEGORY eCategory; +} ma_AudioClientProperties; - (void)pContext; +/* IUnknown */ +typedef struct +{ + /* IUnknown */ + HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IUnknown* pThis, const IID* const riid, void** ppObject); + ULONG (STDMETHODCALLTYPE * AddRef) (ma_IUnknown* pThis); + ULONG (STDMETHODCALLTYPE * Release) (ma_IUnknown* pThis); +} ma_IUnknownVtbl; +struct ma_IUnknown +{ + ma_IUnknownVtbl* lpVtbl; +}; +static MA_INLINE HRESULT ma_IUnknown_QueryInterface(ma_IUnknown* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } +static MA_INLINE ULONG ma_IUnknown_AddRef(ma_IUnknown* pThis) { return pThis->lpVtbl->AddRef(pThis); } +static MA_INLINE ULONG ma_IUnknown_Release(ma_IUnknown* pThis) { return pThis->lpVtbl->Release(pThis); } - ma_assert(pContext != NULL); - ma_assert(pDevice != NULL); +#ifdef MA_WIN32_DESKTOP + /* IMMNotificationClient */ + typedef struct + { + /* IUnknown */ + HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMNotificationClient* pThis, const IID* const riid, void** ppObject); + ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMNotificationClient* pThis); + ULONG (STDMETHODCALLTYPE * Release) (ma_IMMNotificationClient* pThis); - ma_zero_object(&pDevice->wasapi); - pDevice->wasapi.originalBufferSizeInFrames = pConfig->bufferSizeInFrames; - pDevice->wasapi.originalBufferSizeInMilliseconds = pConfig->bufferSizeInMilliseconds; - pDevice->wasapi.originalPeriods = pConfig->periods; - pDevice->wasapi.noAutoConvertSRC = pDevice->wasapi.noAutoConvertSRC; - pDevice->wasapi.noDefaultQualitySRC = pDevice->wasapi.noDefaultQualitySRC; + /* IMMNotificationClient */ + HRESULT (STDMETHODCALLTYPE * OnDeviceStateChanged) (ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID, DWORD dwNewState); + HRESULT (STDMETHODCALLTYPE * OnDeviceAdded) (ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID); + HRESULT (STDMETHODCALLTYPE * OnDeviceRemoved) (ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID); + HRESULT (STDMETHODCALLTYPE * OnDefaultDeviceChanged)(ma_IMMNotificationClient* pThis, ma_EDataFlow dataFlow, ma_ERole role, LPCWSTR pDefaultDeviceID); + HRESULT (STDMETHODCALLTYPE * OnPropertyValueChanged)(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID, const PROPERTYKEY key); + } ma_IMMNotificationClientVtbl; - /* Exclusive mode is not allowed with loopback. */ - if (pConfig->deviceType == ma_device_type_loopback && pConfig->playback.shareMode == ma_share_mode_exclusive) { - return MA_INVALID_DEVICE_CONFIG; - } + /* IMMDeviceEnumerator */ + typedef struct + { + /* IUnknown */ + HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMDeviceEnumerator* pThis, const IID* const riid, void** ppObject); + ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMDeviceEnumerator* pThis); + ULONG (STDMETHODCALLTYPE * Release) (ma_IMMDeviceEnumerator* pThis); - if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex || pConfig->deviceType == ma_device_type_loopback) { - ma_device_init_internal_data__wasapi data; - data.formatIn = pConfig->capture.format; - data.channelsIn = pConfig->capture.channels; - data.sampleRateIn = pConfig->sampleRate; - ma_copy_memory(data.channelMapIn, pConfig->capture.channelMap, sizeof(pConfig->capture.channelMap)); - data.usingDefaultFormat = pDevice->capture.usingDefaultFormat; - data.usingDefaultChannels = pDevice->capture.usingDefaultChannels; - data.usingDefaultSampleRate = pDevice->usingDefaultSampleRate; - data.usingDefaultChannelMap = pDevice->capture.usingDefaultChannelMap; - data.shareMode = pConfig->capture.shareMode; - data.bufferSizeInFramesIn = pConfig->bufferSizeInFrames; - data.bufferSizeInMillisecondsIn = pConfig->bufferSizeInMilliseconds; - data.periodsIn = pConfig->periods; - data.noAutoConvertSRC = pConfig->wasapi.noAutoConvertSRC; - data.noDefaultQualitySRC = pConfig->wasapi.noDefaultQualitySRC; + /* IMMDeviceEnumerator */ + HRESULT (STDMETHODCALLTYPE * EnumAudioEndpoints) (ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, DWORD dwStateMask, ma_IMMDeviceCollection** ppDevices); + HRESULT (STDMETHODCALLTYPE * GetDefaultAudioEndpoint) (ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, ma_ERole role, ma_IMMDevice** ppEndpoint); + HRESULT (STDMETHODCALLTYPE * GetDevice) (ma_IMMDeviceEnumerator* pThis, LPCWSTR pID, ma_IMMDevice** ppDevice); + HRESULT (STDMETHODCALLTYPE * RegisterEndpointNotificationCallback) (ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient); + HRESULT (STDMETHODCALLTYPE * UnregisterEndpointNotificationCallback)(ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient); + } ma_IMMDeviceEnumeratorVtbl; + struct ma_IMMDeviceEnumerator + { + ma_IMMDeviceEnumeratorVtbl* lpVtbl; + }; + static MA_INLINE HRESULT ma_IMMDeviceEnumerator_QueryInterface(ma_IMMDeviceEnumerator* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } + static MA_INLINE ULONG ma_IMMDeviceEnumerator_AddRef(ma_IMMDeviceEnumerator* pThis) { return pThis->lpVtbl->AddRef(pThis); } + static MA_INLINE ULONG ma_IMMDeviceEnumerator_Release(ma_IMMDeviceEnumerator* pThis) { return pThis->lpVtbl->Release(pThis); } + static MA_INLINE HRESULT ma_IMMDeviceEnumerator_EnumAudioEndpoints(ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, DWORD dwStateMask, ma_IMMDeviceCollection** ppDevices) { return pThis->lpVtbl->EnumAudioEndpoints(pThis, dataFlow, dwStateMask, ppDevices); } + static MA_INLINE HRESULT ma_IMMDeviceEnumerator_GetDefaultAudioEndpoint(ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, ma_ERole role, ma_IMMDevice** ppEndpoint) { return pThis->lpVtbl->GetDefaultAudioEndpoint(pThis, dataFlow, role, ppEndpoint); } + static MA_INLINE HRESULT ma_IMMDeviceEnumerator_GetDevice(ma_IMMDeviceEnumerator* pThis, LPCWSTR pID, ma_IMMDevice** ppDevice) { return pThis->lpVtbl->GetDevice(pThis, pID, ppDevice); } + static MA_INLINE HRESULT ma_IMMDeviceEnumerator_RegisterEndpointNotificationCallback(ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient) { return pThis->lpVtbl->RegisterEndpointNotificationCallback(pThis, pClient); } + static MA_INLINE HRESULT ma_IMMDeviceEnumerator_UnregisterEndpointNotificationCallback(ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient) { return pThis->lpVtbl->UnregisterEndpointNotificationCallback(pThis, pClient); } - result = ma_device_init_internal__wasapi(pDevice->pContext, (pConfig->deviceType == ma_device_type_loopback) ? ma_device_type_loopback : ma_device_type_capture, pConfig->capture.pDeviceID, &data); - if (result != MA_SUCCESS) { - return result; - } - pDevice->wasapi.pAudioClientCapture = data.pAudioClient; - pDevice->wasapi.pCaptureClient = data.pCaptureClient; + /* IMMDeviceCollection */ + typedef struct + { + /* IUnknown */ + HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMDeviceCollection* pThis, const IID* const riid, void** ppObject); + ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMDeviceCollection* pThis); + ULONG (STDMETHODCALLTYPE * Release) (ma_IMMDeviceCollection* pThis); - pDevice->capture.internalFormat = data.formatOut; - pDevice->capture.internalChannels = data.channelsOut; - pDevice->capture.internalSampleRate = data.sampleRateOut; - ma_copy_memory(pDevice->capture.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); - pDevice->capture.internalBufferSizeInFrames = data.bufferSizeInFramesOut; - pDevice->capture.internalPeriods = data.periodsOut; - ma_strcpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), data.deviceName); + /* IMMDeviceCollection */ + HRESULT (STDMETHODCALLTYPE * GetCount)(ma_IMMDeviceCollection* pThis, UINT* pDevices); + HRESULT (STDMETHODCALLTYPE * Item) (ma_IMMDeviceCollection* pThis, UINT nDevice, ma_IMMDevice** ppDevice); + } ma_IMMDeviceCollectionVtbl; + struct ma_IMMDeviceCollection + { + ma_IMMDeviceCollectionVtbl* lpVtbl; + }; + static MA_INLINE HRESULT ma_IMMDeviceCollection_QueryInterface(ma_IMMDeviceCollection* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } + static MA_INLINE ULONG ma_IMMDeviceCollection_AddRef(ma_IMMDeviceCollection* pThis) { return pThis->lpVtbl->AddRef(pThis); } + static MA_INLINE ULONG ma_IMMDeviceCollection_Release(ma_IMMDeviceCollection* pThis) { return pThis->lpVtbl->Release(pThis); } + static MA_INLINE HRESULT ma_IMMDeviceCollection_GetCount(ma_IMMDeviceCollection* pThis, UINT* pDevices) { return pThis->lpVtbl->GetCount(pThis, pDevices); } + static MA_INLINE HRESULT ma_IMMDeviceCollection_Item(ma_IMMDeviceCollection* pThis, UINT nDevice, ma_IMMDevice** ppDevice) { return pThis->lpVtbl->Item(pThis, nDevice, ppDevice); } - /* - The event for capture needs to be manual reset for the same reason as playback. We keep the initial state set to unsignaled, - however, because we want to block until we actually have something for the first call to ma_device_read(). - */ - pDevice->wasapi.hEventCapture = CreateEventA(NULL, FALSE, FALSE, NULL); /* Auto reset, unsignaled by default. */ - if (pDevice->wasapi.hEventCapture == NULL) { - if (pDevice->wasapi.pCaptureClient != NULL) { - ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient); - pDevice->wasapi.pCaptureClient = NULL; - } - if (pDevice->wasapi.pAudioClientCapture != NULL) { - ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); - pDevice->wasapi.pAudioClientCapture = NULL; - } - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create event for capture.", MA_FAILED_TO_CREATE_EVENT); - } - ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, pDevice->wasapi.hEventCapture); + /* IMMDevice */ + typedef struct + { + /* IUnknown */ + HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMDevice* pThis, const IID* const riid, void** ppObject); + ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMDevice* pThis); + ULONG (STDMETHODCALLTYPE * Release) (ma_IMMDevice* pThis); + + /* IMMDevice */ + HRESULT (STDMETHODCALLTYPE * Activate) (ma_IMMDevice* pThis, const IID* const iid, DWORD dwClsCtx, PROPVARIANT* pActivationParams, void** ppInterface); + HRESULT (STDMETHODCALLTYPE * OpenPropertyStore)(ma_IMMDevice* pThis, DWORD stgmAccess, ma_IPropertyStore** ppProperties); + HRESULT (STDMETHODCALLTYPE * GetId) (ma_IMMDevice* pThis, LPWSTR *pID); + HRESULT (STDMETHODCALLTYPE * GetState) (ma_IMMDevice* pThis, DWORD *pState); + } ma_IMMDeviceVtbl; + struct ma_IMMDevice + { + ma_IMMDeviceVtbl* lpVtbl; + }; + static MA_INLINE HRESULT ma_IMMDevice_QueryInterface(ma_IMMDevice* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } + static MA_INLINE ULONG ma_IMMDevice_AddRef(ma_IMMDevice* pThis) { return pThis->lpVtbl->AddRef(pThis); } + static MA_INLINE ULONG ma_IMMDevice_Release(ma_IMMDevice* pThis) { return pThis->lpVtbl->Release(pThis); } + static MA_INLINE HRESULT ma_IMMDevice_Activate(ma_IMMDevice* pThis, const IID* const iid, DWORD dwClsCtx, PROPVARIANT* pActivationParams, void** ppInterface) { return pThis->lpVtbl->Activate(pThis, iid, dwClsCtx, pActivationParams, ppInterface); } + static MA_INLINE HRESULT ma_IMMDevice_OpenPropertyStore(ma_IMMDevice* pThis, DWORD stgmAccess, ma_IPropertyStore** ppProperties) { return pThis->lpVtbl->OpenPropertyStore(pThis, stgmAccess, ppProperties); } + static MA_INLINE HRESULT ma_IMMDevice_GetId(ma_IMMDevice* pThis, LPWSTR *pID) { return pThis->lpVtbl->GetId(pThis, pID); } + static MA_INLINE HRESULT ma_IMMDevice_GetState(ma_IMMDevice* pThis, DWORD *pState) { return pThis->lpVtbl->GetState(pThis, pState); } +#else + /* IActivateAudioInterfaceAsyncOperation */ + typedef struct + { + /* IUnknown */ + HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IActivateAudioInterfaceAsyncOperation* pThis, const IID* const riid, void** ppObject); + ULONG (STDMETHODCALLTYPE * AddRef) (ma_IActivateAudioInterfaceAsyncOperation* pThis); + ULONG (STDMETHODCALLTYPE * Release) (ma_IActivateAudioInterfaceAsyncOperation* pThis); + + /* IActivateAudioInterfaceAsyncOperation */ + HRESULT (STDMETHODCALLTYPE * GetActivateResult)(ma_IActivateAudioInterfaceAsyncOperation* pThis, HRESULT *pActivateResult, ma_IUnknown** ppActivatedInterface); + } ma_IActivateAudioInterfaceAsyncOperationVtbl; + struct ma_IActivateAudioInterfaceAsyncOperation + { + ma_IActivateAudioInterfaceAsyncOperationVtbl* lpVtbl; + }; + static MA_INLINE HRESULT ma_IActivateAudioInterfaceAsyncOperation_QueryInterface(ma_IActivateAudioInterfaceAsyncOperation* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } + static MA_INLINE ULONG ma_IActivateAudioInterfaceAsyncOperation_AddRef(ma_IActivateAudioInterfaceAsyncOperation* pThis) { return pThis->lpVtbl->AddRef(pThis); } + static MA_INLINE ULONG ma_IActivateAudioInterfaceAsyncOperation_Release(ma_IActivateAudioInterfaceAsyncOperation* pThis) { return pThis->lpVtbl->Release(pThis); } + static MA_INLINE HRESULT ma_IActivateAudioInterfaceAsyncOperation_GetActivateResult(ma_IActivateAudioInterfaceAsyncOperation* pThis, HRESULT *pActivateResult, ma_IUnknown** ppActivatedInterface) { return pThis->lpVtbl->GetActivateResult(pThis, pActivateResult, ppActivatedInterface); } +#endif - pDevice->wasapi.periodSizeInFramesCapture = data.periodSizeInFramesOut; - ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, &pDevice->wasapi.actualBufferSizeInFramesCapture); - } +/* IPropertyStore */ +typedef struct +{ + /* IUnknown */ + HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IPropertyStore* pThis, const IID* const riid, void** ppObject); + ULONG (STDMETHODCALLTYPE * AddRef) (ma_IPropertyStore* pThis); + ULONG (STDMETHODCALLTYPE * Release) (ma_IPropertyStore* pThis); - if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { - ma_device_init_internal_data__wasapi data; - data.formatIn = pConfig->playback.format; - data.channelsIn = pConfig->playback.channels; - data.sampleRateIn = pConfig->sampleRate; - ma_copy_memory(data.channelMapIn, pConfig->playback.channelMap, sizeof(pConfig->playback.channelMap)); - data.usingDefaultFormat = pDevice->playback.usingDefaultFormat; - data.usingDefaultChannels = pDevice->playback.usingDefaultChannels; - data.usingDefaultSampleRate = pDevice->usingDefaultSampleRate; - data.usingDefaultChannelMap = pDevice->playback.usingDefaultChannelMap; - data.shareMode = pConfig->playback.shareMode; - data.bufferSizeInFramesIn = pConfig->bufferSizeInFrames; - data.bufferSizeInMillisecondsIn = pConfig->bufferSizeInMilliseconds; - data.periodsIn = pConfig->periods; - data.noAutoConvertSRC = pConfig->wasapi.noAutoConvertSRC; - data.noDefaultQualitySRC = pConfig->wasapi.noDefaultQualitySRC; + /* IPropertyStore */ + HRESULT (STDMETHODCALLTYPE * GetCount)(ma_IPropertyStore* pThis, DWORD* pPropCount); + HRESULT (STDMETHODCALLTYPE * GetAt) (ma_IPropertyStore* pThis, DWORD propIndex, PROPERTYKEY* pPropKey); + HRESULT (STDMETHODCALLTYPE * GetValue)(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, PROPVARIANT* pPropVar); + HRESULT (STDMETHODCALLTYPE * SetValue)(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, const PROPVARIANT* const pPropVar); + HRESULT (STDMETHODCALLTYPE * Commit) (ma_IPropertyStore* pThis); +} ma_IPropertyStoreVtbl; +struct ma_IPropertyStore +{ + ma_IPropertyStoreVtbl* lpVtbl; +}; +static MA_INLINE HRESULT ma_IPropertyStore_QueryInterface(ma_IPropertyStore* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } +static MA_INLINE ULONG ma_IPropertyStore_AddRef(ma_IPropertyStore* pThis) { return pThis->lpVtbl->AddRef(pThis); } +static MA_INLINE ULONG ma_IPropertyStore_Release(ma_IPropertyStore* pThis) { return pThis->lpVtbl->Release(pThis); } +static MA_INLINE HRESULT ma_IPropertyStore_GetCount(ma_IPropertyStore* pThis, DWORD* pPropCount) { return pThis->lpVtbl->GetCount(pThis, pPropCount); } +static MA_INLINE HRESULT ma_IPropertyStore_GetAt(ma_IPropertyStore* pThis, DWORD propIndex, PROPERTYKEY* pPropKey) { return pThis->lpVtbl->GetAt(pThis, propIndex, pPropKey); } +static MA_INLINE HRESULT ma_IPropertyStore_GetValue(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, PROPVARIANT* pPropVar) { return pThis->lpVtbl->GetValue(pThis, pKey, pPropVar); } +static MA_INLINE HRESULT ma_IPropertyStore_SetValue(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, const PROPVARIANT* const pPropVar) { return pThis->lpVtbl->SetValue(pThis, pKey, pPropVar); } +static MA_INLINE HRESULT ma_IPropertyStore_Commit(ma_IPropertyStore* pThis) { return pThis->lpVtbl->Commit(pThis); } - result = ma_device_init_internal__wasapi(pDevice->pContext, ma_device_type_playback, pConfig->playback.pDeviceID, &data); - if (result != MA_SUCCESS) { - if (pConfig->deviceType == ma_device_type_duplex) { - if (pDevice->wasapi.pCaptureClient != NULL) { - ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient); - pDevice->wasapi.pCaptureClient = NULL; - } - if (pDevice->wasapi.pAudioClientCapture != NULL) { - ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); - pDevice->wasapi.pAudioClientCapture = NULL; - } - CloseHandle(pDevice->wasapi.hEventCapture); - pDevice->wasapi.hEventCapture = NULL; - } - return result; - } +/* IAudioClient */ +typedef struct +{ + /* IUnknown */ + HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioClient* pThis, const IID* const riid, void** ppObject); + ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioClient* pThis); + ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioClient* pThis); - pDevice->wasapi.pAudioClientPlayback = data.pAudioClient; - pDevice->wasapi.pRenderClient = data.pRenderClient; + /* IAudioClient */ + HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid); + HRESULT (STDMETHODCALLTYPE * GetBufferSize) (ma_IAudioClient* pThis, ma_uint32* pNumBufferFrames); + HRESULT (STDMETHODCALLTYPE * GetStreamLatency) (ma_IAudioClient* pThis, MA_REFERENCE_TIME* pLatency); + HRESULT (STDMETHODCALLTYPE * GetCurrentPadding)(ma_IAudioClient* pThis, ma_uint32* pNumPaddingFrames); + HRESULT (STDMETHODCALLTYPE * IsFormatSupported)(ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch); + HRESULT (STDMETHODCALLTYPE * GetMixFormat) (ma_IAudioClient* pThis, WAVEFORMATEX** ppDeviceFormat); + HRESULT (STDMETHODCALLTYPE * GetDevicePeriod) (ma_IAudioClient* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod); + HRESULT (STDMETHODCALLTYPE * Start) (ma_IAudioClient* pThis); + HRESULT (STDMETHODCALLTYPE * Stop) (ma_IAudioClient* pThis); + HRESULT (STDMETHODCALLTYPE * Reset) (ma_IAudioClient* pThis); + HRESULT (STDMETHODCALLTYPE * SetEventHandle) (ma_IAudioClient* pThis, HANDLE eventHandle); + HRESULT (STDMETHODCALLTYPE * GetService) (ma_IAudioClient* pThis, const IID* const riid, void** pp); +} ma_IAudioClientVtbl; +struct ma_IAudioClient +{ + ma_IAudioClientVtbl* lpVtbl; +}; +static MA_INLINE HRESULT ma_IAudioClient_QueryInterface(ma_IAudioClient* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } +static MA_INLINE ULONG ma_IAudioClient_AddRef(ma_IAudioClient* pThis) { return pThis->lpVtbl->AddRef(pThis); } +static MA_INLINE ULONG ma_IAudioClient_Release(ma_IAudioClient* pThis) { return pThis->lpVtbl->Release(pThis); } +static MA_INLINE HRESULT ma_IAudioClient_Initialize(ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid) { return pThis->lpVtbl->Initialize(pThis, shareMode, streamFlags, bufferDuration, periodicity, pFormat, pAudioSessionGuid); } +static MA_INLINE HRESULT ma_IAudioClient_GetBufferSize(ma_IAudioClient* pThis, ma_uint32* pNumBufferFrames) { return pThis->lpVtbl->GetBufferSize(pThis, pNumBufferFrames); } +static MA_INLINE HRESULT ma_IAudioClient_GetStreamLatency(ma_IAudioClient* pThis, MA_REFERENCE_TIME* pLatency) { return pThis->lpVtbl->GetStreamLatency(pThis, pLatency); } +static MA_INLINE HRESULT ma_IAudioClient_GetCurrentPadding(ma_IAudioClient* pThis, ma_uint32* pNumPaddingFrames) { return pThis->lpVtbl->GetCurrentPadding(pThis, pNumPaddingFrames); } +static MA_INLINE HRESULT ma_IAudioClient_IsFormatSupported(ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch) { return pThis->lpVtbl->IsFormatSupported(pThis, shareMode, pFormat, ppClosestMatch); } +static MA_INLINE HRESULT ma_IAudioClient_GetMixFormat(ma_IAudioClient* pThis, WAVEFORMATEX** ppDeviceFormat) { return pThis->lpVtbl->GetMixFormat(pThis, ppDeviceFormat); } +static MA_INLINE HRESULT ma_IAudioClient_GetDevicePeriod(ma_IAudioClient* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod) { return pThis->lpVtbl->GetDevicePeriod(pThis, pDefaultDevicePeriod, pMinimumDevicePeriod); } +static MA_INLINE HRESULT ma_IAudioClient_Start(ma_IAudioClient* pThis) { return pThis->lpVtbl->Start(pThis); } +static MA_INLINE HRESULT ma_IAudioClient_Stop(ma_IAudioClient* pThis) { return pThis->lpVtbl->Stop(pThis); } +static MA_INLINE HRESULT ma_IAudioClient_Reset(ma_IAudioClient* pThis) { return pThis->lpVtbl->Reset(pThis); } +static MA_INLINE HRESULT ma_IAudioClient_SetEventHandle(ma_IAudioClient* pThis, HANDLE eventHandle) { return pThis->lpVtbl->SetEventHandle(pThis, eventHandle); } +static MA_INLINE HRESULT ma_IAudioClient_GetService(ma_IAudioClient* pThis, const IID* const riid, void** pp) { return pThis->lpVtbl->GetService(pThis, riid, pp); } - pDevice->playback.internalFormat = data.formatOut; - pDevice->playback.internalChannels = data.channelsOut; - pDevice->playback.internalSampleRate = data.sampleRateOut; - ma_copy_memory(pDevice->playback.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); - pDevice->playback.internalBufferSizeInFrames = data.bufferSizeInFramesOut; - pDevice->playback.internalPeriods = data.periodsOut; - ma_strcpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), data.deviceName); +/* IAudioClient2 */ +typedef struct +{ + /* IUnknown */ + HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioClient2* pThis, const IID* const riid, void** ppObject); + ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioClient2* pThis); + ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioClient2* pThis); - /* - The event for playback is needs to be manual reset because we want to explicitly control the fact that it becomes signalled - only after the whole available space has been filled, never before. + /* IAudioClient */ + HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid); + HRESULT (STDMETHODCALLTYPE * GetBufferSize) (ma_IAudioClient2* pThis, ma_uint32* pNumBufferFrames); + HRESULT (STDMETHODCALLTYPE * GetStreamLatency) (ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pLatency); + HRESULT (STDMETHODCALLTYPE * GetCurrentPadding)(ma_IAudioClient2* pThis, ma_uint32* pNumPaddingFrames); + HRESULT (STDMETHODCALLTYPE * IsFormatSupported)(ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch); + HRESULT (STDMETHODCALLTYPE * GetMixFormat) (ma_IAudioClient2* pThis, WAVEFORMATEX** ppDeviceFormat); + HRESULT (STDMETHODCALLTYPE * GetDevicePeriod) (ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod); + HRESULT (STDMETHODCALLTYPE * Start) (ma_IAudioClient2* pThis); + HRESULT (STDMETHODCALLTYPE * Stop) (ma_IAudioClient2* pThis); + HRESULT (STDMETHODCALLTYPE * Reset) (ma_IAudioClient2* pThis); + HRESULT (STDMETHODCALLTYPE * SetEventHandle) (ma_IAudioClient2* pThis, HANDLE eventHandle); + HRESULT (STDMETHODCALLTYPE * GetService) (ma_IAudioClient2* pThis, const IID* const riid, void** pp); - The playback event also needs to be initially set to a signaled state so that the first call to ma_device_write() is able - to get passed WaitForMultipleObjects(). - */ - pDevice->wasapi.hEventPlayback = CreateEventA(NULL, FALSE, TRUE, NULL); /* Auto reset, signaled by default. */ - if (pDevice->wasapi.hEventPlayback == NULL) { - if (pConfig->deviceType == ma_device_type_duplex) { - if (pDevice->wasapi.pCaptureClient != NULL) { - ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient); - pDevice->wasapi.pCaptureClient = NULL; - } - if (pDevice->wasapi.pAudioClientCapture != NULL) { - ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); - pDevice->wasapi.pAudioClientCapture = NULL; - } + /* IAudioClient2 */ + HRESULT (STDMETHODCALLTYPE * IsOffloadCapable) (ma_IAudioClient2* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable); + HRESULT (STDMETHODCALLTYPE * SetClientProperties)(ma_IAudioClient2* pThis, const ma_AudioClientProperties* pProperties); + HRESULT (STDMETHODCALLTYPE * GetBufferSizeLimits)(ma_IAudioClient2* pThis, const WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration); +} ma_IAudioClient2Vtbl; +struct ma_IAudioClient2 +{ + ma_IAudioClient2Vtbl* lpVtbl; +}; +static MA_INLINE HRESULT ma_IAudioClient2_QueryInterface(ma_IAudioClient2* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } +static MA_INLINE ULONG ma_IAudioClient2_AddRef(ma_IAudioClient2* pThis) { return pThis->lpVtbl->AddRef(pThis); } +static MA_INLINE ULONG ma_IAudioClient2_Release(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Release(pThis); } +static MA_INLINE HRESULT ma_IAudioClient2_Initialize(ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid) { return pThis->lpVtbl->Initialize(pThis, shareMode, streamFlags, bufferDuration, periodicity, pFormat, pAudioSessionGuid); } +static MA_INLINE HRESULT ma_IAudioClient2_GetBufferSize(ma_IAudioClient2* pThis, ma_uint32* pNumBufferFrames) { return pThis->lpVtbl->GetBufferSize(pThis, pNumBufferFrames); } +static MA_INLINE HRESULT ma_IAudioClient2_GetStreamLatency(ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pLatency) { return pThis->lpVtbl->GetStreamLatency(pThis, pLatency); } +static MA_INLINE HRESULT ma_IAudioClient2_GetCurrentPadding(ma_IAudioClient2* pThis, ma_uint32* pNumPaddingFrames) { return pThis->lpVtbl->GetCurrentPadding(pThis, pNumPaddingFrames); } +static MA_INLINE HRESULT ma_IAudioClient2_IsFormatSupported(ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch) { return pThis->lpVtbl->IsFormatSupported(pThis, shareMode, pFormat, ppClosestMatch); } +static MA_INLINE HRESULT ma_IAudioClient2_GetMixFormat(ma_IAudioClient2* pThis, WAVEFORMATEX** ppDeviceFormat) { return pThis->lpVtbl->GetMixFormat(pThis, ppDeviceFormat); } +static MA_INLINE HRESULT ma_IAudioClient2_GetDevicePeriod(ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod) { return pThis->lpVtbl->GetDevicePeriod(pThis, pDefaultDevicePeriod, pMinimumDevicePeriod); } +static MA_INLINE HRESULT ma_IAudioClient2_Start(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Start(pThis); } +static MA_INLINE HRESULT ma_IAudioClient2_Stop(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Stop(pThis); } +static MA_INLINE HRESULT ma_IAudioClient2_Reset(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Reset(pThis); } +static MA_INLINE HRESULT ma_IAudioClient2_SetEventHandle(ma_IAudioClient2* pThis, HANDLE eventHandle) { return pThis->lpVtbl->SetEventHandle(pThis, eventHandle); } +static MA_INLINE HRESULT ma_IAudioClient2_GetService(ma_IAudioClient2* pThis, const IID* const riid, void** pp) { return pThis->lpVtbl->GetService(pThis, riid, pp); } +static MA_INLINE HRESULT ma_IAudioClient2_IsOffloadCapable(ma_IAudioClient2* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable) { return pThis->lpVtbl->IsOffloadCapable(pThis, category, pOffloadCapable); } +static MA_INLINE HRESULT ma_IAudioClient2_SetClientProperties(ma_IAudioClient2* pThis, const ma_AudioClientProperties* pProperties) { return pThis->lpVtbl->SetClientProperties(pThis, pProperties); } +static MA_INLINE HRESULT ma_IAudioClient2_GetBufferSizeLimits(ma_IAudioClient2* pThis, const WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration) { return pThis->lpVtbl->GetBufferSizeLimits(pThis, pFormat, eventDriven, pMinBufferDuration, pMaxBufferDuration); } - CloseHandle(pDevice->wasapi.hEventCapture); - pDevice->wasapi.hEventCapture = NULL; - } - if (pDevice->wasapi.pRenderClient != NULL) { - ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient); - pDevice->wasapi.pRenderClient = NULL; - } - if (pDevice->wasapi.pAudioClientPlayback != NULL) { - ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); - pDevice->wasapi.pAudioClientPlayback = NULL; - } +/* IAudioClient3 */ +typedef struct +{ + /* IUnknown */ + HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioClient3* pThis, const IID* const riid, void** ppObject); + ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioClient3* pThis); + ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioClient3* pThis); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create event for playback.", MA_FAILED_TO_CREATE_EVENT); - } - ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, pDevice->wasapi.hEventPlayback); + /* IAudioClient */ + HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid); + HRESULT (STDMETHODCALLTYPE * GetBufferSize) (ma_IAudioClient3* pThis, ma_uint32* pNumBufferFrames); + HRESULT (STDMETHODCALLTYPE * GetStreamLatency) (ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pLatency); + HRESULT (STDMETHODCALLTYPE * GetCurrentPadding)(ma_IAudioClient3* pThis, ma_uint32* pNumPaddingFrames); + HRESULT (STDMETHODCALLTYPE * IsFormatSupported)(ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch); + HRESULT (STDMETHODCALLTYPE * GetMixFormat) (ma_IAudioClient3* pThis, WAVEFORMATEX** ppDeviceFormat); + HRESULT (STDMETHODCALLTYPE * GetDevicePeriod) (ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod); + HRESULT (STDMETHODCALLTYPE * Start) (ma_IAudioClient3* pThis); + HRESULT (STDMETHODCALLTYPE * Stop) (ma_IAudioClient3* pThis); + HRESULT (STDMETHODCALLTYPE * Reset) (ma_IAudioClient3* pThis); + HRESULT (STDMETHODCALLTYPE * SetEventHandle) (ma_IAudioClient3* pThis, HANDLE eventHandle); + HRESULT (STDMETHODCALLTYPE * GetService) (ma_IAudioClient3* pThis, const IID* const riid, void** pp); - pDevice->wasapi.periodSizeInFramesPlayback = data.periodSizeInFramesOut; - ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &pDevice->wasapi.actualBufferSizeInFramesPlayback); - } + /* IAudioClient2 */ + HRESULT (STDMETHODCALLTYPE * IsOffloadCapable) (ma_IAudioClient3* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable); + HRESULT (STDMETHODCALLTYPE * SetClientProperties)(ma_IAudioClient3* pThis, const ma_AudioClientProperties* pProperties); + HRESULT (STDMETHODCALLTYPE * GetBufferSizeLimits)(ma_IAudioClient3* pThis, const WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration); - /* - We need to get notifications of when the default device changes. We do this through a device enumerator by - registering a IMMNotificationClient with it. We only care about this if it's the default device. - */ -#ifdef MA_WIN32_DESKTOP - { - ma_IMMDeviceEnumerator* pDeviceEnumerator; - HRESULT hr = ma_CoCreateInstance(pContext, MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator); - if (FAILED(hr)) { - ma_device_uninit__wasapi(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create device enumerator.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } + /* IAudioClient3 */ + HRESULT (STDMETHODCALLTYPE * GetSharedModeEnginePeriod) (ma_IAudioClient3* pThis, const WAVEFORMATEX* pFormat, UINT32* pDefaultPeriodInFrames, UINT32* pFundamentalPeriodInFrames, UINT32* pMinPeriodInFrames, UINT32* pMaxPeriodInFrames); + HRESULT (STDMETHODCALLTYPE * GetCurrentSharedModeEnginePeriod)(ma_IAudioClient3* pThis, WAVEFORMATEX** ppFormat, UINT32* pCurrentPeriodInFrames); + HRESULT (STDMETHODCALLTYPE * InitializeSharedAudioStream) (ma_IAudioClient3* pThis, DWORD streamFlags, UINT32 periodInFrames, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid); +} ma_IAudioClient3Vtbl; +struct ma_IAudioClient3 +{ + ma_IAudioClient3Vtbl* lpVtbl; +}; +static MA_INLINE HRESULT ma_IAudioClient3_QueryInterface(ma_IAudioClient3* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } +static MA_INLINE ULONG ma_IAudioClient3_AddRef(ma_IAudioClient3* pThis) { return pThis->lpVtbl->AddRef(pThis); } +static MA_INLINE ULONG ma_IAudioClient3_Release(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Release(pThis); } +static MA_INLINE HRESULT ma_IAudioClient3_Initialize(ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid) { return pThis->lpVtbl->Initialize(pThis, shareMode, streamFlags, bufferDuration, periodicity, pFormat, pAudioSessionGuid); } +static MA_INLINE HRESULT ma_IAudioClient3_GetBufferSize(ma_IAudioClient3* pThis, ma_uint32* pNumBufferFrames) { return pThis->lpVtbl->GetBufferSize(pThis, pNumBufferFrames); } +static MA_INLINE HRESULT ma_IAudioClient3_GetStreamLatency(ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pLatency) { return pThis->lpVtbl->GetStreamLatency(pThis, pLatency); } +static MA_INLINE HRESULT ma_IAudioClient3_GetCurrentPadding(ma_IAudioClient3* pThis, ma_uint32* pNumPaddingFrames) { return pThis->lpVtbl->GetCurrentPadding(pThis, pNumPaddingFrames); } +static MA_INLINE HRESULT ma_IAudioClient3_IsFormatSupported(ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch) { return pThis->lpVtbl->IsFormatSupported(pThis, shareMode, pFormat, ppClosestMatch); } +static MA_INLINE HRESULT ma_IAudioClient3_GetMixFormat(ma_IAudioClient3* pThis, WAVEFORMATEX** ppDeviceFormat) { return pThis->lpVtbl->GetMixFormat(pThis, ppDeviceFormat); } +static MA_INLINE HRESULT ma_IAudioClient3_GetDevicePeriod(ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod) { return pThis->lpVtbl->GetDevicePeriod(pThis, pDefaultDevicePeriod, pMinimumDevicePeriod); } +static MA_INLINE HRESULT ma_IAudioClient3_Start(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Start(pThis); } +static MA_INLINE HRESULT ma_IAudioClient3_Stop(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Stop(pThis); } +static MA_INLINE HRESULT ma_IAudioClient3_Reset(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Reset(pThis); } +static MA_INLINE HRESULT ma_IAudioClient3_SetEventHandle(ma_IAudioClient3* pThis, HANDLE eventHandle) { return pThis->lpVtbl->SetEventHandle(pThis, eventHandle); } +static MA_INLINE HRESULT ma_IAudioClient3_GetService(ma_IAudioClient3* pThis, const IID* const riid, void** pp) { return pThis->lpVtbl->GetService(pThis, riid, pp); } +static MA_INLINE HRESULT ma_IAudioClient3_IsOffloadCapable(ma_IAudioClient3* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable) { return pThis->lpVtbl->IsOffloadCapable(pThis, category, pOffloadCapable); } +static MA_INLINE HRESULT ma_IAudioClient3_SetClientProperties(ma_IAudioClient3* pThis, const ma_AudioClientProperties* pProperties) { return pThis->lpVtbl->SetClientProperties(pThis, pProperties); } +static MA_INLINE HRESULT ma_IAudioClient3_GetBufferSizeLimits(ma_IAudioClient3* pThis, const WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration) { return pThis->lpVtbl->GetBufferSizeLimits(pThis, pFormat, eventDriven, pMinBufferDuration, pMaxBufferDuration); } +static MA_INLINE HRESULT ma_IAudioClient3_GetSharedModeEnginePeriod(ma_IAudioClient3* pThis, const WAVEFORMATEX* pFormat, UINT32* pDefaultPeriodInFrames, UINT32* pFundamentalPeriodInFrames, UINT32* pMinPeriodInFrames, UINT32* pMaxPeriodInFrames) { return pThis->lpVtbl->GetSharedModeEnginePeriod(pThis, pFormat, pDefaultPeriodInFrames, pFundamentalPeriodInFrames, pMinPeriodInFrames, pMaxPeriodInFrames); } +static MA_INLINE HRESULT ma_IAudioClient3_GetCurrentSharedModeEnginePeriod(ma_IAudioClient3* pThis, WAVEFORMATEX** ppFormat, UINT32* pCurrentPeriodInFrames) { return pThis->lpVtbl->GetCurrentSharedModeEnginePeriod(pThis, ppFormat, pCurrentPeriodInFrames); } +static MA_INLINE HRESULT ma_IAudioClient3_InitializeSharedAudioStream(ma_IAudioClient3* pThis, DWORD streamFlags, UINT32 periodInFrames, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGUID) { return pThis->lpVtbl->InitializeSharedAudioStream(pThis, streamFlags, periodInFrames, pFormat, pAudioSessionGUID); } - pDevice->wasapi.notificationClient.lpVtbl = (void*)&g_maNotificationCientVtbl; - pDevice->wasapi.notificationClient.counter = 1; - pDevice->wasapi.notificationClient.pDevice = pDevice; - hr = pDeviceEnumerator->lpVtbl->RegisterEndpointNotificationCallback(pDeviceEnumerator, &pDevice->wasapi.notificationClient); - if (SUCCEEDED(hr)) { - pDevice->wasapi.pDeviceEnumerator = (ma_ptr)pDeviceEnumerator; - } else { - /* Not the end of the world if we fail to register the notification callback. We just won't support automatic stream routing. */ - ma_IMMDeviceEnumerator_Release(pDeviceEnumerator); - } - } -#endif +/* IAudioRenderClient */ +typedef struct +{ + /* IUnknown */ + HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioRenderClient* pThis, const IID* const riid, void** ppObject); + ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioRenderClient* pThis); + ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioRenderClient* pThis); - ma_atomic_exchange_32(&pDevice->wasapi.isStartedCapture, MA_FALSE); - ma_atomic_exchange_32(&pDevice->wasapi.isStartedPlayback, MA_FALSE); + /* IAudioRenderClient */ + HRESULT (STDMETHODCALLTYPE * GetBuffer) (ma_IAudioRenderClient* pThis, ma_uint32 numFramesRequested, BYTE** ppData); + HRESULT (STDMETHODCALLTYPE * ReleaseBuffer)(ma_IAudioRenderClient* pThis, ma_uint32 numFramesWritten, DWORD dwFlags); +} ma_IAudioRenderClientVtbl; +struct ma_IAudioRenderClient +{ + ma_IAudioRenderClientVtbl* lpVtbl; +}; +static MA_INLINE HRESULT ma_IAudioRenderClient_QueryInterface(ma_IAudioRenderClient* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } +static MA_INLINE ULONG ma_IAudioRenderClient_AddRef(ma_IAudioRenderClient* pThis) { return pThis->lpVtbl->AddRef(pThis); } +static MA_INLINE ULONG ma_IAudioRenderClient_Release(ma_IAudioRenderClient* pThis) { return pThis->lpVtbl->Release(pThis); } +static MA_INLINE HRESULT ma_IAudioRenderClient_GetBuffer(ma_IAudioRenderClient* pThis, ma_uint32 numFramesRequested, BYTE** ppData) { return pThis->lpVtbl->GetBuffer(pThis, numFramesRequested, ppData); } +static MA_INLINE HRESULT ma_IAudioRenderClient_ReleaseBuffer(ma_IAudioRenderClient* pThis, ma_uint32 numFramesWritten, DWORD dwFlags) { return pThis->lpVtbl->ReleaseBuffer(pThis, numFramesWritten, dwFlags); } - return MA_SUCCESS; -} -ma_result ma_device__get_available_frames__wasapi(ma_device* pDevice, ma_IAudioClient* pAudioClient, ma_uint32* pFrameCount) +/* IAudioCaptureClient */ +typedef struct { - ma_uint32 paddingFramesCount; - HRESULT hr; - ma_share_mode shareMode; + /* IUnknown */ + HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioCaptureClient* pThis, const IID* const riid, void** ppObject); + ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioCaptureClient* pThis); + ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioCaptureClient* pThis); - ma_assert(pDevice != NULL); - ma_assert(pFrameCount != NULL); - - *pFrameCount = 0; + /* IAudioRenderClient */ + HRESULT (STDMETHODCALLTYPE * GetBuffer) (ma_IAudioCaptureClient* pThis, BYTE** ppData, ma_uint32* pNumFramesToRead, DWORD* pFlags, ma_uint64* pDevicePosition, ma_uint64* pQPCPosition); + HRESULT (STDMETHODCALLTYPE * ReleaseBuffer) (ma_IAudioCaptureClient* pThis, ma_uint32 numFramesRead); + HRESULT (STDMETHODCALLTYPE * GetNextPacketSize)(ma_IAudioCaptureClient* pThis, ma_uint32* pNumFramesInNextPacket); +} ma_IAudioCaptureClientVtbl; +struct ma_IAudioCaptureClient +{ + ma_IAudioCaptureClientVtbl* lpVtbl; +}; +static MA_INLINE HRESULT ma_IAudioCaptureClient_QueryInterface(ma_IAudioCaptureClient* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } +static MA_INLINE ULONG ma_IAudioCaptureClient_AddRef(ma_IAudioCaptureClient* pThis) { return pThis->lpVtbl->AddRef(pThis); } +static MA_INLINE ULONG ma_IAudioCaptureClient_Release(ma_IAudioCaptureClient* pThis) { return pThis->lpVtbl->Release(pThis); } +static MA_INLINE HRESULT ma_IAudioCaptureClient_GetBuffer(ma_IAudioCaptureClient* pThis, BYTE** ppData, ma_uint32* pNumFramesToRead, DWORD* pFlags, ma_uint64* pDevicePosition, ma_uint64* pQPCPosition) { return pThis->lpVtbl->GetBuffer(pThis, ppData, pNumFramesToRead, pFlags, pDevicePosition, pQPCPosition); } +static MA_INLINE HRESULT ma_IAudioCaptureClient_ReleaseBuffer(ma_IAudioCaptureClient* pThis, ma_uint32 numFramesRead) { return pThis->lpVtbl->ReleaseBuffer(pThis, numFramesRead); } +static MA_INLINE HRESULT ma_IAudioCaptureClient_GetNextPacketSize(ma_IAudioCaptureClient* pThis, ma_uint32* pNumFramesInNextPacket) { return pThis->lpVtbl->GetNextPacketSize(pThis, pNumFramesInNextPacket); } - if ((ma_ptr)pAudioClient != pDevice->wasapi.pAudioClientPlayback && (ma_ptr)pAudioClient != pDevice->wasapi.pAudioClientCapture) { - return MA_INVALID_OPERATION; - } +#ifndef MA_WIN32_DESKTOP +#include +typedef struct ma_completion_handler_uwp ma_completion_handler_uwp; + +typedef struct +{ + /* IUnknown */ + HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_completion_handler_uwp* pThis, const IID* const riid, void** ppObject); + ULONG (STDMETHODCALLTYPE * AddRef) (ma_completion_handler_uwp* pThis); + ULONG (STDMETHODCALLTYPE * Release) (ma_completion_handler_uwp* pThis); - hr = ma_IAudioClient_GetCurrentPadding(pAudioClient, &paddingFramesCount); - if (FAILED(hr)) { - return MA_DEVICE_UNAVAILABLE; - } + /* IActivateAudioInterfaceCompletionHandler */ + HRESULT (STDMETHODCALLTYPE * ActivateCompleted)(ma_completion_handler_uwp* pThis, ma_IActivateAudioInterfaceAsyncOperation* pActivateOperation); +} ma_completion_handler_uwp_vtbl; +struct ma_completion_handler_uwp +{ + ma_completion_handler_uwp_vtbl* lpVtbl; + ma_uint32 counter; + HANDLE hEvent; +}; - /* Slightly different rules for exclusive and shared modes. */ - shareMode = ((ma_ptr)pAudioClient == pDevice->wasapi.pAudioClientPlayback) ? pDevice->playback.shareMode : pDevice->capture.shareMode; - if (shareMode == ma_share_mode_exclusive) { - *pFrameCount = paddingFramesCount; - } else { - if ((ma_ptr)pAudioClient == pDevice->wasapi.pAudioClientPlayback) { - *pFrameCount = pDevice->wasapi.actualBufferSizeInFramesPlayback - paddingFramesCount; - } else { - *pFrameCount = paddingFramesCount; - } +static HRESULT STDMETHODCALLTYPE ma_completion_handler_uwp_QueryInterface(ma_completion_handler_uwp* pThis, const IID* const riid, void** ppObject) +{ + /* + We need to "implement" IAgileObject which is just an indicator that's used internally by WASAPI for some multithreading management. To + "implement" this, we just make sure we return pThis when the IAgileObject is requested. + */ + if (!ma_is_guid_equal(riid, &MA_IID_IUnknown) && !ma_is_guid_equal(riid, &MA_IID_IActivateAudioInterfaceCompletionHandler) && !ma_is_guid_equal(riid, &MA_IID_IAgileObject)) { + *ppObject = NULL; + return E_NOINTERFACE; } - return MA_SUCCESS; + /* Getting here means the IID is IUnknown or IMMNotificationClient. */ + *ppObject = (void*)pThis; + ((ma_completion_handler_uwp_vtbl*)pThis->lpVtbl)->AddRef(pThis); + return S_OK; } -ma_bool32 ma_device_is_reroute_required__wasapi(ma_device* pDevice, ma_device_type deviceType) +static ULONG STDMETHODCALLTYPE ma_completion_handler_uwp_AddRef(ma_completion_handler_uwp* pThis) { - ma_assert(pDevice != NULL); + return (ULONG)ma_atomic_increment_32(&pThis->counter); +} - if (deviceType == ma_device_type_playback) { - return pDevice->wasapi.hasDefaultPlaybackDeviceChanged; +static ULONG STDMETHODCALLTYPE ma_completion_handler_uwp_Release(ma_completion_handler_uwp* pThis) +{ + ma_uint32 newRefCount = ma_atomic_decrement_32(&pThis->counter); + if (newRefCount == 0) { + return 0; /* We don't free anything here because we never allocate the object on the heap. */ } - if (deviceType == ma_device_type_capture || deviceType == ma_device_type_loopback) { - return pDevice->wasapi.hasDefaultCaptureDeviceChanged; - } - - return MA_FALSE; + return (ULONG)newRefCount; } -ma_result ma_device_reroute__wasapi(ma_device* pDevice, ma_device_type deviceType) +static HRESULT STDMETHODCALLTYPE ma_completion_handler_uwp_ActivateCompleted(ma_completion_handler_uwp* pThis, ma_IActivateAudioInterfaceAsyncOperation* pActivateOperation) { - ma_result result; + (void)pActivateOperation; + SetEvent(pThis->hEvent); + return S_OK; +} - if (deviceType == ma_device_type_duplex) { - return MA_INVALID_ARGS; - } - if (deviceType == ma_device_type_playback) { - ma_atomic_exchange_32(&pDevice->wasapi.hasDefaultPlaybackDeviceChanged, MA_FALSE); - } - if (deviceType == ma_device_type_capture || deviceType == ma_device_type_loopback) { - ma_atomic_exchange_32(&pDevice->wasapi.hasDefaultCaptureDeviceChanged, MA_FALSE); - } - +static ma_completion_handler_uwp_vtbl g_maCompletionHandlerVtblInstance = { + ma_completion_handler_uwp_QueryInterface, + ma_completion_handler_uwp_AddRef, + ma_completion_handler_uwp_Release, + ma_completion_handler_uwp_ActivateCompleted +}; - #ifdef MA_DEBUG_OUTPUT - printf("=== CHANGING DEVICE ===\n"); - #endif +static ma_result ma_completion_handler_uwp_init(ma_completion_handler_uwp* pHandler) +{ + MA_ASSERT(pHandler != NULL); + MA_ZERO_OBJECT(pHandler); - result = ma_device_reinit__wasapi(pDevice, deviceType); - if (result != MA_SUCCESS) { - return result; + pHandler->lpVtbl = &g_maCompletionHandlerVtblInstance; + pHandler->counter = 1; + pHandler->hEvent = CreateEventA(NULL, FALSE, FALSE, NULL); + if (pHandler->hEvent == NULL) { + return MA_ERROR; } - ma_device__post_init_setup(pDevice, deviceType); - return MA_SUCCESS; } +static void ma_completion_handler_uwp_uninit(ma_completion_handler_uwp* pHandler) +{ + if (pHandler->hEvent != NULL) { + CloseHandle(pHandler->hEvent); + } +} -ma_result ma_device_stop__wasapi(ma_device* pDevice) +static void ma_completion_handler_uwp_wait(ma_completion_handler_uwp* pHandler) { - ma_assert(pDevice != NULL); + WaitForSingleObject(pHandler->hEvent, INFINITE); +} +#endif /* !MA_WIN32_DESKTOP */ +/* We need a virtual table for our notification client object that's used for detecting changes to the default device. */ +#ifdef MA_WIN32_DESKTOP +static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_QueryInterface(ma_IMMNotificationClient* pThis, const IID* const riid, void** ppObject) +{ /* - We need to explicitly signal the capture event in loopback mode to ensure we return from WaitForSingleObject() when nothing is being played. When nothing - is being played, the event is never signalled internally by WASAPI which means we will deadlock when stopping the device. + We care about two interfaces - IUnknown and IMMNotificationClient. If the requested IID is something else + we just return E_NOINTERFACE. Otherwise we need to increment the reference counter and return S_OK. */ - if (pDevice->type == ma_device_type_loopback) { - SetEvent((HANDLE)pDevice->wasapi.hEventCapture); + if (!ma_is_guid_equal(riid, &MA_IID_IUnknown) && !ma_is_guid_equal(riid, &MA_IID_IMMNotificationClient)) { + *ppObject = NULL; + return E_NOINTERFACE; } - return MA_SUCCESS; + /* Getting here means the IID is IUnknown or IMMNotificationClient. */ + *ppObject = (void*)pThis; + ((ma_IMMNotificationClientVtbl*)pThis->lpVtbl)->AddRef(pThis); + return S_OK; } - -ma_result ma_device_main_loop__wasapi(ma_device* pDevice) +static ULONG STDMETHODCALLTYPE ma_IMMNotificationClient_AddRef(ma_IMMNotificationClient* pThis) { - ma_result result; - HRESULT hr; - ma_bool32 exitLoop = MA_FALSE; - ma_uint32 framesWrittenToPlaybackDevice = 0; - ma_uint32 mappedBufferSizeInFramesCapture = 0; - ma_uint32 mappedBufferSizeInFramesPlayback = 0; - ma_uint32 mappedBufferFramesRemainingCapture = 0; - ma_uint32 mappedBufferFramesRemainingPlayback = 0; - BYTE* pMappedBufferCapture = NULL; - BYTE* pMappedBufferPlayback = NULL; - ma_uint32 bpfCapture = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 bpfPlayback = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - ma_uint8 inputDataInExternalFormat[4096]; - ma_uint32 inputDataInExternalFormatCap = sizeof(inputDataInExternalFormat) / bpfCapture; - ma_uint8 outputDataInExternalFormat[4096]; - ma_uint32 outputDataInExternalFormatCap = sizeof(outputDataInExternalFormat) / bpfPlayback; - ma_uint32 periodSizeInFramesCapture = 0; - - ma_assert(pDevice != NULL); - - /* The capture device needs to be started immediately. */ - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) { - periodSizeInFramesCapture = (pDevice->capture.internalBufferSizeInFrames / pDevice->capture.internalPeriods); + return (ULONG)ma_atomic_increment_32(&pThis->counter); +} - hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); - if (FAILED(hr)) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal capture device.", MA_FAILED_TO_START_BACKEND_DEVICE); - } - ma_atomic_exchange_32(&pDevice->wasapi.isStartedCapture, MA_TRUE); +static ULONG STDMETHODCALLTYPE ma_IMMNotificationClient_Release(ma_IMMNotificationClient* pThis) +{ + ma_uint32 newRefCount = ma_atomic_decrement_32(&pThis->counter); + if (newRefCount == 0) { + return 0; /* We don't free anything here because we never allocate the object on the heap. */ } - while (ma_device__get_state(pDevice) == MA_STATE_STARTED && !exitLoop) { - /* We may need to reroute the device. */ - if (ma_device_is_reroute_required__wasapi(pDevice, ma_device_type_playback)) { - result = ma_device_reroute__wasapi(pDevice, ma_device_type_playback); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } - } - if (ma_device_is_reroute_required__wasapi(pDevice, ma_device_type_capture)) { - result = ma_device_reroute__wasapi(pDevice, (pDevice->type == ma_device_type_loopback) ? ma_device_type_loopback : ma_device_type_capture); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } - } - - switch (pDevice->type) - { - case ma_device_type_duplex: - { - ma_uint32 framesAvailableCapture; - ma_uint32 framesAvailablePlayback; - DWORD flagsCapture; /* Passed to IAudioCaptureClient_GetBuffer(). */ - - /* The process is to map the playback buffer and fill it as quickly as possible from input data. */ - if (pMappedBufferPlayback == NULL) { - /* WASAPI is weird with exclusive mode. You need to wait on the event _before_ querying the available frames. */ - if (pDevice->playback.shareMode == ma_share_mode_exclusive) { - if (WaitForSingleObject(pDevice->wasapi.hEventPlayback, INFINITE) == WAIT_FAILED) { - return MA_ERROR; /* Wait failed. */ - } - } - - result = ma_device__get_available_frames__wasapi(pDevice, (ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &framesAvailablePlayback); - if (result != MA_SUCCESS) { - return result; - } - - /*printf("TRACE 1: framesAvailablePlayback=%d\n", framesAvailablePlayback);*/ + return (ULONG)newRefCount; +} - /* In exclusive mode, the frame count needs to exactly match the value returned by GetCurrentPadding(). */ - if (pDevice->playback.shareMode != ma_share_mode_exclusive) { - if (framesAvailablePlayback > pDevice->wasapi.periodSizeInFramesPlayback) { - framesAvailablePlayback = pDevice->wasapi.periodSizeInFramesPlayback; - } - } +static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDeviceStateChanged(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID, DWORD dwNewState) +{ +#ifdef MA_DEBUG_OUTPUT + printf("IMMNotificationClient_OnDeviceStateChanged(pDeviceID=%S, dwNewState=%u)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)", (unsigned int)dwNewState); +#endif - /* If there's no frames available in the playback device we need to wait for more. */ - if (framesAvailablePlayback == 0) { - /* In exclusive mode we waited at the top. */ - if (pDevice->playback.shareMode != ma_share_mode_exclusive) { - if (WaitForSingleObject(pDevice->wasapi.hEventPlayback, INFINITE) == WAIT_FAILED) { - return MA_ERROR; /* Wait failed. */ - } - } + (void)pThis; + (void)pDeviceID; + (void)dwNewState; + return S_OK; +} - continue; - } +static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDeviceAdded(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID) +{ +#ifdef MA_DEBUG_OUTPUT + printf("IMMNotificationClient_OnDeviceAdded(pDeviceID=%S)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)"); +#endif - /* We're ready to map the playback device's buffer. We don't release this until it's been entirely filled. */ - hr = ma_IAudioRenderClient_GetBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, framesAvailablePlayback, &pMappedBufferPlayback); - if (FAILED(hr)) { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from playback device in preparation for writing to the device.", MA_FAILED_TO_MAP_DEVICE_BUFFER); - exitLoop = MA_TRUE; - break; - } + /* We don't need to worry about this event for our purposes. */ + (void)pThis; + (void)pDeviceID; + return S_OK; +} - mappedBufferSizeInFramesPlayback = framesAvailablePlayback; - mappedBufferFramesRemainingPlayback = framesAvailablePlayback; - } +static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDeviceRemoved(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID) +{ +#ifdef MA_DEBUG_OUTPUT + printf("IMMNotificationClient_OnDeviceRemoved(pDeviceID=%S)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)"); +#endif - /* At this point we should have a buffer available for output. We need to keep writing input samples to it. */ - for (;;) { - /* Try grabbing some captured data if we haven't already got a mapped buffer. */ - if (pMappedBufferCapture == NULL) { - if (pDevice->capture.shareMode == ma_share_mode_shared) { - if (WaitForSingleObject(pDevice->wasapi.hEventCapture, INFINITE) == WAIT_FAILED) { - return MA_ERROR; /* Wait failed. */ - } - } + /* We don't need to worry about this event for our purposes. */ + (void)pThis; + (void)pDeviceID; + return S_OK; +} - result = ma_device__get_available_frames__wasapi(pDevice, (ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, &framesAvailableCapture); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } +static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDefaultDeviceChanged(ma_IMMNotificationClient* pThis, ma_EDataFlow dataFlow, ma_ERole role, LPCWSTR pDefaultDeviceID) +{ +#ifdef MA_DEBUG_OUTPUT + printf("IMMNotificationClient_OnDefaultDeviceChanged(dataFlow=%d, role=%d, pDefaultDeviceID=%S)\n", dataFlow, role, (pDefaultDeviceID != NULL) ? pDefaultDeviceID : L"(NULL)"); +#endif - /*printf("TRACE 2: framesAvailableCapture=%d\n", framesAvailableCapture);*/ + /* We only ever use the eConsole role in miniaudio. */ + if (role != ma_eConsole) { + return S_OK; + } - /* Wait for more if nothing is available. */ - if (framesAvailableCapture == 0) { - /* In exclusive mode we waited at the top. */ - if (pDevice->capture.shareMode != ma_share_mode_shared) { - if (WaitForSingleObject(pDevice->wasapi.hEventCapture, INFINITE) == WAIT_FAILED) { - return MA_ERROR; /* Wait failed. */ - } - } + /* We only care about devices with the same data flow and role as the current device. */ + if ((pThis->pDevice->type == ma_device_type_playback && dataFlow != ma_eRender) || + (pThis->pDevice->type == ma_device_type_capture && dataFlow != ma_eCapture)) { + return S_OK; + } - continue; - } + /* Don't do automatic stream routing if we're not allowed. */ + if ((dataFlow == ma_eRender && pThis->pDevice->wasapi.allowPlaybackAutoStreamRouting == MA_FALSE) || + (dataFlow == ma_eCapture && pThis->pDevice->wasapi.allowCaptureAutoStreamRouting == MA_FALSE)) { + return S_OK; + } - /* Getting here means there's data available for writing to the output device. */ - mappedBufferSizeInFramesCapture = ma_min(framesAvailableCapture, periodSizeInFramesCapture); - hr = ma_IAudioCaptureClient_GetBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, (BYTE**)&pMappedBufferCapture, &mappedBufferSizeInFramesCapture, &flagsCapture, NULL, NULL); - if (FAILED(hr)) { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from capture device in preparation for writing to the device.", MA_FAILED_TO_MAP_DEVICE_BUFFER); - exitLoop = MA_TRUE; - break; - } + /* + Not currently supporting automatic stream routing in exclusive mode. This is not working correctly on my machine due to + AUDCLNT_E_DEVICE_IN_USE errors when reinitializing the device. If this is a bug in miniaudio, we can try re-enabling this once + it's fixed. + */ + if ((dataFlow == ma_eRender && pThis->pDevice->playback.shareMode == ma_share_mode_exclusive) || + (dataFlow == ma_eCapture && pThis->pDevice->capture.shareMode == ma_share_mode_exclusive)) { + return S_OK; + } + /* + We don't change the device here - we change it in the worker thread to keep synchronization simple. To do this I'm just setting a flag to + indicate that the default device has changed. Loopback devices are treated as capture devices so we need to do a bit of a dance to handle + that properly. + */ + if (dataFlow == ma_eRender && pThis->pDevice->type != ma_device_type_loopback) { + ma_atomic_exchange_32(&pThis->pDevice->wasapi.hasDefaultPlaybackDeviceChanged, MA_TRUE); + } + if (dataFlow == ma_eCapture || pThis->pDevice->type == ma_device_type_loopback) { + ma_atomic_exchange_32(&pThis->pDevice->wasapi.hasDefaultCaptureDeviceChanged, MA_TRUE); + } - /* Overrun detection. */ - if ((flagsCapture & MA_AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY) != 0) { - /* Glitched. Probably due to an overrun. */ - #ifdef MA_DEBUG_OUTPUT - printf("[WASAPI] Data discontinuity (possible overrun). framesAvailableCapture=%d, mappedBufferSizeInFramesCapture=%d\n", framesAvailableCapture, mappedBufferSizeInFramesCapture); - #endif + (void)pDefaultDeviceID; + return S_OK; +} - /* - Exeriment: If we get an overrun it probably means we're straddling the end of the buffer. In order to prevent a never-ending sequence of glitches let's experiment - by dropping every frame until we're left with only a single period. To do this we just keep retrieving and immediately releasing buffers until we're down to the - last period. - */ - if (framesAvailableCapture >= pDevice->wasapi.actualBufferSizeInFramesCapture /*(pDevice->playback.internalBufferSizeInFrames / pDevice->playback.internalPeriods)*/) { - #ifdef MA_DEBUG_OUTPUT - printf("[WASAPI] Synchronizing capture stream. "); - #endif - do - { - hr = ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, mappedBufferSizeInFramesCapture); - if (FAILED(hr)) { - break; - } +static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnPropertyValueChanged(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID, const PROPERTYKEY key) +{ +#ifdef MA_DEBUG_OUTPUT + printf("IMMNotificationClient_OnPropertyValueChanged(pDeviceID=%S)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)"); +#endif - framesAvailableCapture -= mappedBufferSizeInFramesCapture; - - if (framesAvailableCapture > 0) { - mappedBufferSizeInFramesCapture = ma_min(framesAvailableCapture, periodSizeInFramesCapture); - hr = ma_IAudioCaptureClient_GetBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, (BYTE**)&pMappedBufferCapture, &mappedBufferSizeInFramesCapture, &flagsCapture, NULL, NULL); - if (FAILED(hr)) { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from capture device in preparation for writing to the device.", MA_FAILED_TO_MAP_DEVICE_BUFFER); - exitLoop = MA_TRUE; - break; - } - } else { - pMappedBufferCapture = NULL; - mappedBufferSizeInFramesCapture = 0; - } - } while (framesAvailableCapture > periodSizeInFramesCapture); - #ifdef MA_DEBUG_OUTPUT - printf("framesAvailableCapture=%d, mappedBufferSizeInFramesCapture=%d\n", framesAvailableCapture, mappedBufferSizeInFramesCapture); - #endif - } - } else { - #ifdef MA_DEBUG_OUTPUT - if (flagsCapture != 0) { - printf("[WASAPI] Capture Flags: %d\n", flagsCapture); - } - #endif - } + (void)pThis; + (void)pDeviceID; + (void)key; + return S_OK; +} - mappedBufferFramesRemainingCapture = mappedBufferSizeInFramesCapture; +static ma_IMMNotificationClientVtbl g_maNotificationCientVtbl = { + ma_IMMNotificationClient_QueryInterface, + ma_IMMNotificationClient_AddRef, + ma_IMMNotificationClient_Release, + ma_IMMNotificationClient_OnDeviceStateChanged, + ma_IMMNotificationClient_OnDeviceAdded, + ma_IMMNotificationClient_OnDeviceRemoved, + ma_IMMNotificationClient_OnDefaultDeviceChanged, + ma_IMMNotificationClient_OnPropertyValueChanged +}; +#endif /* MA_WIN32_DESKTOP */ - pDevice->capture._dspFrameCount = mappedBufferSizeInFramesCapture; - if ((flagsCapture & MA_AUDCLNT_BUFFERFLAGS_SILENT) == 0) { - pDevice->capture._dspFrames = (const ma_uint8*)pMappedBufferCapture; - } else { - pDevice->capture._dspFrames = NULL; - } - } +#ifdef MA_WIN32_DESKTOP +typedef ma_IMMDevice ma_WASAPIDeviceInterface; +#else +typedef ma_IUnknown ma_WASAPIDeviceInterface; +#endif - /* At this point we should have both input and output data available. We now need to convert the data and post it to the client. */ - for (;;) { - BYTE* pRunningBufferCapture; - BYTE* pRunningBufferPlayback; - ma_uint32 framesToProcess; - ma_uint32 framesProcessed; - pRunningBufferCapture = pMappedBufferCapture + ((mappedBufferSizeInFramesCapture - mappedBufferFramesRemainingCapture ) * bpfPlayback); - pRunningBufferPlayback = pMappedBufferPlayback + ((mappedBufferSizeInFramesPlayback - mappedBufferFramesRemainingPlayback) * bpfPlayback); - - /* There may be some data sitting in the converter that needs to be processed first. Once this is exhaused, run the data callback again. */ - if (!pDevice->playback.converter.isPassthrough) { - framesProcessed = (ma_uint32)ma_pcm_converter_read(&pDevice->playback.converter, pRunningBufferPlayback, mappedBufferFramesRemainingPlayback); - if (framesProcessed > 0) { - mappedBufferFramesRemainingPlayback -= framesProcessed; - if (mappedBufferFramesRemainingPlayback == 0) { - break; - } - } - } +static ma_bool32 ma_context_is_device_id_equal__wasapi(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) +{ + MA_ASSERT(pContext != NULL); + MA_ASSERT(pID0 != NULL); + MA_ASSERT(pID1 != NULL); + (void)pContext; - /* - Getting here means we need to fire the callback. If format conversion is unnecessary, we can optimize this by passing the pointers to the internal - buffers directly to the callback. - */ - if (pDevice->capture.converter.isPassthrough && pDevice->playback.converter.isPassthrough) { - /* Optimal path. We can pass mapped pointers directly to the callback. */ - framesToProcess = ma_min(mappedBufferFramesRemainingCapture, mappedBufferFramesRemainingPlayback); - framesProcessed = framesToProcess; + return memcmp(pID0->wasapi, pID1->wasapi, sizeof(pID0->wasapi)) == 0; +} + +static void ma_set_device_info_from_WAVEFORMATEX(const WAVEFORMATEX* pWF, ma_device_info* pInfo) +{ + MA_ASSERT(pWF != NULL); + MA_ASSERT(pInfo != NULL); + + pInfo->formatCount = 1; + pInfo->formats[0] = ma_format_from_WAVEFORMATEX(pWF); + pInfo->minChannels = pWF->nChannels; + pInfo->maxChannels = pWF->nChannels; + pInfo->minSampleRate = pWF->nSamplesPerSec; + pInfo->maxSampleRate = pWF->nSamplesPerSec; +} - ma_device__on_data(pDevice, pRunningBufferPlayback, pRunningBufferCapture, framesToProcess); +static ma_result ma_context_get_device_info_from_IAudioClient__wasapi(ma_context* pContext, /*ma_IMMDevice**/void* pMMDevice, ma_IAudioClient* pAudioClient, ma_share_mode shareMode, ma_device_info* pInfo) +{ + MA_ASSERT(pAudioClient != NULL); + MA_ASSERT(pInfo != NULL); - mappedBufferFramesRemainingCapture -= framesProcessed; - mappedBufferFramesRemainingPlayback -= framesProcessed; + /* We use a different technique to retrieve the device information depending on whether or not we are using shared or exclusive mode. */ + if (shareMode == ma_share_mode_shared) { + /* Shared Mode. We use GetMixFormat() here. */ + WAVEFORMATEX* pWF = NULL; + HRESULT hr = ma_IAudioClient_GetMixFormat((ma_IAudioClient*)pAudioClient, (WAVEFORMATEX**)&pWF); + if (SUCCEEDED(hr)) { + ma_set_device_info_from_WAVEFORMATEX(pWF, pInfo); + return MA_SUCCESS; + } else { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve mix format for device info retrieval.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } + } else { + /* Exlcusive Mode. We repeatedly call IsFormatSupported() here. This is not currently support on UWP. */ +#ifdef MA_WIN32_DESKTOP + /* + The first thing to do is get the format from PKEY_AudioEngine_DeviceFormat. This should give us a channel count we assume is + correct which will simplify our searching. + */ + ma_IPropertyStore *pProperties; + HRESULT hr = ma_IMMDevice_OpenPropertyStore((ma_IMMDevice*)pMMDevice, STGM_READ, &pProperties); + if (SUCCEEDED(hr)) { + PROPVARIANT var; + ma_PropVariantInit(&var); - if (mappedBufferFramesRemainingCapture == 0) { - break; /* Exhausted input data. */ - } - if (mappedBufferFramesRemainingPlayback == 0) { - break; /* Exhausted output data. */ - } - } else if (pDevice->capture.converter.isPassthrough) { - /* The input buffer is a passthrough, but the playback buffer requires a conversion. */ - framesToProcess = ma_min(mappedBufferFramesRemainingCapture, outputDataInExternalFormatCap); - framesProcessed = framesToProcess; + hr = ma_IPropertyStore_GetValue(pProperties, &MA_PKEY_AudioEngine_DeviceFormat, &var); + if (SUCCEEDED(hr)) { + WAVEFORMATEX* pWF = (WAVEFORMATEX*)var.blob.pBlobData; + ma_set_device_info_from_WAVEFORMATEX(pWF, pInfo); - ma_device__on_data(pDevice, outputDataInExternalFormat, pRunningBufferCapture, framesToProcess); - mappedBufferFramesRemainingCapture -= framesProcessed; + /* + In my testing, the format returned by PKEY_AudioEngine_DeviceFormat is suitable for exclusive mode so we check this format + first. If this fails, fall back to a search. + */ + hr = ma_IAudioClient_IsFormatSupported((ma_IAudioClient*)pAudioClient, MA_AUDCLNT_SHAREMODE_EXCLUSIVE, pWF, NULL); + ma_PropVariantClear(pContext, &var); - pDevice->playback._dspFrameCount = framesProcessed; - pDevice->playback._dspFrames = (const ma_uint8*)outputDataInExternalFormat; + if (FAILED(hr)) { + /* + The format returned by PKEY_AudioEngine_DeviceFormat is not supported, so fall back to a search. We assume the channel + count returned by MA_PKEY_AudioEngine_DeviceFormat is valid and correct. For simplicity we're only returning one format. + */ + ma_uint32 channels = pInfo->minChannels; + ma_format formatsToSearch[] = { + ma_format_s16, + ma_format_s24, + /*ma_format_s24_32,*/ + ma_format_f32, + ma_format_s32, + ma_format_u8 + }; + ma_channel defaultChannelMap[MA_MAX_CHANNELS]; + WAVEFORMATEXTENSIBLE wf; + ma_bool32 found; + ma_uint32 iFormat; - if (mappedBufferFramesRemainingCapture == 0) { - break; /* Exhausted input data. */ - } - } else if (pDevice->playback.converter.isPassthrough) { - /* The input buffer requires conversion, the playback buffer is passthrough. */ - framesToProcess = ma_min(inputDataInExternalFormatCap, mappedBufferFramesRemainingPlayback); - framesProcessed = (ma_uint32)ma_pcm_converter_read(&pDevice->capture.converter, inputDataInExternalFormat, framesToProcess); - if (framesProcessed == 0) { - /* Getting here means we've run out of input data. */ - mappedBufferFramesRemainingCapture = 0; - break; - } + ma_get_standard_channel_map(ma_standard_channel_map_microsoft, channels, defaultChannelMap); - ma_device__on_data(pDevice, pRunningBufferPlayback, inputDataInExternalFormat, framesProcessed); - mappedBufferFramesRemainingPlayback -= framesProcessed; + MA_ZERO_OBJECT(&wf); + wf.Format.cbSize = sizeof(wf); + wf.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE; + wf.Format.nChannels = (WORD)channels; + wf.dwChannelMask = ma_channel_map_to_channel_mask__win32(defaultChannelMap, channels); - if (framesProcessed < framesToProcess) { - mappedBufferFramesRemainingCapture = 0; - break; /* Exhausted input data. */ - } + found = MA_FALSE; + for (iFormat = 0; iFormat < ma_countof(formatsToSearch); ++iFormat) { + ma_format format = formatsToSearch[iFormat]; + ma_uint32 iSampleRate; - if (mappedBufferFramesRemainingPlayback == 0) { - break; /* Exhausted output data. */ - } + wf.Format.wBitsPerSample = (WORD)ma_get_bytes_per_sample(format)*8; + wf.Format.nBlockAlign = (wf.Format.nChannels * wf.Format.wBitsPerSample) / 8; + wf.Format.nAvgBytesPerSec = wf.Format.nBlockAlign * wf.Format.nSamplesPerSec; + wf.Samples.wValidBitsPerSample = /*(format == ma_format_s24_32) ? 24 :*/ wf.Format.wBitsPerSample; + if (format == ma_format_f32) { + wf.SubFormat = MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT; } else { - framesToProcess = ma_min(inputDataInExternalFormatCap, outputDataInExternalFormatCap); - framesProcessed = (ma_uint32)ma_pcm_converter_read(&pDevice->capture.converter, inputDataInExternalFormat, framesToProcess); - if (framesProcessed == 0) { - /* Getting here means we've run out of input data. */ - mappedBufferFramesRemainingCapture = 0; - break; - } - - ma_device__on_data(pDevice, outputDataInExternalFormat, inputDataInExternalFormat, framesProcessed); + wf.SubFormat = MA_GUID_KSDATAFORMAT_SUBTYPE_PCM; + } - pDevice->playback._dspFrameCount = framesProcessed; - pDevice->playback._dspFrames = (const ma_uint8*)outputDataInExternalFormat; + for (iSampleRate = 0; iSampleRate < ma_countof(g_maStandardSampleRatePriorities); ++iSampleRate) { + wf.Format.nSamplesPerSec = g_maStandardSampleRatePriorities[iSampleRate]; - if (framesProcessed < framesToProcess) { - /* Getting here means we've run out of input data. */ - mappedBufferFramesRemainingCapture = 0; + hr = ma_IAudioClient_IsFormatSupported((ma_IAudioClient*)pAudioClient, MA_AUDCLNT_SHAREMODE_EXCLUSIVE, (WAVEFORMATEX*)&wf, NULL); + if (SUCCEEDED(hr)) { + ma_set_device_info_from_WAVEFORMATEX((WAVEFORMATEX*)&wf, pInfo); + found = MA_TRUE; break; } } - } - - /* If at this point we've run out of capture data we need to release the buffer. */ - if (mappedBufferFramesRemainingCapture == 0 && pMappedBufferCapture != NULL) { - hr = ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, mappedBufferSizeInFramesCapture); - if (FAILED(hr)) { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to release internal buffer from capture device after reading from the device.", MA_FAILED_TO_UNMAP_DEVICE_BUFFER); - exitLoop = MA_TRUE; + if (found) { break; } + } - /*printf("TRACE: Released capture buffer\n");*/ + if (!found) { + ma_IPropertyStore_Release(pProperties); + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to find suitable device format for device info retrieval.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } + } + } else { + ma_IPropertyStore_Release(pProperties); + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve device format for device info retrieval.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } + + ma_IPropertyStore_Release(pProperties); + } else { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to open property store for device info retrieval.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } + + return MA_SUCCESS; +#else + /* Exclusive mode not fully supported in UWP right now. */ + return MA_ERROR; +#endif + } +} - pMappedBufferCapture = NULL; - mappedBufferFramesRemainingCapture = 0; - mappedBufferSizeInFramesCapture = 0; - } +#ifdef MA_WIN32_DESKTOP +static ma_EDataFlow ma_device_type_to_EDataFlow(ma_device_type deviceType) +{ + if (deviceType == ma_device_type_playback) { + return ma_eRender; + } else if (deviceType == ma_device_type_capture) { + return ma_eCapture; + } else { + MA_ASSERT(MA_FALSE); + return ma_eRender; /* Should never hit this. */ + } +} - /* Get out of this loop if we're run out of room in the playback buffer. */ - if (mappedBufferFramesRemainingPlayback == 0) { - break; - } - } +static ma_result ma_context_create_IMMDeviceEnumerator__wasapi(ma_context* pContext, ma_IMMDeviceEnumerator** ppDeviceEnumerator) +{ + HRESULT hr; + ma_IMMDeviceEnumerator* pDeviceEnumerator; + MA_ASSERT(pContext != NULL); + MA_ASSERT(ppDeviceEnumerator != NULL); - /* If at this point we've run out of data we need to release the buffer. */ - if (mappedBufferFramesRemainingPlayback == 0 && pMappedBufferPlayback != NULL) { - hr = ma_IAudioRenderClient_ReleaseBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, mappedBufferSizeInFramesPlayback, 0); - if (FAILED(hr)) { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to release internal buffer from playback device after writing to the device.", MA_FAILED_TO_UNMAP_DEVICE_BUFFER); - exitLoop = MA_TRUE; - break; - } + hr = ma_CoCreateInstance(pContext, MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator); + if (FAILED(hr)) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create device enumerator.", MA_ERROR); + } - /*printf("TRACE: Released playback buffer\n");*/ - framesWrittenToPlaybackDevice += mappedBufferSizeInFramesPlayback; + *ppDeviceEnumerator = pDeviceEnumerator; - pMappedBufferPlayback = NULL; - mappedBufferFramesRemainingPlayback = 0; - mappedBufferSizeInFramesPlayback = 0; - } + return MA_SUCCESS; +} - if (!pDevice->wasapi.isStartedPlayback) { - ma_uint32 startThreshold = pDevice->playback.internalBufferSizeInFrames / pDevice->playback.internalPeriods * 1; +static LPWSTR ma_context_get_default_device_id_from_IMMDeviceEnumerator__wasapi(ma_context* pContext, ma_IMMDeviceEnumerator* pDeviceEnumerator, ma_device_type deviceType) +{ + HRESULT hr; + ma_IMMDevice* pMMDefaultDevice = NULL; + LPWSTR pDefaultDeviceID = NULL; + ma_EDataFlow dataFlow; + ma_ERole role; - /* Prevent a deadlock. If we don't clamp against the actual buffer size we'll never end up starting the playback device which will result in a deadlock. */ - if (startThreshold > pDevice->wasapi.actualBufferSizeInFramesPlayback) { - startThreshold = pDevice->wasapi.actualBufferSizeInFramesPlayback; - } + MA_ASSERT(pContext != NULL); + MA_ASSERT(pDeviceEnumerator != NULL); - if (pDevice->playback.shareMode == ma_share_mode_exclusive || framesWrittenToPlaybackDevice >= startThreshold) { - hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); - if (FAILED(hr)) { - ma_IAudioClient_Stop((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); - ma_IAudioClient_Reset((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal playback device.", MA_FAILED_TO_START_BACKEND_DEVICE); - } - ma_atomic_exchange_32(&pDevice->wasapi.isStartedPlayback, MA_TRUE); - } - } - } break; + /* Grab the EDataFlow type from the device type. */ + dataFlow = ma_device_type_to_EDataFlow(deviceType); + /* The role is always eConsole, but we may make this configurable later. */ + role = ma_eConsole; + hr = ma_IMMDeviceEnumerator_GetDefaultAudioEndpoint(pDeviceEnumerator, dataFlow, role, &pMMDefaultDevice); + if (FAILED(hr)) { + return NULL; + } - case ma_device_type_capture: - case ma_device_type_loopback: - { - ma_uint32 framesAvailableCapture; - DWORD flagsCapture; /* Passed to IAudioCaptureClient_GetBuffer(). */ + hr = ma_IMMDevice_GetId(pMMDefaultDevice, &pDefaultDeviceID); - /* Wait for data to become available first. */ - if (WaitForSingleObject(pDevice->wasapi.hEventCapture, INFINITE) == WAIT_FAILED) { - exitLoop = MA_TRUE; - break; /* Wait failed. */ - } + ma_IMMDevice_Release(pMMDefaultDevice); + pMMDefaultDevice = NULL; - /* See how many frames are available. Since we waited at the top, I don't think this should ever return 0. I'm checking for this anyway. */ - result = ma_device__get_available_frames__wasapi(pDevice, (ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, &framesAvailableCapture); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } + if (FAILED(hr)) { + return NULL; + } - if (framesAvailableCapture < pDevice->wasapi.periodSizeInFramesCapture) { - continue; /* Nothing available. Keep waiting. */ - } + return pDefaultDeviceID; +} - /* Map the data buffer in preparation for sending to the client. */ - mappedBufferSizeInFramesCapture = framesAvailableCapture; - hr = ma_IAudioCaptureClient_GetBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, (BYTE**)&pMappedBufferCapture, &mappedBufferSizeInFramesCapture, &flagsCapture, NULL, NULL); - if (FAILED(hr)) { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from capture device in preparation for writing to the device.", MA_FAILED_TO_MAP_DEVICE_BUFFER); - exitLoop = MA_TRUE; - break; - } +static LPWSTR ma_context_get_default_device_id__wasapi(ma_context* pContext, ma_device_type deviceType) /* Free the returned pointer with ma_CoTaskMemFree() */ +{ + ma_result result; + ma_IMMDeviceEnumerator* pDeviceEnumerator; + LPWSTR pDefaultDeviceID = NULL; - /* We should have a buffer at this point. */ - ma_device__send_frames_to_client(pDevice, mappedBufferSizeInFramesCapture, pMappedBufferCapture); + MA_ASSERT(pContext != NULL); - /* At this point we're done with the buffer. */ - hr = ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, mappedBufferSizeInFramesCapture); - pMappedBufferCapture = NULL; /* <-- Important. Not doing this can result in an error once we leave this loop because it will use this to know whether or not a final ReleaseBuffer() needs to be called. */ - mappedBufferSizeInFramesCapture = 0; - if (FAILED(hr)) { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to release internal buffer from capture device after reading from the device.", MA_FAILED_TO_UNMAP_DEVICE_BUFFER); - exitLoop = MA_TRUE; - break; - } - } break; + result = ma_context_create_IMMDeviceEnumerator__wasapi(pContext, &pDeviceEnumerator); + if (result != MA_SUCCESS) { + return NULL; + } + pDefaultDeviceID = ma_context_get_default_device_id_from_IMMDeviceEnumerator__wasapi(pContext, pDeviceEnumerator, deviceType); + + ma_IMMDeviceEnumerator_Release(pDeviceEnumerator); + return pDefaultDeviceID; +} +static ma_result ma_context_get_MMDevice__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_IMMDevice** ppMMDevice) +{ + ma_IMMDeviceEnumerator* pDeviceEnumerator; + HRESULT hr; - case ma_device_type_playback: - { - ma_uint32 framesAvailablePlayback; + MA_ASSERT(pContext != NULL); + MA_ASSERT(ppMMDevice != NULL); - /* Wait for space to become available first. */ - if (WaitForSingleObject(pDevice->wasapi.hEventPlayback, INFINITE) == WAIT_FAILED) { - exitLoop = MA_TRUE; - break; /* Wait failed. */ - } + hr = ma_CoCreateInstance(pContext, MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator); + if (FAILED(hr)) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create IMMDeviceEnumerator.", MA_FAILED_TO_INIT_BACKEND); + } - /* Check how much space is available. If this returns 0 we just keep waiting. */ - result = ma_device__get_available_frames__wasapi(pDevice, (ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &framesAvailablePlayback); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } + if (pDeviceID == NULL) { + hr = ma_IMMDeviceEnumerator_GetDefaultAudioEndpoint(pDeviceEnumerator, (deviceType == ma_device_type_capture) ? ma_eCapture : ma_eRender, ma_eConsole, ppMMDevice); + } else { + hr = ma_IMMDeviceEnumerator_GetDevice(pDeviceEnumerator, pDeviceID->wasapi, ppMMDevice); + } - if (framesAvailablePlayback < pDevice->wasapi.periodSizeInFramesPlayback) { - continue; /* No space available. */ - } + ma_IMMDeviceEnumerator_Release(pDeviceEnumerator); + if (FAILED(hr)) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve IMMDevice.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - /* Map a the data buffer in preparation for the callback. */ - hr = ma_IAudioRenderClient_GetBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, framesAvailablePlayback, &pMappedBufferPlayback); - if (FAILED(hr)) { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from playback device in preparation for writing to the device.", MA_FAILED_TO_MAP_DEVICE_BUFFER); - exitLoop = MA_TRUE; - break; - } + return MA_SUCCESS; +} - /* We should have a buffer at this point. */ - ma_device__read_frames_from_client(pDevice, framesAvailablePlayback, pMappedBufferPlayback); +static ma_result ma_context_get_device_info_from_MMDevice__wasapi(ma_context* pContext, ma_IMMDevice* pMMDevice, ma_share_mode shareMode, LPWSTR pDefaultDeviceID, ma_bool32 onlySimpleInfo, ma_device_info* pInfo) +{ + LPWSTR pDeviceID; + HRESULT hr; - /* At this point we're done writing to the device and we just need to release the buffer. */ - hr = ma_IAudioRenderClient_ReleaseBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, framesAvailablePlayback, 0); - pMappedBufferPlayback = NULL; /* <-- Important. Not doing this can result in an error once we leave this loop because it will use this to know whether or not a final ReleaseBuffer() needs to be called. */ - mappedBufferSizeInFramesPlayback = 0; + MA_ASSERT(pContext != NULL); + MA_ASSERT(pMMDevice != NULL); + MA_ASSERT(pInfo != NULL); - if (FAILED(hr)) { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to release internal buffer from playback device after writing to the device.", MA_FAILED_TO_UNMAP_DEVICE_BUFFER); - exitLoop = MA_TRUE; - break; - } + /* ID. */ + hr = ma_IMMDevice_GetId(pMMDevice, &pDeviceID); + if (SUCCEEDED(hr)) { + size_t idlen = wcslen(pDeviceID); + if (idlen+1 > ma_countof(pInfo->id.wasapi)) { + ma_CoTaskMemFree(pContext, pDeviceID); + MA_ASSERT(MA_FALSE); /* NOTE: If this is triggered, please report it. It means the format of the ID must haved change and is too long to fit in our fixed sized buffer. */ + return MA_ERROR; + } - framesWrittenToPlaybackDevice += framesAvailablePlayback; - if (!pDevice->wasapi.isStartedPlayback) { - if (pDevice->playback.shareMode == ma_share_mode_exclusive || framesWrittenToPlaybackDevice >= (pDevice->playback.internalBufferSizeInFrames/pDevice->playback.internalPeriods)*1) { - hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); - if (FAILED(hr)) { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal playback device.", MA_FAILED_TO_START_BACKEND_DEVICE); - exitLoop = MA_TRUE; - break; - } - ma_atomic_exchange_32(&pDevice->wasapi.isStartedPlayback, MA_TRUE); - } - } - } break; + MA_COPY_MEMORY(pInfo->id.wasapi, pDeviceID, idlen * sizeof(wchar_t)); + pInfo->id.wasapi[idlen] = '\0'; - default: return MA_INVALID_ARGS; + if (pDefaultDeviceID != NULL) { + if (wcscmp(pDeviceID, pDefaultDeviceID) == 0) { + /* It's a default device. */ + pInfo->_private.isDefault = MA_TRUE; + } } + + ma_CoTaskMemFree(pContext, pDeviceID); } - /* Here is where the device needs to be stopped. */ - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) { - /* Any mapped buffers need to be released. */ - if (pMappedBufferCapture != NULL) { - hr = ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, mappedBufferSizeInFramesCapture); - } + { + ma_IPropertyStore *pProperties; + hr = ma_IMMDevice_OpenPropertyStore(pMMDevice, STGM_READ, &pProperties); + if (SUCCEEDED(hr)) { + PROPVARIANT var; - hr = ma_IAudioClient_Stop((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); - if (FAILED(hr)) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to stop internal capture device.", MA_FAILED_TO_STOP_BACKEND_DEVICE); - } + /* Description / Friendly Name */ + ma_PropVariantInit(&var); + hr = ma_IPropertyStore_GetValue(pProperties, &MA_PKEY_Device_FriendlyName, &var); + if (SUCCEEDED(hr)) { + WideCharToMultiByte(CP_UTF8, 0, var.pwszVal, -1, pInfo->name, sizeof(pInfo->name), 0, FALSE); + ma_PropVariantClear(pContext, &var); + } - /* The audio client needs to be reset otherwise restarting will fail. */ - hr = ma_IAudioClient_Reset((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); - if (FAILED(hr)) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to reset internal capture device.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + ma_IPropertyStore_Release(pProperties); } - - ma_atomic_exchange_32(&pDevice->wasapi.isStartedCapture, MA_FALSE); } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - /* Any mapped buffers need to be released. */ - if (pMappedBufferPlayback != NULL) { - hr = ma_IAudioRenderClient_ReleaseBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, mappedBufferSizeInFramesPlayback, 0); + /* Format */ + if (!onlySimpleInfo) { + ma_IAudioClient* pAudioClient; + hr = ma_IMMDevice_Activate(pMMDevice, &MA_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pAudioClient); + if (SUCCEEDED(hr)) { + ma_result result = ma_context_get_device_info_from_IAudioClient__wasapi(pContext, pMMDevice, pAudioClient, shareMode, pInfo); + + ma_IAudioClient_Release(pAudioClient); + return result; + } else { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to activate audio client for device info retrieval.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); } + } - /* - The buffer needs to be drained before stopping the device. Not doing this will result in the last few frames not getting output to - the speakers. This is a problem for very short sounds because it'll result in a significant portion of it not getting played. - */ - if (pDevice->wasapi.isStartedPlayback) { - if (pDevice->playback.shareMode == ma_share_mode_exclusive) { - WaitForSingleObject(pDevice->wasapi.hEventPlayback, INFINITE); - } else { - ma_uint32 prevFramesAvaialablePlayback = (ma_uint32)-1; - ma_uint32 framesAvailablePlayback; - for (;;) { - result = ma_device__get_available_frames__wasapi(pDevice, (ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &framesAvailablePlayback); - if (result != MA_SUCCESS) { - break; - } - - if (framesAvailablePlayback >= pDevice->wasapi.actualBufferSizeInFramesPlayback) { - break; - } + return MA_SUCCESS; +} - /* - Just a safety check to avoid an infinite loop. If this iteration results in a situation where the number of available frames - has not changed, get out of the loop. I don't think this should ever happen, but I think it's nice to have just in case. - */ - if (framesAvailablePlayback == prevFramesAvaialablePlayback) { - break; - } - prevFramesAvaialablePlayback = framesAvailablePlayback; +static ma_result ma_context_enumerate_devices_by_type__wasapi(ma_context* pContext, ma_IMMDeviceEnumerator* pDeviceEnumerator, ma_device_type deviceType, ma_enum_devices_callback_proc callback, void* pUserData) +{ + ma_result result = MA_SUCCESS; + UINT deviceCount; + HRESULT hr; + ma_uint32 iDevice; + LPWSTR pDefaultDeviceID = NULL; + ma_IMMDeviceCollection* pDeviceCollection = NULL; + + MA_ASSERT(pContext != NULL); + MA_ASSERT(callback != NULL); - WaitForSingleObject(pDevice->wasapi.hEventPlayback, INFINITE); - ResetEvent(pDevice->wasapi.hEventPlayback); /* Manual reset. */ - } - } - } + /* Grab the default device. We use this to know whether or not flag the returned device info as being the default. */ + pDefaultDeviceID = ma_context_get_default_device_id_from_IMMDeviceEnumerator__wasapi(pContext, pDeviceEnumerator, deviceType); - hr = ma_IAudioClient_Stop((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); + /* We need to enumerate the devices which returns a device collection. */ + hr = ma_IMMDeviceEnumerator_EnumAudioEndpoints(pDeviceEnumerator, ma_device_type_to_EDataFlow(deviceType), MA_MM_DEVICE_STATE_ACTIVE, &pDeviceCollection); + if (SUCCEEDED(hr)) { + hr = ma_IMMDeviceCollection_GetCount(pDeviceCollection, &deviceCount); if (FAILED(hr)) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to stop internal playback device.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + result = ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to get device count.", MA_NO_DEVICE); + goto done; } - /* The audio client needs to be reset otherwise restarting will fail. */ - hr = ma_IAudioClient_Reset((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); - if (FAILED(hr)) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to reset internal playback device.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + for (iDevice = 0; iDevice < deviceCount; ++iDevice) { + ma_device_info deviceInfo; + ma_IMMDevice* pMMDevice; + + MA_ZERO_OBJECT(&deviceInfo); + + hr = ma_IMMDeviceCollection_Item(pDeviceCollection, iDevice, &pMMDevice); + if (SUCCEEDED(hr)) { + result = ma_context_get_device_info_from_MMDevice__wasapi(pContext, pMMDevice, ma_share_mode_shared, pDefaultDeviceID, MA_TRUE, &deviceInfo); /* MA_TRUE = onlySimpleInfo. */ + + ma_IMMDevice_Release(pMMDevice); + if (result == MA_SUCCESS) { + ma_bool32 cbResult = callback(pContext, deviceType, &deviceInfo, pUserData); + if (cbResult == MA_FALSE) { + break; + } + } + } } - - ma_atomic_exchange_32(&pDevice->wasapi.isStartedPlayback, MA_FALSE); } - return MA_SUCCESS; -} +done: + if (pDefaultDeviceID != NULL) { + ma_CoTaskMemFree(pContext, pDefaultDeviceID); + pDefaultDeviceID = NULL; + } -ma_result ma_context_uninit__wasapi(ma_context* pContext) -{ - ma_assert(pContext != NULL); - ma_assert(pContext->backend == ma_backend_wasapi); - (void)pContext; + if (pDeviceCollection != NULL) { + ma_IMMDeviceCollection_Release(pDeviceCollection); + pDeviceCollection = NULL; + } - return MA_SUCCESS; + return result; } -ma_result ma_context_init__wasapi(const ma_context_config* pConfig, ma_context* pContext) +static ma_result ma_context_get_IAudioClient_Desktop__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_IAudioClient** ppAudioClient, ma_IMMDevice** ppMMDevice) { - ma_result result = MA_SUCCESS; - - ma_assert(pContext != NULL); + ma_result result; + HRESULT hr; - (void)pConfig; + MA_ASSERT(pContext != NULL); + MA_ASSERT(ppAudioClient != NULL); + MA_ASSERT(ppMMDevice != NULL); -#ifdef MA_WIN32_DESKTOP - /* - WASAPI is only supported in Vista SP1 and newer. The reason for SP1 and not the base version of Vista is that event-driven - exclusive mode does not work until SP1. + result = ma_context_get_MMDevice__wasapi(pContext, deviceType, pDeviceID, ppMMDevice); + if (result != MA_SUCCESS) { + return result; + } - Unfortunately older compilers don't define these functions so we need to dynamically load them in order to avoid a lin error. - */ - { - ma_OSVERSIONINFOEXW osvi; - ma_handle kernel32DLL; - ma_PFNVerifyVersionInfoW _VerifyVersionInfoW; - ma_PFNVerSetConditionMask _VerSetConditionMask; + hr = ma_IMMDevice_Activate(*ppMMDevice, &MA_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)ppAudioClient); + if (FAILED(hr)) { + return MA_FAILED_TO_OPEN_BACKEND_DEVICE; + } - kernel32DLL = ma_dlopen(pContext, "kernel32.dll"); - if (kernel32DLL == NULL) { - return MA_NO_BACKEND; - } + return MA_SUCCESS; +} +#else +static ma_result ma_context_get_IAudioClient_UWP__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_IAudioClient** ppAudioClient, ma_IUnknown** ppActivatedInterface) +{ + ma_IActivateAudioInterfaceAsyncOperation *pAsyncOp = NULL; + ma_completion_handler_uwp completionHandler; + IID iid; + LPOLESTR iidStr; + HRESULT hr; + ma_result result; + HRESULT activateResult; + ma_IUnknown* pActivatedInterface; - _VerifyVersionInfoW = (ma_PFNVerifyVersionInfoW)ma_dlsym(pContext, kernel32DLL, "VerifyVersionInfoW"); - _VerSetConditionMask = (ma_PFNVerSetConditionMask)ma_dlsym(pContext, kernel32DLL, "VerSetConditionMask"); - if (_VerifyVersionInfoW == NULL || _VerSetConditionMask == NULL) { - ma_dlclose(pContext, kernel32DLL); - return MA_NO_BACKEND; - } + MA_ASSERT(pContext != NULL); + MA_ASSERT(ppAudioClient != NULL); - ma_zero_object(&osvi); - osvi.dwOSVersionInfoSize = sizeof(osvi); - osvi.dwMajorVersion = HIBYTE(MA_WIN32_WINNT_VISTA); - osvi.dwMinorVersion = LOBYTE(MA_WIN32_WINNT_VISTA); - osvi.wServicePackMajor = 1; - if (_VerifyVersionInfoW(&osvi, MA_VER_MAJORVERSION | MA_VER_MINORVERSION | MA_VER_SERVICEPACKMAJOR, _VerSetConditionMask(_VerSetConditionMask(_VerSetConditionMask(0, MA_VER_MAJORVERSION, MA_VER_GREATER_EQUAL), MA_VER_MINORVERSION, MA_VER_GREATER_EQUAL), MA_VER_SERVICEPACKMAJOR, MA_VER_GREATER_EQUAL))) { - result = MA_SUCCESS; + if (pDeviceID != NULL) { + MA_COPY_MEMORY(&iid, pDeviceID->wasapi, sizeof(iid)); + } else { + if (deviceType == ma_device_type_playback) { + iid = MA_IID_DEVINTERFACE_AUDIO_RENDER; } else { - result = MA_NO_BACKEND; + iid = MA_IID_DEVINTERFACE_AUDIO_CAPTURE; } - - ma_dlclose(pContext, kernel32DLL); } + +#if defined(__cplusplus) + hr = StringFromIID(iid, &iidStr); +#else + hr = StringFromIID(&iid, &iidStr); #endif + if (FAILED(hr)) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to convert device IID to string for ActivateAudioInterfaceAsync(). Out of memory.", MA_OUT_OF_MEMORY); + } + result = ma_completion_handler_uwp_init(&completionHandler); if (result != MA_SUCCESS) { - return result; + ma_CoTaskMemFree(pContext, iidStr); + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create event for waiting for ActivateAudioInterfaceAsync().", MA_FAILED_TO_OPEN_BACKEND_DEVICE); } - pContext->onUninit = ma_context_uninit__wasapi; - pContext->onDeviceIDEqual = ma_context_is_device_id_equal__wasapi; - pContext->onEnumDevices = ma_context_enumerate_devices__wasapi; - pContext->onGetDeviceInfo = ma_context_get_device_info__wasapi; - pContext->onDeviceInit = ma_device_init__wasapi; - pContext->onDeviceUninit = ma_device_uninit__wasapi; - pContext->onDeviceStart = NULL; /* Not used. Started in onDeviceMainLoop. */ - pContext->onDeviceStop = ma_device_stop__wasapi; /* Required to ensure the capture event is signalled when stopping a loopback device while nothing is playing. */ - pContext->onDeviceMainLoop = ma_device_main_loop__wasapi; - - return result; -} +#if defined(__cplusplus) + hr = ActivateAudioInterfaceAsync(iidStr, MA_IID_IAudioClient, NULL, (IActivateAudioInterfaceCompletionHandler*)&completionHandler, (IActivateAudioInterfaceAsyncOperation**)&pAsyncOp); +#else + hr = ActivateAudioInterfaceAsync(iidStr, &MA_IID_IAudioClient, NULL, (IActivateAudioInterfaceCompletionHandler*)&completionHandler, (IActivateAudioInterfaceAsyncOperation**)&pAsyncOp); #endif + if (FAILED(hr)) { + ma_completion_handler_uwp_uninit(&completionHandler); + ma_CoTaskMemFree(pContext, iidStr); + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] ActivateAudioInterfaceAsync() failed.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } -/****************************************************************************** + ma_CoTaskMemFree(pContext, iidStr); -DirectSound Backend + /* Wait for the async operation for finish. */ + ma_completion_handler_uwp_wait(&completionHandler); + ma_completion_handler_uwp_uninit(&completionHandler); -******************************************************************************/ -#ifdef MA_HAS_DSOUND -/*#include */ + hr = ma_IActivateAudioInterfaceAsyncOperation_GetActivateResult(pAsyncOp, &activateResult, &pActivatedInterface); + ma_IActivateAudioInterfaceAsyncOperation_Release(pAsyncOp); -GUID MA_GUID_IID_DirectSoundNotify = {0xb0210783, 0x89cd, 0x11d0, {0xaf, 0x08, 0x00, 0xa0, 0xc9, 0x25, 0xcd, 0x16}}; + if (FAILED(hr) || FAILED(activateResult)) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to activate device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } -/* miniaudio only uses priority or exclusive modes. */ -#define MA_DSSCL_NORMAL 1 -#define MA_DSSCL_PRIORITY 2 -#define MA_DSSCL_EXCLUSIVE 3 -#define MA_DSSCL_WRITEPRIMARY 4 + /* Here is where we grab the IAudioClient interface. */ + hr = ma_IUnknown_QueryInterface(pActivatedInterface, &MA_IID_IAudioClient, (void**)ppAudioClient); + if (FAILED(hr)) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to query IAudioClient interface.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } -#define MA_DSCAPS_PRIMARYMONO 0x00000001 -#define MA_DSCAPS_PRIMARYSTEREO 0x00000002 -#define MA_DSCAPS_PRIMARY8BIT 0x00000004 -#define MA_DSCAPS_PRIMARY16BIT 0x00000008 -#define MA_DSCAPS_CONTINUOUSRATE 0x00000010 -#define MA_DSCAPS_EMULDRIVER 0x00000020 -#define MA_DSCAPS_CERTIFIED 0x00000040 -#define MA_DSCAPS_SECONDARYMONO 0x00000100 -#define MA_DSCAPS_SECONDARYSTEREO 0x00000200 -#define MA_DSCAPS_SECONDARY8BIT 0x00000400 -#define MA_DSCAPS_SECONDARY16BIT 0x00000800 + if (ppActivatedInterface) { + *ppActivatedInterface = pActivatedInterface; + } else { + ma_IUnknown_Release(pActivatedInterface); + } -#define MA_DSBCAPS_PRIMARYBUFFER 0x00000001 -#define MA_DSBCAPS_STATIC 0x00000002 -#define MA_DSBCAPS_LOCHARDWARE 0x00000004 -#define MA_DSBCAPS_LOCSOFTWARE 0x00000008 -#define MA_DSBCAPS_CTRL3D 0x00000010 -#define MA_DSBCAPS_CTRLFREQUENCY 0x00000020 -#define MA_DSBCAPS_CTRLPAN 0x00000040 -#define MA_DSBCAPS_CTRLVOLUME 0x00000080 -#define MA_DSBCAPS_CTRLPOSITIONNOTIFY 0x00000100 -#define MA_DSBCAPS_CTRLFX 0x00000200 -#define MA_DSBCAPS_STICKYFOCUS 0x00004000 -#define MA_DSBCAPS_GLOBALFOCUS 0x00008000 -#define MA_DSBCAPS_GETCURRENTPOSITION2 0x00010000 -#define MA_DSBCAPS_MUTE3DATMAXDISTANCE 0x00020000 -#define MA_DSBCAPS_LOCDEFER 0x00040000 -#define MA_DSBCAPS_TRUEPLAYPOSITION 0x00080000 + return MA_SUCCESS; +} +#endif -#define MA_DSBPLAY_LOOPING 0x00000001 -#define MA_DSBPLAY_LOCHARDWARE 0x00000002 -#define MA_DSBPLAY_LOCSOFTWARE 0x00000004 -#define MA_DSBPLAY_TERMINATEBY_TIME 0x00000008 -#define MA_DSBPLAY_TERMINATEBY_DISTANCE 0x00000010 -#define MA_DSBPLAY_TERMINATEBY_PRIORITY 0x00000020 +static ma_result ma_context_get_IAudioClient__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_IAudioClient** ppAudioClient, ma_WASAPIDeviceInterface** ppDeviceInterface) +{ +#ifdef MA_WIN32_DESKTOP + return ma_context_get_IAudioClient_Desktop__wasapi(pContext, deviceType, pDeviceID, ppAudioClient, ppDeviceInterface); +#else + return ma_context_get_IAudioClient_UWP__wasapi(pContext, deviceType, pDeviceID, ppAudioClient, ppDeviceInterface); +#endif +} -#define MA_DSCBSTART_LOOPING 0x00000001 -typedef struct +static ma_result ma_context_enumerate_devices__wasapi(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { - DWORD dwSize; - DWORD dwFlags; - DWORD dwBufferBytes; - DWORD dwReserved; - WAVEFORMATEX* lpwfxFormat; - GUID guid3DAlgorithm; -} MA_DSBUFFERDESC; + /* Different enumeration for desktop and UWP. */ +#ifdef MA_WIN32_DESKTOP + /* Desktop */ + HRESULT hr; + ma_IMMDeviceEnumerator* pDeviceEnumerator; -typedef struct -{ - DWORD dwSize; - DWORD dwFlags; - DWORD dwBufferBytes; - DWORD dwReserved; - WAVEFORMATEX* lpwfxFormat; - DWORD dwFXCount; - void* lpDSCFXDesc; /* <-- miniaudio doesn't use this, so set to void*. */ -} MA_DSCBUFFERDESC; + hr = ma_CoCreateInstance(pContext, MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator); + if (FAILED(hr)) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create device enumerator.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } -typedef struct -{ - DWORD dwSize; - DWORD dwFlags; - DWORD dwMinSecondarySampleRate; - DWORD dwMaxSecondarySampleRate; - DWORD dwPrimaryBuffers; - DWORD dwMaxHwMixingAllBuffers; - DWORD dwMaxHwMixingStaticBuffers; - DWORD dwMaxHwMixingStreamingBuffers; - DWORD dwFreeHwMixingAllBuffers; - DWORD dwFreeHwMixingStaticBuffers; - DWORD dwFreeHwMixingStreamingBuffers; - DWORD dwMaxHw3DAllBuffers; - DWORD dwMaxHw3DStaticBuffers; - DWORD dwMaxHw3DStreamingBuffers; - DWORD dwFreeHw3DAllBuffers; - DWORD dwFreeHw3DStaticBuffers; - DWORD dwFreeHw3DStreamingBuffers; - DWORD dwTotalHwMemBytes; - DWORD dwFreeHwMemBytes; - DWORD dwMaxContigFreeHwMemBytes; - DWORD dwUnlockTransferRateHwBuffers; - DWORD dwPlayCpuOverheadSwBuffers; - DWORD dwReserved1; - DWORD dwReserved2; -} MA_DSCAPS; + ma_context_enumerate_devices_by_type__wasapi(pContext, pDeviceEnumerator, ma_device_type_playback, callback, pUserData); + ma_context_enumerate_devices_by_type__wasapi(pContext, pDeviceEnumerator, ma_device_type_capture, callback, pUserData); + + ma_IMMDeviceEnumerator_Release(pDeviceEnumerator); +#else + /* + UWP + + The MMDevice API is only supported on desktop applications. For now, while I'm still figuring out how to properly enumerate + over devices without using MMDevice, I'm restricting devices to defaults. + + Hint: DeviceInformation::FindAllAsync() with DeviceClass.AudioCapture/AudioRender. https://blogs.windows.com/buildingapps/2014/05/15/real-time-audio-in-windows-store-and-windows-phone-apps/ + */ + if (callback) { + ma_bool32 cbResult = MA_TRUE; + + /* Playback. */ + if (cbResult) { + ma_device_info deviceInfo; + MA_ZERO_OBJECT(&deviceInfo); + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); + deviceInfo._private.isDefault = MA_TRUE; + cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); + } -typedef struct -{ - DWORD dwSize; - DWORD dwFlags; - DWORD dwBufferBytes; - DWORD dwUnlockTransferRate; - DWORD dwPlayCpuOverhead; -} MA_DSBCAPS; + /* Capture. */ + if (cbResult) { + ma_device_info deviceInfo; + MA_ZERO_OBJECT(&deviceInfo); + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); + deviceInfo._private.isDefault = MA_TRUE; + cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); + } + } +#endif -typedef struct -{ - DWORD dwSize; - DWORD dwFlags; - DWORD dwFormats; - DWORD dwChannels; -} MA_DSCCAPS; + return MA_SUCCESS; +} -typedef struct +static ma_result ma_context_get_device_info__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) { - DWORD dwSize; - DWORD dwFlags; - DWORD dwBufferBytes; - DWORD dwReserved; -} MA_DSCBCAPS; +#ifdef MA_WIN32_DESKTOP + ma_result result; + ma_IMMDevice* pMMDevice = NULL; + LPWSTR pDefaultDeviceID = NULL; + + result = ma_context_get_MMDevice__wasapi(pContext, deviceType, pDeviceID, &pMMDevice); + if (result != MA_SUCCESS) { + return result; + } -typedef struct -{ - DWORD dwOffset; - HANDLE hEventNotify; -} MA_DSBPOSITIONNOTIFY; + /* We need the default device ID so we can set the isDefault flag in the device info. */ + pDefaultDeviceID = ma_context_get_default_device_id__wasapi(pContext, deviceType); -typedef struct ma_IDirectSound ma_IDirectSound; -typedef struct ma_IDirectSoundBuffer ma_IDirectSoundBuffer; -typedef struct ma_IDirectSoundCapture ma_IDirectSoundCapture; -typedef struct ma_IDirectSoundCaptureBuffer ma_IDirectSoundCaptureBuffer; -typedef struct ma_IDirectSoundNotify ma_IDirectSoundNotify; + result = ma_context_get_device_info_from_MMDevice__wasapi(pContext, pMMDevice, shareMode, pDefaultDeviceID, MA_FALSE, pDeviceInfo); /* MA_FALSE = !onlySimpleInfo. */ + if (pDefaultDeviceID != NULL) { + ma_CoTaskMemFree(pContext, pDefaultDeviceID); + pDefaultDeviceID = NULL; + } -/* -COM objects. The way these work is that you have a vtable (a list of function pointers, kind of -like how C++ works internally), and then you have a structure with a single member, which is a -pointer to the vtable. The vtable is where the methods of the object are defined. Methods need -to be in a specific order, and parent classes need to have their methods declared first. -*/ + ma_IMMDevice_Release(pMMDevice); -/* IDirectSound */ -typedef struct -{ - /* IUnknown */ - HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSound* pThis, const IID* const riid, void** ppObject); - ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSound* pThis); - ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSound* pThis); + return result; +#else + ma_IAudioClient* pAudioClient; + ma_result result; - /* IDirectSound */ - HRESULT (STDMETHODCALLTYPE * CreateSoundBuffer) (ma_IDirectSound* pThis, const MA_DSBUFFERDESC* pDSBufferDesc, ma_IDirectSoundBuffer** ppDSBuffer, void* pUnkOuter); - HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSound* pThis, MA_DSCAPS* pDSCaps); - HRESULT (STDMETHODCALLTYPE * DuplicateSoundBuffer)(ma_IDirectSound* pThis, ma_IDirectSoundBuffer* pDSBufferOriginal, ma_IDirectSoundBuffer** ppDSBufferDuplicate); - HRESULT (STDMETHODCALLTYPE * SetCooperativeLevel) (ma_IDirectSound* pThis, HWND hwnd, DWORD dwLevel); - HRESULT (STDMETHODCALLTYPE * Compact) (ma_IDirectSound* pThis); - HRESULT (STDMETHODCALLTYPE * GetSpeakerConfig) (ma_IDirectSound* pThis, DWORD* pSpeakerConfig); - HRESULT (STDMETHODCALLTYPE * SetSpeakerConfig) (ma_IDirectSound* pThis, DWORD dwSpeakerConfig); - HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSound* pThis, const GUID* pGuidDevice); -} ma_IDirectSoundVtbl; -struct ma_IDirectSound -{ - ma_IDirectSoundVtbl* lpVtbl; -}; -HRESULT ma_IDirectSound_QueryInterface(ma_IDirectSound* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } -ULONG ma_IDirectSound_AddRef(ma_IDirectSound* pThis) { return pThis->lpVtbl->AddRef(pThis); } -ULONG ma_IDirectSound_Release(ma_IDirectSound* pThis) { return pThis->lpVtbl->Release(pThis); } -HRESULT ma_IDirectSound_CreateSoundBuffer(ma_IDirectSound* pThis, const MA_DSBUFFERDESC* pDSBufferDesc, ma_IDirectSoundBuffer** ppDSBuffer, void* pUnkOuter) { return pThis->lpVtbl->CreateSoundBuffer(pThis, pDSBufferDesc, ppDSBuffer, pUnkOuter); } -HRESULT ma_IDirectSound_GetCaps(ma_IDirectSound* pThis, MA_DSCAPS* pDSCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSCaps); } -HRESULT ma_IDirectSound_DuplicateSoundBuffer(ma_IDirectSound* pThis, ma_IDirectSoundBuffer* pDSBufferOriginal, ma_IDirectSoundBuffer** ppDSBufferDuplicate) { return pThis->lpVtbl->DuplicateSoundBuffer(pThis, pDSBufferOriginal, ppDSBufferDuplicate); } -HRESULT ma_IDirectSound_SetCooperativeLevel(ma_IDirectSound* pThis, HWND hwnd, DWORD dwLevel) { return pThis->lpVtbl->SetCooperativeLevel(pThis, hwnd, dwLevel); } -HRESULT ma_IDirectSound_Compact(ma_IDirectSound* pThis) { return pThis->lpVtbl->Compact(pThis); } -HRESULT ma_IDirectSound_GetSpeakerConfig(ma_IDirectSound* pThis, DWORD* pSpeakerConfig) { return pThis->lpVtbl->GetSpeakerConfig(pThis, pSpeakerConfig); } -HRESULT ma_IDirectSound_SetSpeakerConfig(ma_IDirectSound* pThis, DWORD dwSpeakerConfig) { return pThis->lpVtbl->SetSpeakerConfig(pThis, dwSpeakerConfig); } -HRESULT ma_IDirectSound_Initialize(ma_IDirectSound* pThis, const GUID* pGuidDevice) { return pThis->lpVtbl->Initialize(pThis, pGuidDevice); } + /* UWP currently only uses default devices. */ + if (deviceType == ma_device_type_playback) { + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); + } else { + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); + } + /* Not currently supporting exclusive mode on UWP. */ + if (shareMode == ma_share_mode_exclusive) { + return MA_ERROR; + } -/* IDirectSoundBuffer */ -typedef struct -{ - /* IUnknown */ - HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundBuffer* pThis, const IID* const riid, void** ppObject); - ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundBuffer* pThis); - ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundBuffer* pThis); + result = ma_context_get_IAudioClient_UWP__wasapi(pContext, deviceType, pDeviceID, &pAudioClient, NULL); + if (result != MA_SUCCESS) { + return result; + } - /* IDirectSoundBuffer */ - HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSoundBuffer* pThis, MA_DSBCAPS* pDSBufferCaps); - HRESULT (STDMETHODCALLTYPE * GetCurrentPosition)(ma_IDirectSoundBuffer* pThis, DWORD* pCurrentPlayCursor, DWORD* pCurrentWriteCursor); - HRESULT (STDMETHODCALLTYPE * GetFormat) (ma_IDirectSoundBuffer* pThis, WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten); - HRESULT (STDMETHODCALLTYPE * GetVolume) (ma_IDirectSoundBuffer* pThis, LONG* pVolume); - HRESULT (STDMETHODCALLTYPE * GetPan) (ma_IDirectSoundBuffer* pThis, LONG* pPan); - HRESULT (STDMETHODCALLTYPE * GetFrequency) (ma_IDirectSoundBuffer* pThis, DWORD* pFrequency); - HRESULT (STDMETHODCALLTYPE * GetStatus) (ma_IDirectSoundBuffer* pThis, DWORD* pStatus); - HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSoundBuffer* pThis, ma_IDirectSound* pDirectSound, const MA_DSBUFFERDESC* pDSBufferDesc); - HRESULT (STDMETHODCALLTYPE * Lock) (ma_IDirectSoundBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags); - HRESULT (STDMETHODCALLTYPE * Play) (ma_IDirectSoundBuffer* pThis, DWORD dwReserved1, DWORD dwPriority, DWORD dwFlags); - HRESULT (STDMETHODCALLTYPE * SetCurrentPosition)(ma_IDirectSoundBuffer* pThis, DWORD dwNewPosition); - HRESULT (STDMETHODCALLTYPE * SetFormat) (ma_IDirectSoundBuffer* pThis, const WAVEFORMATEX* pFormat); - HRESULT (STDMETHODCALLTYPE * SetVolume) (ma_IDirectSoundBuffer* pThis, LONG volume); - HRESULT (STDMETHODCALLTYPE * SetPan) (ma_IDirectSoundBuffer* pThis, LONG pan); - HRESULT (STDMETHODCALLTYPE * SetFrequency) (ma_IDirectSoundBuffer* pThis, DWORD dwFrequency); - HRESULT (STDMETHODCALLTYPE * Stop) (ma_IDirectSoundBuffer* pThis); - HRESULT (STDMETHODCALLTYPE * Unlock) (ma_IDirectSoundBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2); - HRESULT (STDMETHODCALLTYPE * Restore) (ma_IDirectSoundBuffer* pThis); -} ma_IDirectSoundBufferVtbl; -struct ma_IDirectSoundBuffer -{ - ma_IDirectSoundBufferVtbl* lpVtbl; -}; -HRESULT ma_IDirectSoundBuffer_QueryInterface(ma_IDirectSoundBuffer* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } -ULONG ma_IDirectSoundBuffer_AddRef(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->AddRef(pThis); } -ULONG ma_IDirectSoundBuffer_Release(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->Release(pThis); } -HRESULT ma_IDirectSoundBuffer_GetCaps(ma_IDirectSoundBuffer* pThis, MA_DSBCAPS* pDSBufferCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSBufferCaps); } -HRESULT ma_IDirectSoundBuffer_GetCurrentPosition(ma_IDirectSoundBuffer* pThis, DWORD* pCurrentPlayCursor, DWORD* pCurrentWriteCursor) { return pThis->lpVtbl->GetCurrentPosition(pThis, pCurrentPlayCursor, pCurrentWriteCursor); } -HRESULT ma_IDirectSoundBuffer_GetFormat(ma_IDirectSoundBuffer* pThis, WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten) { return pThis->lpVtbl->GetFormat(pThis, pFormat, dwSizeAllocated, pSizeWritten); } -HRESULT ma_IDirectSoundBuffer_GetVolume(ma_IDirectSoundBuffer* pThis, LONG* pVolume) { return pThis->lpVtbl->GetVolume(pThis, pVolume); } -HRESULT ma_IDirectSoundBuffer_GetPan(ma_IDirectSoundBuffer* pThis, LONG* pPan) { return pThis->lpVtbl->GetPan(pThis, pPan); } -HRESULT ma_IDirectSoundBuffer_GetFrequency(ma_IDirectSoundBuffer* pThis, DWORD* pFrequency) { return pThis->lpVtbl->GetFrequency(pThis, pFrequency); } -HRESULT ma_IDirectSoundBuffer_GetStatus(ma_IDirectSoundBuffer* pThis, DWORD* pStatus) { return pThis->lpVtbl->GetStatus(pThis, pStatus); } -HRESULT ma_IDirectSoundBuffer_Initialize(ma_IDirectSoundBuffer* pThis, ma_IDirectSound* pDirectSound, const MA_DSBUFFERDESC* pDSBufferDesc) { return pThis->lpVtbl->Initialize(pThis, pDirectSound, pDSBufferDesc); } -HRESULT ma_IDirectSoundBuffer_Lock(ma_IDirectSoundBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags) { return pThis->lpVtbl->Lock(pThis, dwOffset, dwBytes, ppAudioPtr1, pAudioBytes1, ppAudioPtr2, pAudioBytes2, dwFlags); } -HRESULT ma_IDirectSoundBuffer_Play(ma_IDirectSoundBuffer* pThis, DWORD dwReserved1, DWORD dwPriority, DWORD dwFlags) { return pThis->lpVtbl->Play(pThis, dwReserved1, dwPriority, dwFlags); } -HRESULT ma_IDirectSoundBuffer_SetCurrentPosition(ma_IDirectSoundBuffer* pThis, DWORD dwNewPosition) { return pThis->lpVtbl->SetCurrentPosition(pThis, dwNewPosition); } -HRESULT ma_IDirectSoundBuffer_SetFormat(ma_IDirectSoundBuffer* pThis, const WAVEFORMATEX* pFormat) { return pThis->lpVtbl->SetFormat(pThis, pFormat); } -HRESULT ma_IDirectSoundBuffer_SetVolume(ma_IDirectSoundBuffer* pThis, LONG volume) { return pThis->lpVtbl->SetVolume(pThis, volume); } -HRESULT ma_IDirectSoundBuffer_SetPan(ma_IDirectSoundBuffer* pThis, LONG pan) { return pThis->lpVtbl->SetPan(pThis, pan); } -HRESULT ma_IDirectSoundBuffer_SetFrequency(ma_IDirectSoundBuffer* pThis, DWORD dwFrequency) { return pThis->lpVtbl->SetFrequency(pThis, dwFrequency); } -HRESULT ma_IDirectSoundBuffer_Stop(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->Stop(pThis); } -HRESULT ma_IDirectSoundBuffer_Unlock(ma_IDirectSoundBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2) { return pThis->lpVtbl->Unlock(pThis, pAudioPtr1, dwAudioBytes1, pAudioPtr2, dwAudioBytes2); } -HRESULT ma_IDirectSoundBuffer_Restore(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->Restore(pThis); } + result = ma_context_get_device_info_from_IAudioClient__wasapi(pContext, NULL, pAudioClient, shareMode, pDeviceInfo); + pDeviceInfo->_private.isDefault = MA_TRUE; /* UWP only supports default devices. */ -/* IDirectSoundCapture */ -typedef struct -{ - /* IUnknown */ - HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundCapture* pThis, const IID* const riid, void** ppObject); - ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundCapture* pThis); - ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundCapture* pThis); + ma_IAudioClient_Release(pAudioClient); + return result; +#endif +} - /* IDirectSoundCapture */ - HRESULT (STDMETHODCALLTYPE * CreateCaptureBuffer)(ma_IDirectSoundCapture* pThis, const MA_DSCBUFFERDESC* pDSCBufferDesc, ma_IDirectSoundCaptureBuffer** ppDSCBuffer, void* pUnkOuter); - HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSoundCapture* pThis, MA_DSCCAPS* pDSCCaps); - HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSoundCapture* pThis, const GUID* pGuidDevice); -} ma_IDirectSoundCaptureVtbl; -struct ma_IDirectSoundCapture +static void ma_device_uninit__wasapi(ma_device* pDevice) { - ma_IDirectSoundCaptureVtbl* lpVtbl; -}; -HRESULT ma_IDirectSoundCapture_QueryInterface(ma_IDirectSoundCapture* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } -ULONG ma_IDirectSoundCapture_AddRef(ma_IDirectSoundCapture* pThis) { return pThis->lpVtbl->AddRef(pThis); } -ULONG ma_IDirectSoundCapture_Release(ma_IDirectSoundCapture* pThis) { return pThis->lpVtbl->Release(pThis); } -HRESULT ma_IDirectSoundCapture_CreateCaptureBuffer(ma_IDirectSoundCapture* pThis, const MA_DSCBUFFERDESC* pDSCBufferDesc, ma_IDirectSoundCaptureBuffer** ppDSCBuffer, void* pUnkOuter) { return pThis->lpVtbl->CreateCaptureBuffer(pThis, pDSCBufferDesc, ppDSCBuffer, pUnkOuter); } -HRESULT ma_IDirectSoundCapture_GetCaps (ma_IDirectSoundCapture* pThis, MA_DSCCAPS* pDSCCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSCCaps); } -HRESULT ma_IDirectSoundCapture_Initialize (ma_IDirectSoundCapture* pThis, const GUID* pGuidDevice) { return pThis->lpVtbl->Initialize(pThis, pGuidDevice); } + MA_ASSERT(pDevice != NULL); +#ifdef MA_WIN32_DESKTOP + if (pDevice->wasapi.pDeviceEnumerator) { + ((ma_IMMDeviceEnumerator*)pDevice->wasapi.pDeviceEnumerator)->lpVtbl->UnregisterEndpointNotificationCallback((ma_IMMDeviceEnumerator*)pDevice->wasapi.pDeviceEnumerator, &pDevice->wasapi.notificationClient); + ma_IMMDeviceEnumerator_Release((ma_IMMDeviceEnumerator*)pDevice->wasapi.pDeviceEnumerator); + } +#endif -/* IDirectSoundCaptureBuffer */ -typedef struct -{ - /* IUnknown */ - HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundCaptureBuffer* pThis, const IID* const riid, void** ppObject); - ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundCaptureBuffer* pThis); - ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundCaptureBuffer* pThis); + if (pDevice->wasapi.pRenderClient) { + ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient); + } + if (pDevice->wasapi.pCaptureClient) { + ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient); + } - /* IDirectSoundCaptureBuffer */ - HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSoundCaptureBuffer* pThis, MA_DSCBCAPS* pDSCBCaps); - HRESULT (STDMETHODCALLTYPE * GetCurrentPosition)(ma_IDirectSoundCaptureBuffer* pThis, DWORD* pCapturePosition, DWORD* pReadPosition); - HRESULT (STDMETHODCALLTYPE * GetFormat) (ma_IDirectSoundCaptureBuffer* pThis, WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten); - HRESULT (STDMETHODCALLTYPE * GetStatus) (ma_IDirectSoundCaptureBuffer* pThis, DWORD* pStatus); - HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSoundCaptureBuffer* pThis, ma_IDirectSoundCapture* pDirectSoundCapture, const MA_DSCBUFFERDESC* pDSCBufferDesc); - HRESULT (STDMETHODCALLTYPE * Lock) (ma_IDirectSoundCaptureBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags); - HRESULT (STDMETHODCALLTYPE * Start) (ma_IDirectSoundCaptureBuffer* pThis, DWORD dwFlags); - HRESULT (STDMETHODCALLTYPE * Stop) (ma_IDirectSoundCaptureBuffer* pThis); - HRESULT (STDMETHODCALLTYPE * Unlock) (ma_IDirectSoundCaptureBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2); -} ma_IDirectSoundCaptureBufferVtbl; -struct ma_IDirectSoundCaptureBuffer -{ - ma_IDirectSoundCaptureBufferVtbl* lpVtbl; -}; -HRESULT ma_IDirectSoundCaptureBuffer_QueryInterface(ma_IDirectSoundCaptureBuffer* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } -ULONG ma_IDirectSoundCaptureBuffer_AddRef(ma_IDirectSoundCaptureBuffer* pThis) { return pThis->lpVtbl->AddRef(pThis); } -ULONG ma_IDirectSoundCaptureBuffer_Release(ma_IDirectSoundCaptureBuffer* pThis) { return pThis->lpVtbl->Release(pThis); } -HRESULT ma_IDirectSoundCaptureBuffer_GetCaps(ma_IDirectSoundCaptureBuffer* pThis, MA_DSCBCAPS* pDSCBCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSCBCaps); } -HRESULT ma_IDirectSoundCaptureBuffer_GetCurrentPosition(ma_IDirectSoundCaptureBuffer* pThis, DWORD* pCapturePosition, DWORD* pReadPosition) { return pThis->lpVtbl->GetCurrentPosition(pThis, pCapturePosition, pReadPosition); } -HRESULT ma_IDirectSoundCaptureBuffer_GetFormat(ma_IDirectSoundCaptureBuffer* pThis, WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten) { return pThis->lpVtbl->GetFormat(pThis, pFormat, dwSizeAllocated, pSizeWritten); } -HRESULT ma_IDirectSoundCaptureBuffer_GetStatus(ma_IDirectSoundCaptureBuffer* pThis, DWORD* pStatus) { return pThis->lpVtbl->GetStatus(pThis, pStatus); } -HRESULT ma_IDirectSoundCaptureBuffer_Initialize(ma_IDirectSoundCaptureBuffer* pThis, ma_IDirectSoundCapture* pDirectSoundCapture, const MA_DSCBUFFERDESC* pDSCBufferDesc) { return pThis->lpVtbl->Initialize(pThis, pDirectSoundCapture, pDSCBufferDesc); } -HRESULT ma_IDirectSoundCaptureBuffer_Lock(ma_IDirectSoundCaptureBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags) { return pThis->lpVtbl->Lock(pThis, dwOffset, dwBytes, ppAudioPtr1, pAudioBytes1, ppAudioPtr2, pAudioBytes2, dwFlags); } -HRESULT ma_IDirectSoundCaptureBuffer_Start(ma_IDirectSoundCaptureBuffer* pThis, DWORD dwFlags) { return pThis->lpVtbl->Start(pThis, dwFlags); } -HRESULT ma_IDirectSoundCaptureBuffer_Stop(ma_IDirectSoundCaptureBuffer* pThis) { return pThis->lpVtbl->Stop(pThis); } -HRESULT ma_IDirectSoundCaptureBuffer_Unlock(ma_IDirectSoundCaptureBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2) { return pThis->lpVtbl->Unlock(pThis, pAudioPtr1, dwAudioBytes1, pAudioPtr2, dwAudioBytes2); } + if (pDevice->wasapi.pAudioClientPlayback) { + ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); + } + if (pDevice->wasapi.pAudioClientCapture) { + ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); + } + if (pDevice->wasapi.hEventPlayback) { + CloseHandle(pDevice->wasapi.hEventPlayback); + } + if (pDevice->wasapi.hEventCapture) { + CloseHandle(pDevice->wasapi.hEventCapture); + } +} -/* IDirectSoundNotify */ -typedef struct -{ - /* IUnknown */ - HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundNotify* pThis, const IID* const riid, void** ppObject); - ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundNotify* pThis); - ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundNotify* pThis); - /* IDirectSoundNotify */ - HRESULT (STDMETHODCALLTYPE * SetNotificationPositions)(ma_IDirectSoundNotify* pThis, DWORD dwPositionNotifies, const MA_DSBPOSITIONNOTIFY* pPositionNotifies); -} ma_IDirectSoundNotifyVtbl; -struct ma_IDirectSoundNotify +typedef struct { - ma_IDirectSoundNotifyVtbl* lpVtbl; -}; -HRESULT ma_IDirectSoundNotify_QueryInterface(ma_IDirectSoundNotify* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } -ULONG ma_IDirectSoundNotify_AddRef(ma_IDirectSoundNotify* pThis) { return pThis->lpVtbl->AddRef(pThis); } -ULONG ma_IDirectSoundNotify_Release(ma_IDirectSoundNotify* pThis) { return pThis->lpVtbl->Release(pThis); } -HRESULT ma_IDirectSoundNotify_SetNotificationPositions(ma_IDirectSoundNotify* pThis, DWORD dwPositionNotifies, const MA_DSBPOSITIONNOTIFY* pPositionNotifies) { return pThis->lpVtbl->SetNotificationPositions(pThis, dwPositionNotifies, pPositionNotifies); } - + /* Input. */ + ma_format formatIn; + ma_uint32 channelsIn; + ma_uint32 sampleRateIn; + ma_channel channelMapIn[MA_MAX_CHANNELS]; + ma_uint32 periodSizeInFramesIn; + ma_uint32 periodSizeInMillisecondsIn; + ma_uint32 periodsIn; + ma_bool32 usingDefaultFormat; + ma_bool32 usingDefaultChannels; + ma_bool32 usingDefaultSampleRate; + ma_bool32 usingDefaultChannelMap; + ma_share_mode shareMode; + ma_bool32 noAutoConvertSRC; + ma_bool32 noDefaultQualitySRC; + ma_bool32 noHardwareOffloading; -typedef BOOL (CALLBACK * ma_DSEnumCallbackAProc) (LPGUID pDeviceGUID, LPCSTR pDeviceDescription, LPCSTR pModule, LPVOID pContext); -typedef HRESULT (WINAPI * ma_DirectSoundCreateProc) (const GUID* pcGuidDevice, ma_IDirectSound** ppDS8, LPUNKNOWN pUnkOuter); -typedef HRESULT (WINAPI * ma_DirectSoundEnumerateAProc) (ma_DSEnumCallbackAProc pDSEnumCallback, LPVOID pContext); -typedef HRESULT (WINAPI * ma_DirectSoundCaptureCreateProc) (const GUID* pcGuidDevice, ma_IDirectSoundCapture** ppDSC8, LPUNKNOWN pUnkOuter); -typedef HRESULT (WINAPI * ma_DirectSoundCaptureEnumerateAProc)(ma_DSEnumCallbackAProc pDSEnumCallback, LPVOID pContext); + /* Output. */ + ma_IAudioClient* pAudioClient; + ma_IAudioRenderClient* pRenderClient; + ma_IAudioCaptureClient* pCaptureClient; + ma_format formatOut; + ma_uint32 channelsOut; + ma_uint32 sampleRateOut; + ma_channel channelMapOut[MA_MAX_CHANNELS]; + ma_uint32 periodSizeInFramesOut; + ma_uint32 periodsOut; + ma_bool32 usingAudioClient3; + char deviceName[256]; +} ma_device_init_internal_data__wasapi; +static ma_result ma_device_init_internal__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_init_internal_data__wasapi* pData) +{ + HRESULT hr; + ma_result result = MA_SUCCESS; + const char* errorMsg = ""; + MA_AUDCLNT_SHAREMODE shareMode = MA_AUDCLNT_SHAREMODE_SHARED; + DWORD streamFlags = 0; + MA_REFERENCE_TIME periodDurationInMicroseconds; + ma_bool32 wasInitializedUsingIAudioClient3 = MA_FALSE; + WAVEFORMATEXTENSIBLE wf = {0}; + ma_WASAPIDeviceInterface* pDeviceInterface = NULL; + ma_IAudioClient2* pAudioClient2; + ma_uint32 nativeSampleRate; -/* -Retrieves the channel count and channel map for the given speaker configuration. If the speaker configuration is unknown, -the channel count and channel map will be left unmodified. -*/ -void ma_get_channels_from_speaker_config__dsound(DWORD speakerConfig, WORD* pChannelsOut, DWORD* pChannelMapOut) -{ - WORD channels; - DWORD channelMap; + MA_ASSERT(pContext != NULL); + MA_ASSERT(pData != NULL); - channels = 0; - if (pChannelsOut != NULL) { - channels = *pChannelsOut; + /* This function is only used to initialize one device type: either playback, capture or loopback. Never full-duplex. */ + if (deviceType == ma_device_type_duplex) { + return MA_INVALID_ARGS; } - channelMap = 0; - if (pChannelMapOut != NULL) { - channelMap = *pChannelMapOut; - } + pData->pAudioClient = NULL; + pData->pRenderClient = NULL; + pData->pCaptureClient = NULL; - /* - The speaker configuration is a combination of speaker config and speaker geometry. The lower 8 bits is what we care about. The upper - 16 bits is for the geometry. - */ - switch ((BYTE)(speakerConfig)) { - case 1 /*DSSPEAKER_HEADPHONE*/: channels = 2; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT; break; - case 2 /*DSSPEAKER_MONO*/: channels = 1; channelMap = SPEAKER_FRONT_CENTER; break; - case 3 /*DSSPEAKER_QUAD*/: channels = 4; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT; break; - case 4 /*DSSPEAKER_STEREO*/: channels = 2; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT; break; - case 5 /*DSSPEAKER_SURROUND*/: channels = 4; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_BACK_CENTER; break; - case 6 /*DSSPEAKER_5POINT1_BACK*/ /*DSSPEAKER_5POINT1*/: channels = 6; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT; break; - case 7 /*DSSPEAKER_7POINT1_WIDE*/ /*DSSPEAKER_7POINT1*/: channels = 8; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_FRONT_LEFT_OF_CENTER | SPEAKER_FRONT_RIGHT_OF_CENTER; break; - case 8 /*DSSPEAKER_7POINT1_SURROUND*/: channels = 8; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT; break; - case 9 /*DSSPEAKER_5POINT1_SURROUND*/: channels = 6; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT; break; - default: break; + streamFlags = MA_AUDCLNT_STREAMFLAGS_EVENTCALLBACK; + if (!pData->noAutoConvertSRC && !pData->usingDefaultSampleRate && pData->shareMode != ma_share_mode_exclusive) { /* <-- Exclusive streams must use the native sample rate. */ + streamFlags |= MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM; } - - if (pChannelsOut != NULL) { - *pChannelsOut = channels; + if (!pData->noDefaultQualitySRC && !pData->usingDefaultSampleRate && (streamFlags & MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM) != 0) { + streamFlags |= MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY; } - - if (pChannelMapOut != NULL) { - *pChannelMapOut = channelMap; + if (deviceType == ma_device_type_loopback) { + streamFlags |= MA_AUDCLNT_STREAMFLAGS_LOOPBACK; } -} - -ma_result ma_context_create_IDirectSound__dsound(ma_context* pContext, ma_share_mode shareMode, const ma_device_id* pDeviceID, ma_IDirectSound** ppDirectSound) -{ - ma_IDirectSound* pDirectSound; - HWND hWnd; + result = ma_context_get_IAudioClient__wasapi(pContext, deviceType, pDeviceID, &pData->pAudioClient, &pDeviceInterface); + if (result != MA_SUCCESS) { + goto done; + } - ma_assert(pContext != NULL); - ma_assert(ppDirectSound != NULL); - *ppDirectSound = NULL; - pDirectSound = NULL; + /* Try enabling hardware offloading. */ + if (!pData->noHardwareOffloading) { + hr = ma_IAudioClient_QueryInterface(pData->pAudioClient, &MA_IID_IAudioClient2, (void**)&pAudioClient2); + if (SUCCEEDED(hr)) { + BOOL isHardwareOffloadingSupported = 0; + hr = ma_IAudioClient2_IsOffloadCapable(pAudioClient2, MA_AudioCategory_Other, &isHardwareOffloadingSupported); + if (SUCCEEDED(hr) && isHardwareOffloadingSupported) { + ma_AudioClientProperties clientProperties; + MA_ZERO_OBJECT(&clientProperties); + clientProperties.cbSize = sizeof(clientProperties); + clientProperties.bIsOffload = 1; + clientProperties.eCategory = MA_AudioCategory_Other; + ma_IAudioClient2_SetClientProperties(pAudioClient2, &clientProperties); + } - if (FAILED(((ma_DirectSoundCreateProc)pContext->dsound.DirectSoundCreate)((pDeviceID == NULL) ? NULL : (const GUID*)pDeviceID->dsound, &pDirectSound, NULL))) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[DirectSound] DirectSoundCreate() failed for playback device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + pAudioClient2->lpVtbl->Release(pAudioClient2); + } } - /* The cooperative level must be set before doing anything else. */ - hWnd = ((MA_PFN_GetForegroundWindow)pContext->win32.GetForegroundWindow)(); - if (hWnd == NULL) { - hWnd = ((MA_PFN_GetDesktopWindow)pContext->win32.GetDesktopWindow)(); - } - if (FAILED(ma_IDirectSound_SetCooperativeLevel(pDirectSound, hWnd, (shareMode == ma_share_mode_exclusive) ? MA_DSSCL_EXCLUSIVE : MA_DSSCL_PRIORITY))) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_SetCooperateiveLevel() failed for playback device.", MA_SHARE_MODE_NOT_SUPPORTED); - } + /* Here is where we try to determine the best format to use with the device. If the client if wanting exclusive mode, first try finding the best format for that. If this fails, fall back to shared mode. */ + result = MA_FORMAT_NOT_SUPPORTED; + if (pData->shareMode == ma_share_mode_exclusive) { + #ifdef MA_WIN32_DESKTOP + /* In exclusive mode on desktop we always use the backend's native format. */ + ma_IPropertyStore* pStore = NULL; + hr = ma_IMMDevice_OpenPropertyStore(pDeviceInterface, STGM_READ, &pStore); + if (SUCCEEDED(hr)) { + PROPVARIANT prop; + ma_PropVariantInit(&prop); + hr = ma_IPropertyStore_GetValue(pStore, &MA_PKEY_AudioEngine_DeviceFormat, &prop); + if (SUCCEEDED(hr)) { + WAVEFORMATEX* pActualFormat = (WAVEFORMATEX*)prop.blob.pBlobData; + hr = ma_IAudioClient_IsFormatSupported((ma_IAudioClient*)pData->pAudioClient, MA_AUDCLNT_SHAREMODE_EXCLUSIVE, pActualFormat, NULL); + if (SUCCEEDED(hr)) { + MA_COPY_MEMORY(&wf, pActualFormat, sizeof(WAVEFORMATEXTENSIBLE)); + } - *ppDirectSound = pDirectSound; - return MA_SUCCESS; -} + ma_PropVariantClear(pContext, &prop); + } -ma_result ma_context_create_IDirectSoundCapture__dsound(ma_context* pContext, ma_share_mode shareMode, const ma_device_id* pDeviceID, ma_IDirectSoundCapture** ppDirectSoundCapture) -{ - ma_IDirectSoundCapture* pDirectSoundCapture; + ma_IPropertyStore_Release(pStore); + } + #else + /* + I do not know how to query the device's native format on UWP so for now I'm just disabling support for + exclusive mode. The alternative is to enumerate over different formats and check IsFormatSupported() + until you find one that works. + + TODO: Add support for exclusive mode to UWP. + */ + hr = S_FALSE; + #endif - ma_assert(pContext != NULL); - ma_assert(ppDirectSoundCapture != NULL); + if (hr == S_OK) { + shareMode = MA_AUDCLNT_SHAREMODE_EXCLUSIVE; + result = MA_SUCCESS; + } else { + result = MA_SHARE_MODE_NOT_SUPPORTED; + } + } else { + /* In shared mode we are always using the format reported by the operating system. */ + WAVEFORMATEXTENSIBLE* pNativeFormat = NULL; + hr = ma_IAudioClient_GetMixFormat((ma_IAudioClient*)pData->pAudioClient, (WAVEFORMATEX**)&pNativeFormat); + if (hr != S_OK) { + result = MA_FORMAT_NOT_SUPPORTED; + } else { + MA_COPY_MEMORY(&wf, pNativeFormat, sizeof(wf)); + result = MA_SUCCESS; + } - /* DirectSound does not support exclusive mode for capture. */ - if (shareMode == ma_share_mode_exclusive) { - return MA_SHARE_MODE_NOT_SUPPORTED; - } + ma_CoTaskMemFree(pContext, pNativeFormat); - *ppDirectSoundCapture = NULL; - pDirectSoundCapture = NULL; + shareMode = MA_AUDCLNT_SHAREMODE_SHARED; + } - if (FAILED(((ma_DirectSoundCaptureCreateProc)pContext->dsound.DirectSoundCaptureCreate)((pDeviceID == NULL) ? NULL : (const GUID*)pDeviceID->dsound, &pDirectSoundCapture, NULL))) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[DirectSound] DirectSoundCaptureCreate() failed for capture device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + /* Return an error if we still haven't found a format. */ + if (result != MA_SUCCESS) { + errorMsg = "[WASAPI] Failed to find best device mix format."; + goto done; } - *ppDirectSoundCapture = pDirectSoundCapture; - return MA_SUCCESS; -} + /* + Override the native sample rate with the one requested by the caller, but only if we're not using the default sample rate. We'll use + WASAPI to perform the sample rate conversion. + */ + nativeSampleRate = wf.Format.nSamplesPerSec; + if (streamFlags & MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM) { + wf.Format.nSamplesPerSec = pData->sampleRateIn; + wf.Format.nAvgBytesPerSec = wf.Format.nSamplesPerSec * wf.Format.nBlockAlign; + } -ma_result ma_context_get_format_info_for_IDirectSoundCapture__dsound(ma_context* pContext, ma_IDirectSoundCapture* pDirectSoundCapture, WORD* pChannels, WORD* pBitsPerSample, DWORD* pSampleRate) -{ - MA_DSCCAPS caps; - WORD bitsPerSample; - DWORD sampleRate; + pData->formatOut = ma_format_from_WAVEFORMATEX((WAVEFORMATEX*)&wf); + pData->channelsOut = wf.Format.nChannels; + pData->sampleRateOut = wf.Format.nSamplesPerSec; - ma_assert(pContext != NULL); - ma_assert(pDirectSoundCapture != NULL); + /* Get the internal channel map based on the channel mask. */ + ma_channel_mask_to_channel_map__win32(wf.dwChannelMask, pData->channelsOut, pData->channelMapOut); - if (pChannels) { - *pChannels = 0; - } - if (pBitsPerSample) { - *pBitsPerSample = 0; - } - if (pSampleRate) { - *pSampleRate = 0; + /* Period size. */ + pData->periodsOut = pData->periodsIn; + pData->periodSizeInFramesOut = pData->periodSizeInFramesIn; + if (pData->periodSizeInFramesOut == 0) { + pData->periodSizeInFramesOut = ma_calculate_buffer_size_in_frames_from_milliseconds(pData->periodSizeInMillisecondsIn, wf.Format.nSamplesPerSec); } - ma_zero_object(&caps); - caps.dwSize = sizeof(caps); - if (FAILED(ma_IDirectSoundCapture_GetCaps(pDirectSoundCapture, &caps))) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCapture_GetCaps() failed for capture device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } + periodDurationInMicroseconds = ((ma_uint64)pData->periodSizeInFramesOut * 1000 * 1000) / wf.Format.nSamplesPerSec; - if (pChannels) { - *pChannels = (WORD)caps.dwChannels; - } - /* The device can support multiple formats. We just go through the different formats in order of priority and pick the first one. This the same type of system as the WinMM backend. */ - bitsPerSample = 16; - sampleRate = 48000; + /* Slightly different initialization for shared and exclusive modes. We try exclusive mode first, and if it fails, fall back to shared mode. */ + if (shareMode == MA_AUDCLNT_SHAREMODE_EXCLUSIVE) { + MA_REFERENCE_TIME bufferDuration = periodDurationInMicroseconds * 10; - if (caps.dwChannels == 1) { - if ((caps.dwFormats & WAVE_FORMAT_48M16) != 0) { - sampleRate = 48000; - } else if ((caps.dwFormats & WAVE_FORMAT_44M16) != 0) { - sampleRate = 44100; - } else if ((caps.dwFormats & WAVE_FORMAT_2M16) != 0) { - sampleRate = 22050; - } else if ((caps.dwFormats & WAVE_FORMAT_1M16) != 0) { - sampleRate = 11025; - } else if ((caps.dwFormats & WAVE_FORMAT_96M16) != 0) { - sampleRate = 96000; - } else { - bitsPerSample = 8; - if ((caps.dwFormats & WAVE_FORMAT_48M08) != 0) { - sampleRate = 48000; - } else if ((caps.dwFormats & WAVE_FORMAT_44M08) != 0) { - sampleRate = 44100; - } else if ((caps.dwFormats & WAVE_FORMAT_2M08) != 0) { - sampleRate = 22050; - } else if ((caps.dwFormats & WAVE_FORMAT_1M08) != 0) { - sampleRate = 11025; - } else if ((caps.dwFormats & WAVE_FORMAT_96M08) != 0) { - sampleRate = 96000; + /* + If the periodicy is too small, Initialize() will fail with AUDCLNT_E_INVALID_DEVICE_PERIOD. In this case we should just keep increasing + it and trying it again. + */ + hr = E_FAIL; + for (;;) { + hr = ma_IAudioClient_Initialize((ma_IAudioClient*)pData->pAudioClient, shareMode, streamFlags, bufferDuration, bufferDuration, (WAVEFORMATEX*)&wf, NULL); + if (hr == MA_AUDCLNT_E_INVALID_DEVICE_PERIOD) { + if (bufferDuration > 500*10000) { + break; + } else { + if (bufferDuration == 0) { /* <-- Just a sanity check to prevent an infinit loop. Should never happen, but it makes me feel better. */ + break; + } + + bufferDuration = bufferDuration * 2; + continue; + } } else { - bitsPerSample = 16; /* Didn't find it. Just fall back to 16-bit. */ + break; } } - } else if (caps.dwChannels == 2) { - if ((caps.dwFormats & WAVE_FORMAT_48S16) != 0) { - sampleRate = 48000; - } else if ((caps.dwFormats & WAVE_FORMAT_44S16) != 0) { - sampleRate = 44100; - } else if ((caps.dwFormats & WAVE_FORMAT_2S16) != 0) { - sampleRate = 22050; - } else if ((caps.dwFormats & WAVE_FORMAT_1S16) != 0) { - sampleRate = 11025; - } else if ((caps.dwFormats & WAVE_FORMAT_96S16) != 0) { - sampleRate = 96000; - } else { - bitsPerSample = 8; - if ((caps.dwFormats & WAVE_FORMAT_48S08) != 0) { - sampleRate = 48000; - } else if ((caps.dwFormats & WAVE_FORMAT_44S08) != 0) { - sampleRate = 44100; - } else if ((caps.dwFormats & WAVE_FORMAT_2S08) != 0) { - sampleRate = 22050; - } else if ((caps.dwFormats & WAVE_FORMAT_1S08) != 0) { - sampleRate = 11025; - } else if ((caps.dwFormats & WAVE_FORMAT_96S08) != 0) { - sampleRate = 96000; + + if (hr == MA_AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED) { + ma_uint32 bufferSizeInFrames; + hr = ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pData->pAudioClient, &bufferSizeInFrames); + if (SUCCEEDED(hr)) { + bufferDuration = (MA_REFERENCE_TIME)((10000.0 * 1000 / wf.Format.nSamplesPerSec * bufferSizeInFrames) + 0.5); + + /* Unfortunately we need to release and re-acquire the audio client according to MSDN. Seems silly - why not just call IAudioClient_Initialize() again?! */ + ma_IAudioClient_Release((ma_IAudioClient*)pData->pAudioClient); + + #ifdef MA_WIN32_DESKTOP + hr = ma_IMMDevice_Activate(pDeviceInterface, &MA_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pData->pAudioClient); + #else + hr = ma_IUnknown_QueryInterface(pDeviceInterface, &MA_IID_IAudioClient, (void**)&pData->pAudioClient); + #endif + + if (SUCCEEDED(hr)) { + hr = ma_IAudioClient_Initialize((ma_IAudioClient*)pData->pAudioClient, shareMode, streamFlags, bufferDuration, bufferDuration, (WAVEFORMATEX*)&wf, NULL); + } + } + } + + if (FAILED(hr)) { + /* Failed to initialize in exclusive mode. Don't fall back to shared mode - instead tell the client about it. They can reinitialize in shared mode if they want. */ + if (hr == E_ACCESSDENIED) { + errorMsg = "[WASAPI] Failed to initialize device in exclusive mode. Access denied.", result = MA_ACCESS_DENIED; + } else if (hr == MA_AUDCLNT_E_DEVICE_IN_USE) { + errorMsg = "[WASAPI] Failed to initialize device in exclusive mode. Device in use.", result = MA_DEVICE_BUSY; } else { - bitsPerSample = 16; /* Didn't find it. Just fall back to 16-bit. */ + errorMsg = "[WASAPI] Failed to initialize device in exclusive mode."; result = MA_SHARE_MODE_NOT_SUPPORTED; } + goto done; } } - if (pBitsPerSample) { - *pBitsPerSample = bitsPerSample; - } - if (pSampleRate) { - *pSampleRate = sampleRate; - } + if (shareMode == MA_AUDCLNT_SHAREMODE_SHARED) { + /* + Low latency shared mode via IAudioClient3. - return MA_SUCCESS; -} + NOTE + ==== + Contrary to the documentation on MSDN (https://docs.microsoft.com/en-us/windows/win32/api/audioclient/nf-audioclient-iaudioclient3-initializesharedaudiostream), the + use of AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM and AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY with IAudioClient3_InitializeSharedAudioStream() absolutely does not work. Using + any of these flags will result in HRESULT code 0x88890021. The other problem is that calling IAudioClient3_GetSharedModeEnginePeriod() with a sample rate different to + that returned by IAudioClient_GetMixFormat() also results in an error. I'm therefore disabling low-latency shared mode with AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM. + */ +#ifndef MA_WASAPI_NO_LOW_LATENCY_SHARED_MODE + if ((streamFlags & MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM) == 0 || nativeSampleRate == wf.Format.nSamplesPerSec) { + ma_IAudioClient3* pAudioClient3 = NULL; + hr = ma_IAudioClient_QueryInterface(pData->pAudioClient, &MA_IID_IAudioClient3, (void**)&pAudioClient3); + if (SUCCEEDED(hr)) { + UINT32 defaultPeriodInFrames; + UINT32 fundamentalPeriodInFrames; + UINT32 minPeriodInFrames; + UINT32 maxPeriodInFrames; + hr = ma_IAudioClient3_GetSharedModeEnginePeriod(pAudioClient3, (WAVEFORMATEX*)&wf, &defaultPeriodInFrames, &fundamentalPeriodInFrames, &minPeriodInFrames, &maxPeriodInFrames); + if (SUCCEEDED(hr)) { + UINT32 desiredPeriodInFrames = pData->periodSizeInFramesOut; + UINT32 actualPeriodInFrames = desiredPeriodInFrames; -ma_bool32 ma_context_is_device_id_equal__dsound(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) -{ - ma_assert(pContext != NULL); - ma_assert(pID0 != NULL); - ma_assert(pID1 != NULL); - (void)pContext; + /* Make sure the period size is a multiple of fundamentalPeriodInFrames. */ + actualPeriodInFrames = actualPeriodInFrames / fundamentalPeriodInFrames; + actualPeriodInFrames = actualPeriodInFrames * fundamentalPeriodInFrames; - return memcmp(pID0->dsound, pID1->dsound, sizeof(pID0->dsound)) == 0; -} + /* The period needs to be clamped between minPeriodInFrames and maxPeriodInFrames. */ + actualPeriodInFrames = ma_clamp(actualPeriodInFrames, minPeriodInFrames, maxPeriodInFrames); + #if defined(MA_DEBUG_OUTPUT) + printf("[WASAPI] Trying IAudioClient3_InitializeSharedAudioStream(actualPeriodInFrames=%d)\n", actualPeriodInFrames); + printf(" defaultPeriodInFrames=%d\n", defaultPeriodInFrames); + printf(" fundamentalPeriodInFrames=%d\n", fundamentalPeriodInFrames); + printf(" minPeriodInFrames=%d\n", minPeriodInFrames); + printf(" maxPeriodInFrames=%d\n", maxPeriodInFrames); + #endif -typedef struct -{ - ma_context* pContext; - ma_device_type deviceType; - ma_enum_devices_callback_proc callback; - void* pUserData; - ma_bool32 terminated; -} ma_context_enumerate_devices_callback_data__dsound; + /* If the client requested a largish buffer than we don't actually want to use low latency shared mode because it forces small buffers. */ + if (actualPeriodInFrames >= desiredPeriodInFrames) { + /* + MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM | MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY must not be in the stream flags. If either of these are specified, + IAudioClient3_InitializeSharedAudioStream() will fail. + */ + hr = ma_IAudioClient3_InitializeSharedAudioStream(pAudioClient3, streamFlags & ~(MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM | MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY), actualPeriodInFrames, (WAVEFORMATEX*)&wf, NULL); + if (SUCCEEDED(hr)) { + wasInitializedUsingIAudioClient3 = MA_TRUE; + pData->periodSizeInFramesOut = actualPeriodInFrames; + #if defined(MA_DEBUG_OUTPUT) + printf("[WASAPI] Using IAudioClient3\n"); + printf(" periodSizeInFramesOut=%d\n", pData->periodSizeInFramesOut); + #endif + } else { + #if defined(MA_DEBUG_OUTPUT) + printf("[WASAPI] IAudioClient3_InitializeSharedAudioStream failed. Falling back to IAudioClient.\n"); + #endif + } + } else { + #if defined(MA_DEBUG_OUTPUT) + printf("[WASAPI] Not using IAudioClient3 because the desired period size is larger than the maximum supported by IAudioClient3.\n"); + #endif + } + } else { + #if defined(MA_DEBUG_OUTPUT) + printf("[WASAPI] IAudioClient3_GetSharedModeEnginePeriod failed. Falling back to IAudioClient.\n"); + #endif + } -BOOL CALLBACK ma_context_enumerate_devices_callback__dsound(LPGUID lpGuid, LPCSTR lpcstrDescription, LPCSTR lpcstrModule, LPVOID lpContext) -{ - ma_context_enumerate_devices_callback_data__dsound* pData = (ma_context_enumerate_devices_callback_data__dsound*)lpContext; - ma_device_info deviceInfo; + ma_IAudioClient3_Release(pAudioClient3); + pAudioClient3 = NULL; + } + } +#else + #if defined(MA_DEBUG_OUTPUT) + printf("[WASAPI] Not using IAudioClient3 because MA_WASAPI_NO_LOW_LATENCY_SHARED_MODE is enabled.\n"); + #endif +#endif - ma_zero_object(&deviceInfo); + /* If we don't have an IAudioClient3 then we need to use the normal initialization routine. */ + if (!wasInitializedUsingIAudioClient3) { + MA_REFERENCE_TIME bufferDuration = periodDurationInMicroseconds * pData->periodsOut * 10; /* <-- Multiply by 10 for microseconds to 100-nanoseconds. */ + hr = ma_IAudioClient_Initialize((ma_IAudioClient*)pData->pAudioClient, shareMode, streamFlags, bufferDuration, 0, (WAVEFORMATEX*)&wf, NULL); + if (FAILED(hr)) { + if (hr == E_ACCESSDENIED) { + errorMsg = "[WASAPI] Failed to initialize device. Access denied.", result = MA_ACCESS_DENIED; + } else if (hr == MA_AUDCLNT_E_DEVICE_IN_USE) { + errorMsg = "[WASAPI] Failed to initialize device. Device in use.", result = MA_DEVICE_BUSY; + } else { + errorMsg = "[WASAPI] Failed to initialize device.", result = MA_FAILED_TO_OPEN_BACKEND_DEVICE; + } - /* ID. */ - if (lpGuid != NULL) { - ma_copy_memory(deviceInfo.id.dsound, lpGuid, 16); - } else { - ma_zero_memory(deviceInfo.id.dsound, 16); + goto done; + } + } } - /* Name / Description */ - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), lpcstrDescription, (size_t)-1); - + if (!wasInitializedUsingIAudioClient3) { + ma_uint32 bufferSizeInFrames; + hr = ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pData->pAudioClient, &bufferSizeInFrames); + if (FAILED(hr)) { + errorMsg = "[WASAPI] Failed to get audio client's actual buffer size.", result = MA_FAILED_TO_OPEN_BACKEND_DEVICE; + goto done; + } - /* Call the callback function, but make sure we stop enumerating if the callee requested so. */ - ma_assert(pData != NULL); - pData->terminated = !pData->callback(pData->pContext, pData->deviceType, &deviceInfo, pData->pUserData); - if (pData->terminated) { - return FALSE; /* Stop enumeration. */ - } else { - return TRUE; /* Continue enumeration. */ + pData->periodSizeInFramesOut = bufferSizeInFrames / pData->periodsOut; } - (void)lpcstrModule; -} - -ma_result ma_context_enumerate_devices__dsound(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) -{ - ma_context_enumerate_devices_callback_data__dsound data; - - ma_assert(pContext != NULL); - ma_assert(callback != NULL); - - data.pContext = pContext; - data.callback = callback; - data.pUserData = pUserData; - data.terminated = MA_FALSE; + pData->usingAudioClient3 = wasInitializedUsingIAudioClient3; - /* Playback. */ - if (!data.terminated) { - data.deviceType = ma_device_type_playback; - ((ma_DirectSoundEnumerateAProc)pContext->dsound.DirectSoundEnumerateA)(ma_context_enumerate_devices_callback__dsound, &data); + if (deviceType == ma_device_type_playback) { + hr = ma_IAudioClient_GetService((ma_IAudioClient*)pData->pAudioClient, &MA_IID_IAudioRenderClient, (void**)&pData->pRenderClient); + } else { + hr = ma_IAudioClient_GetService((ma_IAudioClient*)pData->pAudioClient, &MA_IID_IAudioCaptureClient, (void**)&pData->pCaptureClient); } - /* Capture. */ - if (!data.terminated) { - data.deviceType = ma_device_type_capture; - ((ma_DirectSoundCaptureEnumerateAProc)pContext->dsound.DirectSoundCaptureEnumerateA)(ma_context_enumerate_devices_callback__dsound, &data); + if (FAILED(hr)) { + errorMsg = "[WASAPI] Failed to get audio client service.", result = MA_API_NOT_FOUND; + goto done; } - return MA_SUCCESS; -} - - -typedef struct -{ - const ma_device_id* pDeviceID; - ma_device_info* pDeviceInfo; - ma_bool32 found; -} ma_context_get_device_info_callback_data__dsound; - -BOOL CALLBACK ma_context_get_device_info_callback__dsound(LPGUID lpGuid, LPCSTR lpcstrDescription, LPCSTR lpcstrModule, LPVOID lpContext) -{ - ma_context_get_device_info_callback_data__dsound* pData = (ma_context_get_device_info_callback_data__dsound*)lpContext; - ma_assert(pData != NULL); - if ((pData->pDeviceID == NULL || ma_is_guid_equal(pData->pDeviceID->dsound, &MA_GUID_NULL)) && (lpGuid == NULL || ma_is_guid_equal(lpGuid, &MA_GUID_NULL))) { - /* Default device. */ - ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), lpcstrDescription, (size_t)-1); - pData->found = MA_TRUE; - return FALSE; /* Stop enumeration. */ - } else { - /* Not the default device. */ - if (lpGuid != NULL) { - if (memcmp(pData->pDeviceID->dsound, lpGuid, sizeof(pData->pDeviceID->dsound)) == 0) { - ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), lpcstrDescription, (size_t)-1); - pData->found = MA_TRUE; - return FALSE; /* Stop enumeration. */ + /* Grab the name of the device. */ +#ifdef MA_WIN32_DESKTOP + { + ma_IPropertyStore *pProperties; + hr = ma_IMMDevice_OpenPropertyStore(pDeviceInterface, STGM_READ, &pProperties); + if (SUCCEEDED(hr)) { + PROPVARIANT varName; + ma_PropVariantInit(&varName); + hr = ma_IPropertyStore_GetValue(pProperties, &MA_PKEY_Device_FriendlyName, &varName); + if (SUCCEEDED(hr)) { + WideCharToMultiByte(CP_UTF8, 0, varName.pwszVal, -1, pData->deviceName, sizeof(pData->deviceName), 0, FALSE); + ma_PropVariantClear(pContext, &varName); } + + ma_IPropertyStore_Release(pProperties); } } +#endif - (void)lpcstrModule; - return TRUE; -} - -ma_result ma_context_get_device_info__dsound(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) -{ - /* Exclusive mode and capture not supported with DirectSound. */ - if (deviceType == ma_device_type_capture && shareMode == ma_share_mode_exclusive) { - return MA_SHARE_MODE_NOT_SUPPORTED; +done: + /* Clean up. */ +#ifdef MA_WIN32_DESKTOP + if (pDeviceInterface != NULL) { + ma_IMMDevice_Release(pDeviceInterface); } +#else + if (pDeviceInterface != NULL) { + ma_IUnknown_Release(pDeviceInterface); + } +#endif - if (pDeviceID != NULL) { - ma_context_get_device_info_callback_data__dsound data; - - /* ID. */ - ma_copy_memory(pDeviceInfo->id.dsound, pDeviceID->dsound, 16); - - /* Name / Description. This is retrieved by enumerating over each device until we find that one that matches the input ID. */ - data.pDeviceID = pDeviceID; - data.pDeviceInfo = pDeviceInfo; - data.found = MA_FALSE; - if (deviceType == ma_device_type_playback) { - ((ma_DirectSoundEnumerateAProc)pContext->dsound.DirectSoundEnumerateA)(ma_context_get_device_info_callback__dsound, &data); - } else { - ((ma_DirectSoundCaptureEnumerateAProc)pContext->dsound.DirectSoundCaptureEnumerateA)(ma_context_get_device_info_callback__dsound, &data); + if (result != MA_SUCCESS) { + if (pData->pRenderClient) { + ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pData->pRenderClient); + pData->pRenderClient = NULL; + } + if (pData->pCaptureClient) { + ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pData->pCaptureClient); + pData->pCaptureClient = NULL; + } + if (pData->pAudioClient) { + ma_IAudioClient_Release((ma_IAudioClient*)pData->pAudioClient); + pData->pAudioClient = NULL; } - if (!data.found) { - return MA_NO_DEVICE; - } + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, errorMsg, result); } else { - /* I don't think there's a way to get the name of the default device with DirectSound. In this case we just need to use defaults. */ + return MA_SUCCESS; + } +} - /* ID */ - ma_zero_memory(pDeviceInfo->id.dsound, 16); +static ma_result ma_device_reinit__wasapi(ma_device* pDevice, ma_device_type deviceType) +{ + ma_device_init_internal_data__wasapi data; + ma_result result; - /* Name / Description */ - if (deviceType == ma_device_type_playback) { - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); - } else { - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); - } + MA_ASSERT(pDevice != NULL); + + /* We only re-initialize the playback or capture device. Never a full-duplex device. */ + if (deviceType == ma_device_type_duplex) { + return MA_INVALID_ARGS; } - /* Retrieving detailed information is slightly different depending on the device type. */ if (deviceType == ma_device_type_playback) { - /* Playback. */ - ma_IDirectSound* pDirectSound; - ma_result result; - MA_DSCAPS caps; - ma_uint32 iFormat; + data.formatIn = pDevice->playback.format; + data.channelsIn = pDevice->playback.channels; + MA_COPY_MEMORY(data.channelMapIn, pDevice->playback.channelMap, sizeof(pDevice->playback.channelMap)); + data.shareMode = pDevice->playback.shareMode; + data.usingDefaultFormat = pDevice->playback.usingDefaultFormat; + data.usingDefaultChannels = pDevice->playback.usingDefaultChannels; + data.usingDefaultChannelMap = pDevice->playback.usingDefaultChannelMap; + } else { + data.formatIn = pDevice->capture.format; + data.channelsIn = pDevice->capture.channels; + MA_COPY_MEMORY(data.channelMapIn, pDevice->capture.channelMap, sizeof(pDevice->capture.channelMap)); + data.shareMode = pDevice->capture.shareMode; + data.usingDefaultFormat = pDevice->capture.usingDefaultFormat; + data.usingDefaultChannels = pDevice->capture.usingDefaultChannels; + data.usingDefaultChannelMap = pDevice->capture.usingDefaultChannelMap; + } + + data.sampleRateIn = pDevice->sampleRate; + data.usingDefaultSampleRate = pDevice->usingDefaultSampleRate; + data.periodSizeInFramesIn = pDevice->wasapi.originalPeriodSizeInFrames; + data.periodSizeInMillisecondsIn = pDevice->wasapi.originalPeriodSizeInMilliseconds; + data.periodsIn = pDevice->wasapi.originalPeriods; + data.noAutoConvertSRC = pDevice->wasapi.noAutoConvertSRC; + data.noDefaultQualitySRC = pDevice->wasapi.noDefaultQualitySRC; + data.noHardwareOffloading = pDevice->wasapi.noHardwareOffloading; + result = ma_device_init_internal__wasapi(pDevice->pContext, deviceType, NULL, &data); + if (result != MA_SUCCESS) { + return result; + } - result = ma_context_create_IDirectSound__dsound(pContext, shareMode, pDeviceID, &pDirectSound); - if (result != MA_SUCCESS) { - return result; + /* At this point we have some new objects ready to go. We need to uninitialize the previous ones and then set the new ones. */ + if (deviceType == ma_device_type_capture || deviceType == ma_device_type_loopback) { + if (pDevice->wasapi.pCaptureClient) { + ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient); + pDevice->wasapi.pCaptureClient = NULL; } - ma_zero_object(&caps); - caps.dwSize = sizeof(caps); - if (FAILED(ma_IDirectSound_GetCaps(pDirectSound, &caps))) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_GetCaps() failed for playback device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + if (pDevice->wasapi.pAudioClientCapture) { + ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); + pDevice->wasapi.pAudioClientCapture = NULL; } - if ((caps.dwFlags & MA_DSCAPS_PRIMARYSTEREO) != 0) { - /* It supports at least stereo, but could support more. */ - WORD channels = 2; + pDevice->wasapi.pAudioClientCapture = data.pAudioClient; + pDevice->wasapi.pCaptureClient = data.pCaptureClient; - /* Look at the speaker configuration to get a better idea on the channel count. */ - DWORD speakerConfig; - if (SUCCEEDED(ma_IDirectSound_GetSpeakerConfig(pDirectSound, &speakerConfig))) { - ma_get_channels_from_speaker_config__dsound(speakerConfig, &channels, NULL); - } + pDevice->capture.internalFormat = data.formatOut; + pDevice->capture.internalChannels = data.channelsOut; + pDevice->capture.internalSampleRate = data.sampleRateOut; + MA_COPY_MEMORY(pDevice->capture.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); + pDevice->capture.internalPeriodSizeInFrames = data.periodSizeInFramesOut; + pDevice->capture.internalPeriods = data.periodsOut; + ma_strcpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), data.deviceName); - pDeviceInfo->minChannels = channels; - pDeviceInfo->maxChannels = channels; - } else { - /* It does not support stereo, which means we are stuck with mono. */ - pDeviceInfo->minChannels = 1; - pDeviceInfo->maxChannels = 1; - } + ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, pDevice->wasapi.hEventCapture); - /* Sample rate. */ - if ((caps.dwFlags & MA_DSCAPS_CONTINUOUSRATE) != 0) { - pDeviceInfo->minSampleRate = caps.dwMinSecondarySampleRate; - pDeviceInfo->maxSampleRate = caps.dwMaxSecondarySampleRate; + pDevice->wasapi.periodSizeInFramesCapture = data.periodSizeInFramesOut; + ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, &pDevice->wasapi.actualPeriodSizeInFramesCapture); - /* - On my machine the min and max sample rates can return 100 and 200000 respectively. I'd rather these be within - the range of our standard sample rates so I'm clamping. - */ - if (caps.dwMinSecondarySampleRate < MA_MIN_SAMPLE_RATE && caps.dwMaxSecondarySampleRate >= MA_MIN_SAMPLE_RATE) { - pDeviceInfo->minSampleRate = MA_MIN_SAMPLE_RATE; - } - if (caps.dwMaxSecondarySampleRate > MA_MAX_SAMPLE_RATE && caps.dwMinSecondarySampleRate <= MA_MAX_SAMPLE_RATE) { - pDeviceInfo->maxSampleRate = MA_MAX_SAMPLE_RATE; + /* The device may be in a started state. If so we need to immediately restart it. */ + if (pDevice->wasapi.isStartedCapture) { + HRESULT hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); + if (FAILED(hr)) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal capture device after reinitialization.", MA_FAILED_TO_START_BACKEND_DEVICE); } - } else { - /* Only supports a single sample rate. Set both min an max to the same thing. Do not clamp within the standard rates. */ - pDeviceInfo->minSampleRate = caps.dwMaxSecondarySampleRate; - pDeviceInfo->maxSampleRate = caps.dwMaxSecondarySampleRate; - } - - /* DirectSound can support all formats. */ - pDeviceInfo->formatCount = ma_format_count - 1; /* Minus one because we don't want to include ma_format_unknown. */ - for (iFormat = 0; iFormat < pDeviceInfo->formatCount; ++iFormat) { - pDeviceInfo->formats[iFormat] = (ma_format)(iFormat + 1); /* +1 to skip over ma_format_unknown. */ - } - - ma_IDirectSound_Release(pDirectSound); - } else { - /* - Capture. This is a little different to playback due to the say the supported formats are reported. Technically capture - devices can support a number of different formats, but for simplicity and consistency with ma_device_init() I'm just - reporting the best format. - */ - ma_IDirectSoundCapture* pDirectSoundCapture; - ma_result result; - WORD channels; - WORD bitsPerSample; - DWORD sampleRate; - - result = ma_context_create_IDirectSoundCapture__dsound(pContext, shareMode, pDeviceID, &pDirectSoundCapture); - if (result != MA_SUCCESS) { - return result; } + } - result = ma_context_get_format_info_for_IDirectSoundCapture__dsound(pContext, pDirectSoundCapture, &channels, &bitsPerSample, &sampleRate); - if (result != MA_SUCCESS) { - ma_IDirectSoundCapture_Release(pDirectSoundCapture); - return result; + if (deviceType == ma_device_type_playback) { + if (pDevice->wasapi.pRenderClient) { + ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient); + pDevice->wasapi.pRenderClient = NULL; } - pDeviceInfo->minChannels = channels; - pDeviceInfo->maxChannels = channels; - pDeviceInfo->minSampleRate = sampleRate; - pDeviceInfo->maxSampleRate = sampleRate; - pDeviceInfo->formatCount = 1; - if (bitsPerSample == 8) { - pDeviceInfo->formats[0] = ma_format_u8; - } else if (bitsPerSample == 16) { - pDeviceInfo->formats[0] = ma_format_s16; - } else if (bitsPerSample == 24) { - pDeviceInfo->formats[0] = ma_format_s24; - } else if (bitsPerSample == 32) { - pDeviceInfo->formats[0] = ma_format_s32; - } else { - ma_IDirectSoundCapture_Release(pDirectSoundCapture); - return MA_FORMAT_NOT_SUPPORTED; + if (pDevice->wasapi.pAudioClientPlayback) { + ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); + pDevice->wasapi.pAudioClientPlayback = NULL; } - ma_IDirectSoundCapture_Release(pDirectSoundCapture); - } - - return MA_SUCCESS; -} - + pDevice->wasapi.pAudioClientPlayback = data.pAudioClient; + pDevice->wasapi.pRenderClient = data.pRenderClient; -typedef struct -{ - ma_uint32 deviceCount; - ma_uint32 infoCount; - ma_device_info* pInfo; -} ma_device_enum_data__dsound; + pDevice->playback.internalFormat = data.formatOut; + pDevice->playback.internalChannels = data.channelsOut; + pDevice->playback.internalSampleRate = data.sampleRateOut; + MA_COPY_MEMORY(pDevice->playback.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); + pDevice->playback.internalPeriodSizeInFrames = data.periodSizeInFramesOut; + pDevice->playback.internalPeriods = data.periodsOut; + ma_strcpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), data.deviceName); -BOOL CALLBACK ma_enum_devices_callback__dsound(LPGUID lpGuid, LPCSTR lpcstrDescription, LPCSTR lpcstrModule, LPVOID lpContext) -{ - ma_device_enum_data__dsound* pData = (ma_device_enum_data__dsound*)lpContext; - ma_assert(pData != NULL); + ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, pDevice->wasapi.hEventPlayback); - if (pData->pInfo != NULL) { - if (pData->infoCount > 0) { - ma_zero_object(pData->pInfo); - ma_strncpy_s(pData->pInfo->name, sizeof(pData->pInfo->name), lpcstrDescription, (size_t)-1); + pDevice->wasapi.periodSizeInFramesPlayback = data.periodSizeInFramesOut; + ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &pDevice->wasapi.actualPeriodSizeInFramesPlayback); - if (lpGuid != NULL) { - ma_copy_memory(pData->pInfo->id.dsound, lpGuid, 16); - } else { - ma_zero_memory(pData->pInfo->id.dsound, 16); + /* The device may be in a started state. If so we need to immediately restart it. */ + if (pDevice->wasapi.isStartedPlayback) { + HRESULT hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); + if (FAILED(hr)) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal playback device after reinitialization.", MA_FAILED_TO_START_BACKEND_DEVICE); } - - pData->pInfo += 1; - pData->infoCount -= 1; - pData->deviceCount += 1; } - } else { - pData->deviceCount += 1; } - (void)lpcstrModule; - return TRUE; + return MA_SUCCESS; } -void ma_device_uninit__dsound(ma_device* pDevice) +static ma_result ma_device_init__wasapi(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) { - ma_assert(pDevice != NULL); - - if (pDevice->dsound.pCaptureBuffer != NULL) { - ma_IDirectSoundCaptureBuffer_Release((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer); - } - if (pDevice->dsound.pCapture != NULL) { - ma_IDirectSoundCapture_Release((ma_IDirectSoundCapture*)pDevice->dsound.pCapture); - } - - if (pDevice->dsound.pPlaybackBuffer != NULL) { - ma_IDirectSoundBuffer_Release((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer); - } - if (pDevice->dsound.pPlaybackPrimaryBuffer != NULL) { - ma_IDirectSoundBuffer_Release((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer); - } - if (pDevice->dsound.pPlayback != NULL) { - ma_IDirectSound_Release((ma_IDirectSound*)pDevice->dsound.pPlayback); - } -} + ma_result result = MA_SUCCESS; -ma_result ma_config_to_WAVEFORMATEXTENSIBLE(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, const ma_channel* pChannelMap, WAVEFORMATEXTENSIBLE* pWF) -{ - GUID subformat; + (void)pContext; - switch (format) - { - case ma_format_u8: - case ma_format_s16: - case ma_format_s24: - /*case ma_format_s24_32:*/ - case ma_format_s32: - { - subformat = MA_GUID_KSDATAFORMAT_SUBTYPE_PCM; - } break; + MA_ASSERT(pContext != NULL); + MA_ASSERT(pDevice != NULL); - case ma_format_f32: - { - subformat = MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT; - } break; + MA_ZERO_OBJECT(&pDevice->wasapi); + pDevice->wasapi.originalPeriodSizeInFrames = pConfig->periodSizeInFrames; + pDevice->wasapi.originalPeriodSizeInMilliseconds = pConfig->periodSizeInMilliseconds; + pDevice->wasapi.originalPeriods = pConfig->periods; + pDevice->wasapi.noAutoConvertSRC = pConfig->wasapi.noAutoConvertSRC; + pDevice->wasapi.noDefaultQualitySRC = pConfig->wasapi.noDefaultQualitySRC; + pDevice->wasapi.noHardwareOffloading = pConfig->wasapi.noHardwareOffloading; - default: - return MA_FORMAT_NOT_SUPPORTED; + /* Exclusive mode is not allowed with loopback. */ + if (pConfig->deviceType == ma_device_type_loopback && pConfig->playback.shareMode == ma_share_mode_exclusive) { + return MA_INVALID_DEVICE_CONFIG; } - ma_zero_object(pWF); - pWF->Format.cbSize = sizeof(*pWF); - pWF->Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE; - pWF->Format.nChannels = (WORD)channels; - pWF->Format.nSamplesPerSec = (DWORD)sampleRate; - pWF->Format.wBitsPerSample = (WORD)ma_get_bytes_per_sample(format)*8; - pWF->Format.nBlockAlign = (pWF->Format.nChannels * pWF->Format.wBitsPerSample) / 8; - pWF->Format.nAvgBytesPerSec = pWF->Format.nBlockAlign * pWF->Format.nSamplesPerSec; - pWF->Samples.wValidBitsPerSample = pWF->Format.wBitsPerSample; - pWF->dwChannelMask = ma_channel_map_to_channel_mask__win32(pChannelMap, channels); - pWF->SubFormat = subformat; + if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex || pConfig->deviceType == ma_device_type_loopback) { + ma_device_init_internal_data__wasapi data; + data.formatIn = pConfig->capture.format; + data.channelsIn = pConfig->capture.channels; + data.sampleRateIn = pConfig->sampleRate; + MA_COPY_MEMORY(data.channelMapIn, pConfig->capture.channelMap, sizeof(pConfig->capture.channelMap)); + data.usingDefaultFormat = pDevice->capture.usingDefaultFormat; + data.usingDefaultChannels = pDevice->capture.usingDefaultChannels; + data.usingDefaultSampleRate = pDevice->usingDefaultSampleRate; + data.usingDefaultChannelMap = pDevice->capture.usingDefaultChannelMap; + data.shareMode = pConfig->capture.shareMode; + data.periodSizeInFramesIn = pConfig->periodSizeInFrames; + data.periodSizeInMillisecondsIn = pConfig->periodSizeInMilliseconds; + data.periodsIn = pConfig->periods; + data.noAutoConvertSRC = pConfig->wasapi.noAutoConvertSRC; + data.noDefaultQualitySRC = pConfig->wasapi.noDefaultQualitySRC; + data.noHardwareOffloading = pConfig->wasapi.noHardwareOffloading; - return MA_SUCCESS; -} + result = ma_device_init_internal__wasapi(pDevice->pContext, (pConfig->deviceType == ma_device_type_loopback) ? ma_device_type_loopback : ma_device_type_capture, pConfig->capture.pDeviceID, &data); + if (result != MA_SUCCESS) { + return result; + } -ma_result ma_device_init__dsound(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) -{ - ma_result result; - ma_uint32 bufferSizeInMilliseconds; + pDevice->wasapi.pAudioClientCapture = data.pAudioClient; + pDevice->wasapi.pCaptureClient = data.pCaptureClient; - ma_assert(pDevice != NULL); - ma_zero_object(&pDevice->dsound); + pDevice->capture.internalFormat = data.formatOut; + pDevice->capture.internalChannels = data.channelsOut; + pDevice->capture.internalSampleRate = data.sampleRateOut; + MA_COPY_MEMORY(pDevice->capture.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); + pDevice->capture.internalPeriodSizeInFrames = data.periodSizeInFramesOut; + pDevice->capture.internalPeriods = data.periodsOut; + ma_strcpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), data.deviceName); - if (pConfig->deviceType == ma_device_type_loopback) { - return MA_DEVICE_TYPE_NOT_SUPPORTED; - } + /* + The event for capture needs to be manual reset for the same reason as playback. We keep the initial state set to unsignaled, + however, because we want to block until we actually have something for the first call to ma_device_read(). + */ + pDevice->wasapi.hEventCapture = CreateEventA(NULL, FALSE, FALSE, NULL); /* Auto reset, unsignaled by default. */ + if (pDevice->wasapi.hEventCapture == NULL) { + if (pDevice->wasapi.pCaptureClient != NULL) { + ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient); + pDevice->wasapi.pCaptureClient = NULL; + } + if (pDevice->wasapi.pAudioClientCapture != NULL) { + ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); + pDevice->wasapi.pAudioClientCapture = NULL; + } - bufferSizeInMilliseconds = pConfig->bufferSizeInMilliseconds; - if (bufferSizeInMilliseconds == 0) { - bufferSizeInMilliseconds = ma_calculate_buffer_size_in_milliseconds_from_frames(pConfig->bufferSizeInFrames, pConfig->sampleRate); - } - - /* DirectSound should use a latency of about 20ms per period for low latency mode. */ - if (pDevice->usingDefaultBufferSize) { - if (pConfig->performanceProfile == ma_performance_profile_low_latency) { - bufferSizeInMilliseconds = 20 * pConfig->periods; - } else { - bufferSizeInMilliseconds = 200 * pConfig->periods; + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create event for capture.", MA_FAILED_TO_CREATE_EVENT); } - } + ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, pDevice->wasapi.hEventCapture); - /* DirectSound breaks down with tiny buffer sizes (bad glitching and silent output). I am therefore restricting the size of the buffer to a minimum of 20 milliseconds. */ - if ((bufferSizeInMilliseconds/pConfig->periods) < 20) { - bufferSizeInMilliseconds = pConfig->periods * 20; + pDevice->wasapi.periodSizeInFramesCapture = data.periodSizeInFramesOut; + ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, &pDevice->wasapi.actualPeriodSizeInFramesCapture); } - /* - Unfortunately DirectSound uses different APIs and data structures for playback and catpure devices. We need to initialize - the capture device first because we'll want to match it's buffer size and period count on the playback side if we're using - full-duplex mode. - */ - if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { - WAVEFORMATEXTENSIBLE wf; - MA_DSCBUFFERDESC descDS; - ma_uint32 bufferSizeInFrames; - char rawdata[1024]; /* <-- Ugly hack to avoid a malloc() due to a crappy DirectSound API. */ - WAVEFORMATEXTENSIBLE* pActualFormat; + if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { + ma_device_init_internal_data__wasapi data; + data.formatIn = pConfig->playback.format; + data.channelsIn = pConfig->playback.channels; + data.sampleRateIn = pConfig->sampleRate; + MA_COPY_MEMORY(data.channelMapIn, pConfig->playback.channelMap, sizeof(pConfig->playback.channelMap)); + data.usingDefaultFormat = pDevice->playback.usingDefaultFormat; + data.usingDefaultChannels = pDevice->playback.usingDefaultChannels; + data.usingDefaultSampleRate = pDevice->usingDefaultSampleRate; + data.usingDefaultChannelMap = pDevice->playback.usingDefaultChannelMap; + data.shareMode = pConfig->playback.shareMode; + data.periodSizeInFramesIn = pConfig->periodSizeInFrames; + data.periodSizeInMillisecondsIn = pConfig->periodSizeInMilliseconds; + data.periodsIn = pConfig->periods; + data.noAutoConvertSRC = pConfig->wasapi.noAutoConvertSRC; + data.noDefaultQualitySRC = pConfig->wasapi.noDefaultQualitySRC; + data.noHardwareOffloading = pConfig->wasapi.noHardwareOffloading; - result = ma_config_to_WAVEFORMATEXTENSIBLE(pConfig->capture.format, pConfig->capture.channels, pConfig->sampleRate, pConfig->capture.channelMap, &wf); + result = ma_device_init_internal__wasapi(pDevice->pContext, ma_device_type_playback, pConfig->playback.pDeviceID, &data); if (result != MA_SUCCESS) { - return result; - } + if (pConfig->deviceType == ma_device_type_duplex) { + if (pDevice->wasapi.pCaptureClient != NULL) { + ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient); + pDevice->wasapi.pCaptureClient = NULL; + } + if (pDevice->wasapi.pAudioClientCapture != NULL) { + ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); + pDevice->wasapi.pAudioClientCapture = NULL; + } - result = ma_context_create_IDirectSoundCapture__dsound(pContext, pConfig->capture.shareMode, pConfig->capture.pDeviceID, (ma_IDirectSoundCapture**)&pDevice->dsound.pCapture); - if (result != MA_SUCCESS) { - ma_device_uninit__dsound(pDevice); + CloseHandle(pDevice->wasapi.hEventCapture); + pDevice->wasapi.hEventCapture = NULL; + } return result; } - result = ma_context_get_format_info_for_IDirectSoundCapture__dsound(pContext, (ma_IDirectSoundCapture*)pDevice->dsound.pCapture, &wf.Format.nChannels, &wf.Format.wBitsPerSample, &wf.Format.nSamplesPerSec); - if (result != MA_SUCCESS) { - ma_device_uninit__dsound(pDevice); - return result; - } + pDevice->wasapi.pAudioClientPlayback = data.pAudioClient; + pDevice->wasapi.pRenderClient = data.pRenderClient; - wf.Format.nBlockAlign = (wf.Format.nChannels * wf.Format.wBitsPerSample) / 8; - wf.Format.nAvgBytesPerSec = wf.Format.nBlockAlign * wf.Format.nSamplesPerSec; - wf.Samples.wValidBitsPerSample = wf.Format.wBitsPerSample; - wf.SubFormat = MA_GUID_KSDATAFORMAT_SUBTYPE_PCM; + pDevice->playback.internalFormat = data.formatOut; + pDevice->playback.internalChannels = data.channelsOut; + pDevice->playback.internalSampleRate = data.sampleRateOut; + MA_COPY_MEMORY(pDevice->playback.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); + pDevice->playback.internalPeriodSizeInFrames = data.periodSizeInFramesOut; + pDevice->playback.internalPeriods = data.periodsOut; + ma_strcpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), data.deviceName); - /* The size of the buffer must be a clean multiple of the period count. */ - bufferSizeInFrames = (ma_calculate_buffer_size_in_frames_from_milliseconds(bufferSizeInMilliseconds, wf.Format.nSamplesPerSec) / pConfig->periods) * pConfig->periods; + /* + The event for playback is needs to be manual reset because we want to explicitly control the fact that it becomes signalled + only after the whole available space has been filled, never before. - ma_zero_object(&descDS); - descDS.dwSize = sizeof(descDS); - descDS.dwFlags = 0; - descDS.dwBufferBytes = bufferSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.internalFormat, wf.Format.nChannels); - descDS.lpwfxFormat = (WAVEFORMATEX*)&wf; - if (FAILED(ma_IDirectSoundCapture_CreateCaptureBuffer((ma_IDirectSoundCapture*)pDevice->dsound.pCapture, &descDS, (ma_IDirectSoundCaptureBuffer**)&pDevice->dsound.pCaptureBuffer, NULL))) { - ma_device_uninit__dsound(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCapture_CreateCaptureBuffer() failed for capture device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } + The playback event also needs to be initially set to a signaled state so that the first call to ma_device_write() is able + to get passed WaitForMultipleObjects(). + */ + pDevice->wasapi.hEventPlayback = CreateEventA(NULL, FALSE, TRUE, NULL); /* Auto reset, signaled by default. */ + if (pDevice->wasapi.hEventPlayback == NULL) { + if (pConfig->deviceType == ma_device_type_duplex) { + if (pDevice->wasapi.pCaptureClient != NULL) { + ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient); + pDevice->wasapi.pCaptureClient = NULL; + } + if (pDevice->wasapi.pAudioClientCapture != NULL) { + ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); + pDevice->wasapi.pAudioClientCapture = NULL; + } - /* Get the _actual_ properties of the buffer. */ - pActualFormat = (WAVEFORMATEXTENSIBLE*)rawdata; - if (FAILED(ma_IDirectSoundCaptureBuffer_GetFormat((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, (WAVEFORMATEX*)pActualFormat, sizeof(rawdata), NULL))) { - ma_device_uninit__dsound(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to retrieve the actual format of the capture device's buffer.", MA_FORMAT_NOT_SUPPORTED); + CloseHandle(pDevice->wasapi.hEventCapture); + pDevice->wasapi.hEventCapture = NULL; + } + + if (pDevice->wasapi.pRenderClient != NULL) { + ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient); + pDevice->wasapi.pRenderClient = NULL; + } + if (pDevice->wasapi.pAudioClientPlayback != NULL) { + ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); + pDevice->wasapi.pAudioClientPlayback = NULL; + } + + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create event for playback.", MA_FAILED_TO_CREATE_EVENT); } + ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, pDevice->wasapi.hEventPlayback); - pDevice->capture.internalFormat = ma_format_from_WAVEFORMATEX((WAVEFORMATEX*)pActualFormat); - pDevice->capture.internalChannels = pActualFormat->Format.nChannels; - pDevice->capture.internalSampleRate = pActualFormat->Format.nSamplesPerSec; + pDevice->wasapi.periodSizeInFramesPlayback = data.periodSizeInFramesOut; + ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &pDevice->wasapi.actualPeriodSizeInFramesPlayback); + } - /* Get the internal channel map based on the channel mask. */ - if (pActualFormat->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE) { - ma_channel_mask_to_channel_map__win32(pActualFormat->dwChannelMask, pDevice->capture.internalChannels, pDevice->capture.internalChannelMap); - } else { - ma_channel_mask_to_channel_map__win32(wf.dwChannelMask, pDevice->capture.internalChannels, pDevice->capture.internalChannelMap); + /* + We need to get notifications of when the default device changes. We do this through a device enumerator by + registering a IMMNotificationClient with it. We only care about this if it's the default device. + */ +#ifdef MA_WIN32_DESKTOP + if (pConfig->wasapi.noAutoStreamRouting == MA_FALSE) { + if ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.pDeviceID == NULL) { + pDevice->wasapi.allowCaptureAutoStreamRouting = MA_TRUE; + } + if ((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.pDeviceID == NULL) { + pDevice->wasapi.allowPlaybackAutoStreamRouting = MA_TRUE; } - /* - After getting the actual format the size of the buffer in frames may have actually changed. However, we want this to be as close to what the - user has asked for as possible, so let's go ahead and release the old capture buffer and create a new one in this case. - */ - if (bufferSizeInFrames != (descDS.dwBufferBytes / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels))) { - descDS.dwBufferBytes = bufferSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.internalFormat, wf.Format.nChannels); - ma_IDirectSoundCaptureBuffer_Release((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer); + if (pDevice->wasapi.allowCaptureAutoStreamRouting || pDevice->wasapi.allowPlaybackAutoStreamRouting) { + ma_IMMDeviceEnumerator* pDeviceEnumerator; + HRESULT hr = ma_CoCreateInstance(pContext, MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator); + if (FAILED(hr)) { + ma_device_uninit__wasapi(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create device enumerator.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - if (FAILED(ma_IDirectSoundCapture_CreateCaptureBuffer((ma_IDirectSoundCapture*)pDevice->dsound.pCapture, &descDS, (ma_IDirectSoundCaptureBuffer**)&pDevice->dsound.pCaptureBuffer, NULL))) { - ma_device_uninit__dsound(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Second attempt at IDirectSoundCapture_CreateCaptureBuffer() failed for capture device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + pDevice->wasapi.notificationClient.lpVtbl = (void*)&g_maNotificationCientVtbl; + pDevice->wasapi.notificationClient.counter = 1; + pDevice->wasapi.notificationClient.pDevice = pDevice; + + hr = pDeviceEnumerator->lpVtbl->RegisterEndpointNotificationCallback(pDeviceEnumerator, &pDevice->wasapi.notificationClient); + if (SUCCEEDED(hr)) { + pDevice->wasapi.pDeviceEnumerator = (ma_ptr)pDeviceEnumerator; + } else { + /* Not the end of the world if we fail to register the notification callback. We just won't support automatic stream routing. */ + ma_IMMDeviceEnumerator_Release(pDeviceEnumerator); } } - - /* DirectSound should give us a buffer exactly the size we asked for. */ - pDevice->capture.internalBufferSizeInFrames = bufferSizeInFrames; - pDevice->capture.internalPeriods = pConfig->periods; } +#endif - if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { - WAVEFORMATEXTENSIBLE wf; - MA_DSBUFFERDESC descDSPrimary; - MA_DSCAPS caps; - char rawdata[1024]; /* <-- Ugly hack to avoid a malloc() due to a crappy DirectSound API. */ - WAVEFORMATEXTENSIBLE* pActualFormat; - ma_uint32 bufferSizeInFrames; - MA_DSBUFFERDESC descDS; + ma_atomic_exchange_32(&pDevice->wasapi.isStartedCapture, MA_FALSE); + ma_atomic_exchange_32(&pDevice->wasapi.isStartedPlayback, MA_FALSE); - result = ma_config_to_WAVEFORMATEXTENSIBLE(pConfig->playback.format, pConfig->playback.channels, pConfig->sampleRate, pConfig->playback.channelMap, &wf); - if (result != MA_SUCCESS) { - return result; - } + return MA_SUCCESS; +} - result = ma_context_create_IDirectSound__dsound(pContext, pConfig->playback.shareMode, pConfig->playback.pDeviceID, (ma_IDirectSound**)&pDevice->dsound.pPlayback); - if (result != MA_SUCCESS) { - ma_device_uninit__dsound(pDevice); - return result; - } +static ma_result ma_device__get_available_frames__wasapi(ma_device* pDevice, ma_IAudioClient* pAudioClient, ma_uint32* pFrameCount) +{ + ma_uint32 paddingFramesCount; + HRESULT hr; + ma_share_mode shareMode; - ma_zero_object(&descDSPrimary); - descDSPrimary.dwSize = sizeof(MA_DSBUFFERDESC); - descDSPrimary.dwFlags = MA_DSBCAPS_PRIMARYBUFFER | MA_DSBCAPS_CTRLVOLUME; - if (FAILED(ma_IDirectSound_CreateSoundBuffer((ma_IDirectSound*)pDevice->dsound.pPlayback, &descDSPrimary, (ma_IDirectSoundBuffer**)&pDevice->dsound.pPlaybackPrimaryBuffer, NULL))) { - ma_device_uninit__dsound(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_CreateSoundBuffer() failed for playback device's primary buffer.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } + MA_ASSERT(pDevice != NULL); + MA_ASSERT(pFrameCount != NULL); + + *pFrameCount = 0; + if ((ma_ptr)pAudioClient != pDevice->wasapi.pAudioClientPlayback && (ma_ptr)pAudioClient != pDevice->wasapi.pAudioClientCapture) { + return MA_INVALID_OPERATION; + } - /* We may want to make some adjustments to the format if we are using defaults. */ - ma_zero_object(&caps); - caps.dwSize = sizeof(caps); - if (FAILED(ma_IDirectSound_GetCaps((ma_IDirectSound*)pDevice->dsound.pPlayback, &caps))) { - ma_device_uninit__dsound(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_GetCaps() failed for playback device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + hr = ma_IAudioClient_GetCurrentPadding(pAudioClient, &paddingFramesCount); + if (FAILED(hr)) { + return MA_DEVICE_UNAVAILABLE; + } + + /* Slightly different rules for exclusive and shared modes. */ + shareMode = ((ma_ptr)pAudioClient == pDevice->wasapi.pAudioClientPlayback) ? pDevice->playback.shareMode : pDevice->capture.shareMode; + if (shareMode == ma_share_mode_exclusive) { + *pFrameCount = paddingFramesCount; + } else { + if ((ma_ptr)pAudioClient == pDevice->wasapi.pAudioClientPlayback) { + *pFrameCount = pDevice->wasapi.actualPeriodSizeInFramesPlayback - paddingFramesCount; + } else { + *pFrameCount = paddingFramesCount; } + } - if (pDevice->playback.usingDefaultChannels) { - if ((caps.dwFlags & MA_DSCAPS_PRIMARYSTEREO) != 0) { - DWORD speakerConfig; + return MA_SUCCESS; +} - /* It supports at least stereo, but could support more. */ - wf.Format.nChannels = 2; +static ma_bool32 ma_device_is_reroute_required__wasapi(ma_device* pDevice, ma_device_type deviceType) +{ + MA_ASSERT(pDevice != NULL); + + if (deviceType == ma_device_type_playback) { + return pDevice->wasapi.hasDefaultPlaybackDeviceChanged; + } + + if (deviceType == ma_device_type_capture || deviceType == ma_device_type_loopback) { + return pDevice->wasapi.hasDefaultCaptureDeviceChanged; + } + + return MA_FALSE; +} - /* Look at the speaker configuration to get a better idea on the channel count. */ - if (SUCCEEDED(ma_IDirectSound_GetSpeakerConfig((ma_IDirectSound*)pDevice->dsound.pPlayback, &speakerConfig))) { - ma_get_channels_from_speaker_config__dsound(speakerConfig, &wf.Format.nChannels, &wf.dwChannelMask); - } - } else { - /* It does not support stereo, which means we are stuck with mono. */ - wf.Format.nChannels = 1; - } - } +static ma_result ma_device_reroute__wasapi(ma_device* pDevice, ma_device_type deviceType) +{ + ma_result result; - if (pDevice->usingDefaultSampleRate) { - /* We base the sample rate on the values returned by GetCaps(). */ - if ((caps.dwFlags & MA_DSCAPS_CONTINUOUSRATE) != 0) { - wf.Format.nSamplesPerSec = ma_get_best_sample_rate_within_range(caps.dwMinSecondarySampleRate, caps.dwMaxSecondarySampleRate); - } else { - wf.Format.nSamplesPerSec = caps.dwMaxSecondarySampleRate; - } - } + if (deviceType == ma_device_type_duplex) { + return MA_INVALID_ARGS; + } - wf.Format.nBlockAlign = (wf.Format.nChannels * wf.Format.wBitsPerSample) / 8; - wf.Format.nAvgBytesPerSec = wf.Format.nBlockAlign * wf.Format.nSamplesPerSec; + if (deviceType == ma_device_type_playback) { + ma_atomic_exchange_32(&pDevice->wasapi.hasDefaultPlaybackDeviceChanged, MA_FALSE); + } + if (deviceType == ma_device_type_capture || deviceType == ma_device_type_loopback) { + ma_atomic_exchange_32(&pDevice->wasapi.hasDefaultCaptureDeviceChanged, MA_FALSE); + } + - /* - From MSDN: - - The method succeeds even if the hardware does not support the requested format; DirectSound sets the buffer to the closest - supported format. To determine whether this has happened, an application can call the GetFormat method for the primary buffer - and compare the result with the format that was requested with the SetFormat method. - */ - if (FAILED(ma_IDirectSoundBuffer_SetFormat((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer, (WAVEFORMATEX*)&wf))) { - ma_device_uninit__dsound(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to set format of playback device's primary buffer.", MA_FORMAT_NOT_SUPPORTED); - } + #ifdef MA_DEBUG_OUTPUT + printf("=== CHANGING DEVICE ===\n"); + #endif - /* Get the _actual_ properties of the buffer. */ - pActualFormat = (WAVEFORMATEXTENSIBLE*)rawdata; - if (FAILED(ma_IDirectSoundBuffer_GetFormat((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer, (WAVEFORMATEX*)pActualFormat, sizeof(rawdata), NULL))) { - ma_device_uninit__dsound(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to retrieve the actual format of the playback device's primary buffer.", MA_FORMAT_NOT_SUPPORTED); - } + result = ma_device_reinit__wasapi(pDevice, deviceType); + if (result != MA_SUCCESS) { + return result; + } - pDevice->playback.internalFormat = ma_format_from_WAVEFORMATEX((WAVEFORMATEX*)pActualFormat); - pDevice->playback.internalChannels = pActualFormat->Format.nChannels; - pDevice->playback.internalSampleRate = pActualFormat->Format.nSamplesPerSec; + ma_device__post_init_setup(pDevice, deviceType); - /* Get the internal channel map based on the channel mask. */ - if (pActualFormat->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE) { - ma_channel_mask_to_channel_map__win32(pActualFormat->dwChannelMask, pDevice->playback.internalChannels, pDevice->playback.internalChannelMap); - } else { - ma_channel_mask_to_channel_map__win32(wf.dwChannelMask, pDevice->playback.internalChannels, pDevice->playback.internalChannelMap); - } + return MA_SUCCESS; +} - /* The size of the buffer must be a clean multiple of the period count. */ - bufferSizeInFrames = (ma_calculate_buffer_size_in_frames_from_milliseconds(bufferSizeInMilliseconds, pDevice->playback.internalSampleRate) / pConfig->periods) * pConfig->periods; - /* - Meaning of dwFlags (from MSDN): - - DSBCAPS_CTRLPOSITIONNOTIFY - The buffer has position notification capability. - - DSBCAPS_GLOBALFOCUS - With this flag set, an application using DirectSound can continue to play its buffers if the user switches focus to - another application, even if the new application uses DirectSound. - - DSBCAPS_GETCURRENTPOSITION2 - In the first version of DirectSound, the play cursor was significantly ahead of the actual playing sound on emulated - sound cards; it was directly behind the write cursor. Now, if the DSBCAPS_GETCURRENTPOSITION2 flag is specified, the - application can get a more accurate play cursor. - */ - ma_zero_object(&descDS); - descDS.dwSize = sizeof(descDS); - descDS.dwFlags = MA_DSBCAPS_CTRLPOSITIONNOTIFY | MA_DSBCAPS_GLOBALFOCUS | MA_DSBCAPS_GETCURRENTPOSITION2; - descDS.dwBufferBytes = bufferSizeInFrames * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - descDS.lpwfxFormat = (WAVEFORMATEX*)&wf; - if (FAILED(ma_IDirectSound_CreateSoundBuffer((ma_IDirectSound*)pDevice->dsound.pPlayback, &descDS, (ma_IDirectSoundBuffer**)&pDevice->dsound.pPlaybackBuffer, NULL))) { - ma_device_uninit__dsound(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_CreateSoundBuffer() failed for playback device's secondary buffer.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } +static ma_result ma_device_stop__wasapi(ma_device* pDevice) +{ + MA_ASSERT(pDevice != NULL); - /* DirectSound should give us a buffer exactly the size we asked for. */ - pDevice->playback.internalBufferSizeInFrames = bufferSizeInFrames; - pDevice->playback.internalPeriods = pConfig->periods; + /* + We need to explicitly signal the capture event in loopback mode to ensure we return from WaitForSingleObject() when nothing is being played. When nothing + is being played, the event is never signalled internally by WASAPI which means we will deadlock when stopping the device. + */ + if (pDevice->type == ma_device_type_loopback) { + SetEvent((HANDLE)pDevice->wasapi.hEventCapture); } - (void)pContext; return MA_SUCCESS; } -ma_result ma_device_main_loop__dsound(ma_device* pDevice) +static ma_result ma_device_main_loop__wasapi(ma_device* pDevice) { - ma_result result = MA_SUCCESS; - ma_uint32 bpfCapture = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 bpfPlayback = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + ma_result result; HRESULT hr; - DWORD lockOffsetInBytesCapture; - DWORD lockSizeInBytesCapture; - DWORD mappedSizeInBytesCapture; - void* pMappedBufferCapture; - DWORD lockOffsetInBytesPlayback; - DWORD lockSizeInBytesPlayback; - DWORD mappedSizeInBytesPlayback; - void* pMappedBufferPlayback; - DWORD prevReadCursorInBytesCapture = 0; - DWORD prevPlayCursorInBytesPlayback = 0; - ma_bool32 physicalPlayCursorLoopFlagPlayback = 0; - DWORD virtualWriteCursorInBytesPlayback = 0; - ma_bool32 virtualWriteCursorLoopFlagPlayback = 0; - ma_bool32 isPlaybackDeviceStarted = MA_FALSE; - ma_uint32 framesWrittenToPlaybackDevice = 0; /* For knowing whether or not the playback device needs to be started. */ - ma_uint32 waitTimeInMilliseconds = 1; + ma_bool32 exitLoop = MA_FALSE; + ma_uint32 framesWrittenToPlaybackDevice = 0; + ma_uint32 mappedDeviceBufferSizeInFramesCapture = 0; + ma_uint32 mappedDeviceBufferSizeInFramesPlayback = 0; + ma_uint32 mappedDeviceBufferFramesRemainingCapture = 0; + ma_uint32 mappedDeviceBufferFramesRemainingPlayback = 0; + BYTE* pMappedDeviceBufferCapture = NULL; + BYTE* pMappedDeviceBufferPlayback = NULL; + ma_uint32 bpfCaptureDevice = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + ma_uint32 bpfPlaybackDevice = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + ma_uint32 bpfCaptureClient = ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); + ma_uint32 bpfPlaybackClient = ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); + ma_uint8 inputDataInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 inputDataInClientFormatCap = sizeof(inputDataInClientFormat) / bpfCaptureClient; + ma_uint8 outputDataInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 outputDataInClientFormatCap = sizeof(outputDataInClientFormat) / bpfPlaybackClient; + ma_uint32 outputDataInClientFormatCount = 0; + ma_uint32 outputDataInClientFormatConsumed = 0; + ma_uint32 periodSizeInFramesCapture = 0; - ma_assert(pDevice != NULL); + MA_ASSERT(pDevice != NULL); - /* The first thing to do is start the capture device. The playback device is only started after the first period is written. */ - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - if (FAILED(ma_IDirectSoundCaptureBuffer_Start((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, MA_DSCBSTART_LOOPING))) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCaptureBuffer_Start() failed.", MA_FAILED_TO_START_BACKEND_DEVICE); + /* The capture device needs to be started immediately. */ + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) { + periodSizeInFramesCapture = pDevice->capture.internalPeriodSizeInFrames; + + hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); + if (FAILED(hr)) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal capture device.", MA_FAILED_TO_START_BACKEND_DEVICE); } + ma_atomic_exchange_32(&pDevice->wasapi.isStartedCapture, MA_TRUE); } - - while (ma_device__get_state(pDevice) == MA_STATE_STARTED) { + + while (ma_device__get_state(pDevice) == MA_STATE_STARTED && !exitLoop) { + /* We may need to reroute the device. */ + if (ma_device_is_reroute_required__wasapi(pDevice, ma_device_type_playback)) { + result = ma_device_reroute__wasapi(pDevice, ma_device_type_playback); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } + } + if (ma_device_is_reroute_required__wasapi(pDevice, ma_device_type_capture)) { + result = ma_device_reroute__wasapi(pDevice, (pDevice->type == ma_device_type_loopback) ? ma_device_type_loopback : ma_device_type_capture); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } + } + switch (pDevice->type) { case ma_device_type_duplex: { - DWORD physicalCaptureCursorInBytes; - DWORD physicalReadCursorInBytes; - if (FAILED(ma_IDirectSoundCaptureBuffer_GetCurrentPosition((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, &physicalCaptureCursorInBytes, &physicalReadCursorInBytes))) { - return MA_ERROR; - } + ma_uint32 framesAvailableCapture; + ma_uint32 framesAvailablePlayback; + DWORD flagsCapture; /* Passed to IAudioCaptureClient_GetBuffer(). */ - /* If nothing is available we just sleep for a bit and return from this iteration. */ - if (physicalReadCursorInBytes == prevReadCursorInBytesCapture) { - ma_sleep(waitTimeInMilliseconds); - continue; /* Nothing is available in the capture buffer. */ - } + /* The process is to map the playback buffer and fill it as quickly as possible from input data. */ + if (pMappedDeviceBufferPlayback == NULL) { + /* WASAPI is weird with exclusive mode. You need to wait on the event _before_ querying the available frames. */ + if (pDevice->playback.shareMode == ma_share_mode_exclusive) { + if (WaitForSingleObject(pDevice->wasapi.hEventPlayback, INFINITE) == WAIT_FAILED) { + return MA_ERROR; /* Wait failed. */ + } + } - /* - The current position has moved. We need to map all of the captured samples and write them to the playback device, making sure - we don't return until every frame has been copied over. - */ - if (prevReadCursorInBytesCapture < physicalReadCursorInBytes) { - /* The capture position has not looped. This is the simple case. */ - lockOffsetInBytesCapture = prevReadCursorInBytesCapture; - lockSizeInBytesCapture = (physicalReadCursorInBytes - prevReadCursorInBytesCapture); - } else { - /* - The capture position has looped. This is the more complex case. Map to the end of the buffer. If this does not return anything, - do it again from the start. - */ - if (prevReadCursorInBytesCapture < pDevice->capture.internalBufferSizeInFrames*bpfCapture) { - /* Lock up to the end of the buffer. */ - lockOffsetInBytesCapture = prevReadCursorInBytesCapture; - lockSizeInBytesCapture = (pDevice->capture.internalBufferSizeInFrames*bpfCapture) - prevReadCursorInBytesCapture; - } else { - /* Lock starting from the start of the buffer. */ - lockOffsetInBytesCapture = 0; - lockSizeInBytesCapture = physicalReadCursorInBytes; + result = ma_device__get_available_frames__wasapi(pDevice, (ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &framesAvailablePlayback); + if (result != MA_SUCCESS) { + return result; } - } - if (lockSizeInBytesCapture == 0) { - ma_sleep(waitTimeInMilliseconds); - continue; /* Nothing is available in the capture buffer. */ - } + /*printf("TRACE 1: framesAvailablePlayback=%d\n", framesAvailablePlayback);*/ - hr = ma_IDirectSoundCaptureBuffer_Lock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, lockOffsetInBytesCapture, lockSizeInBytesCapture, &pMappedBufferCapture, &mappedSizeInBytesCapture, NULL, NULL, 0); - if (FAILED(hr)) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from capture device in preparation for writing to the device.", MA_FAILED_TO_MAP_DEVICE_BUFFER); - } + /* In exclusive mode, the frame count needs to exactly match the value returned by GetCurrentPadding(). */ + if (pDevice->playback.shareMode != ma_share_mode_exclusive) { + if (framesAvailablePlayback > pDevice->wasapi.periodSizeInFramesPlayback) { + framesAvailablePlayback = pDevice->wasapi.periodSizeInFramesPlayback; + } + } + + /* If there's no frames available in the playback device we need to wait for more. */ + if (framesAvailablePlayback == 0) { + /* In exclusive mode we waited at the top. */ + if (pDevice->playback.shareMode != ma_share_mode_exclusive) { + if (WaitForSingleObject(pDevice->wasapi.hEventPlayback, INFINITE) == WAIT_FAILED) { + return MA_ERROR; /* Wait failed. */ + } + } + + continue; + } - /* At this point we have some input data that we need to output. We do not return until every mapped frame of the input data is written to the playback device. */ - pDevice->capture._dspFrameCount = mappedSizeInBytesCapture / bpfCapture; - pDevice->capture._dspFrames = (const ma_uint8*)pMappedBufferCapture; - for (;;) { /* Keep writing to the playback device. */ - ma_uint8 inputFramesInExternalFormat[4096]; - ma_uint32 inputFramesInExternalFormatCap = sizeof(inputFramesInExternalFormat) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); - ma_uint32 inputFramesInExternalFormatCount; - ma_uint8 outputFramesInExternalFormat[4096]; - ma_uint32 outputFramesInExternalFormatCap = sizeof(outputFramesInExternalFormat) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); - - inputFramesInExternalFormatCount = (ma_uint32)ma_pcm_converter_read(&pDevice->capture.converter, inputFramesInExternalFormat, ma_min(inputFramesInExternalFormatCap, outputFramesInExternalFormatCap)); - if (inputFramesInExternalFormatCount == 0) { - break; /* No more input data. */ + /* We're ready to map the playback device's buffer. We don't release this until it's been entirely filled. */ + hr = ma_IAudioRenderClient_GetBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, framesAvailablePlayback, &pMappedDeviceBufferPlayback); + if (FAILED(hr)) { + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from playback device in preparation for writing to the device.", MA_FAILED_TO_MAP_DEVICE_BUFFER); + exitLoop = MA_TRUE; + break; } - ma_device__on_data(pDevice, outputFramesInExternalFormat, inputFramesInExternalFormat, inputFramesInExternalFormatCount); + mappedDeviceBufferSizeInFramesPlayback = framesAvailablePlayback; + mappedDeviceBufferFramesRemainingPlayback = framesAvailablePlayback; + } - /* At this point we have input and output data in external format. All we need to do now is convert it to the output format. This may take a few passes. */ - pDevice->playback._dspFrameCount = inputFramesInExternalFormatCount; - pDevice->playback._dspFrames = (const ma_uint8*)outputFramesInExternalFormat; - for (;;) { - ma_uint32 framesWrittenThisIteration; - DWORD physicalPlayCursorInBytes; - DWORD physicalWriteCursorInBytes; - DWORD availableBytesPlayback; - DWORD silentPaddingInBytes = 0; /* <-- Must be initialized to 0. */ + /* At this point we should have a buffer available for output. We need to keep writing input samples to it. */ + for (;;) { + /* Try grabbing some captured data if we haven't already got a mapped buffer. */ + if (pMappedDeviceBufferCapture == NULL) { + if (pDevice->capture.shareMode == ma_share_mode_shared) { + if (WaitForSingleObject(pDevice->wasapi.hEventCapture, INFINITE) == WAIT_FAILED) { + return MA_ERROR; /* Wait failed. */ + } + } + + result = ma_device__get_available_frames__wasapi(pDevice, (ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, &framesAvailableCapture); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } + + /*printf("TRACE 2: framesAvailableCapture=%d\n", framesAvailableCapture);*/ + + /* Wait for more if nothing is available. */ + if (framesAvailableCapture == 0) { + /* In exclusive mode we waited at the top. */ + if (pDevice->capture.shareMode != ma_share_mode_shared) { + if (WaitForSingleObject(pDevice->wasapi.hEventCapture, INFINITE) == WAIT_FAILED) { + return MA_ERROR; /* Wait failed. */ + } + } - /* We need the physical play and write cursors. */ - if (FAILED(ma_IDirectSoundBuffer_GetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &physicalPlayCursorInBytes, &physicalWriteCursorInBytes))) { - break; + continue; } - if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) { - physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback; + /* Getting here means there's data available for writing to the output device. */ + mappedDeviceBufferSizeInFramesCapture = ma_min(framesAvailableCapture, periodSizeInFramesCapture); + hr = ma_IAudioCaptureClient_GetBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, (BYTE**)&pMappedDeviceBufferCapture, &mappedDeviceBufferSizeInFramesCapture, &flagsCapture, NULL, NULL); + if (FAILED(hr)) { + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from capture device in preparation for writing to the device.", MA_FAILED_TO_MAP_DEVICE_BUFFER); + exitLoop = MA_TRUE; + break; } - prevPlayCursorInBytesPlayback = physicalPlayCursorInBytes; - /* If there's any bytes available for writing we can do that now. The space between the virtual cursor position and play cursor. */ - if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) { - /* Same loop iteration. The available bytes wraps all the way around from the virtual write cursor to the physical play cursor. */ - if (physicalPlayCursorInBytes <= virtualWriteCursorInBytesPlayback) { - availableBytesPlayback = (pDevice->playback.internalBufferSizeInFrames*bpfPlayback) - virtualWriteCursorInBytesPlayback; - availableBytesPlayback += physicalPlayCursorInBytes; /* Wrap around. */ - } else { - /* This is an error. */ + + /* Overrun detection. */ + if ((flagsCapture & MA_AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY) != 0) { + /* Glitched. Probably due to an overrun. */ + #ifdef MA_DEBUG_OUTPUT + printf("[WASAPI] Data discontinuity (possible overrun). framesAvailableCapture=%d, mappedBufferSizeInFramesCapture=%d\n", framesAvailableCapture, mappedDeviceBufferSizeInFramesCapture); + #endif + + /* + Exeriment: If we get an overrun it probably means we're straddling the end of the buffer. In order to prevent a never-ending sequence of glitches let's experiment + by dropping every frame until we're left with only a single period. To do this we just keep retrieving and immediately releasing buffers until we're down to the + last period. + */ + if (framesAvailableCapture >= pDevice->wasapi.actualPeriodSizeInFramesCapture) { #ifdef MA_DEBUG_OUTPUT - printf("[DirectSound] (Duplex/Playback) WARNING: Play cursor has moved in front of the write cursor (same loop iterations). physicalPlayCursorInBytes=%d, virtualWriteCursorInBytes=%d.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback); + printf("[WASAPI] Synchronizing capture stream. "); #endif - availableBytesPlayback = 0; - } - } else { - /* Different loop iterations. The available bytes only goes from the virtual write cursor to the physical play cursor. */ - if (physicalPlayCursorInBytes >= virtualWriteCursorInBytesPlayback) { - availableBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback; - } else { - /* This is an error. */ + do + { + hr = ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, mappedDeviceBufferSizeInFramesCapture); + if (FAILED(hr)) { + break; + } + + framesAvailableCapture -= mappedDeviceBufferSizeInFramesCapture; + + if (framesAvailableCapture > 0) { + mappedDeviceBufferSizeInFramesCapture = ma_min(framesAvailableCapture, periodSizeInFramesCapture); + hr = ma_IAudioCaptureClient_GetBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, (BYTE**)&pMappedDeviceBufferCapture, &mappedDeviceBufferSizeInFramesCapture, &flagsCapture, NULL, NULL); + if (FAILED(hr)) { + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from capture device in preparation for writing to the device.", MA_FAILED_TO_MAP_DEVICE_BUFFER); + exitLoop = MA_TRUE; + break; + } + } else { + pMappedDeviceBufferCapture = NULL; + mappedDeviceBufferSizeInFramesCapture = 0; + } + } while (framesAvailableCapture > periodSizeInFramesCapture); #ifdef MA_DEBUG_OUTPUT - printf("[DirectSound] (Duplex/Playback) WARNING: Write cursor has moved behind the play cursor (different loop iterations). physicalPlayCursorInBytes=%d, virtualWriteCursorInBytes=%d.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback); + printf("framesAvailableCapture=%d, mappedBufferSizeInFramesCapture=%d\n", framesAvailableCapture, mappedDeviceBufferSizeInFramesCapture); #endif - availableBytesPlayback = 0; } + } else { + #ifdef MA_DEBUG_OUTPUT + if (flagsCapture != 0) { + printf("[WASAPI] Capture Flags: %d\n", flagsCapture); + } + #endif } - #ifdef MA_DEBUG_OUTPUT - /*printf("[DirectSound] (Duplex/Playback) physicalPlayCursorInBytes=%d, availableBytesPlayback=%d\n", physicalPlayCursorInBytes, availableBytesPlayback);*/ - #endif + mappedDeviceBufferFramesRemainingCapture = mappedDeviceBufferSizeInFramesCapture; + } - /* If there's no room available for writing we need to wait for more. */ - if (availableBytesPlayback == 0) { - /* If we haven't started the device yet, this will never get beyond 0. In this case we need to get the device started. */ - if (!isPlaybackDeviceStarted) { - if (FAILED(ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING))) { - ma_IDirectSoundCaptureBuffer_Stop((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.", MA_FAILED_TO_START_BACKEND_DEVICE); - } - isPlaybackDeviceStarted = MA_TRUE; - } else { - ma_sleep(waitTimeInMilliseconds); - continue; - } - } + /* At this point we should have both input and output data available. We now need to convert the data and post it to the client. */ + for (;;) { + BYTE* pRunningDeviceBufferCapture; + BYTE* pRunningDeviceBufferPlayback; + ma_uint32 framesToProcess; + ma_uint32 framesProcessed; - /* Getting here means there room available somewhere. We limit this to either the end of the buffer or the physical play cursor, whichever is closest. */ - lockOffsetInBytesPlayback = virtualWriteCursorInBytesPlayback; - if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) { - /* Same loop iteration. Go up to the end of the buffer. */ - lockSizeInBytesPlayback = (pDevice->playback.internalBufferSizeInFrames*bpfPlayback) - virtualWriteCursorInBytesPlayback; - } else { - /* Different loop iterations. Go up to the physical play cursor. */ - lockSizeInBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback; - } + pRunningDeviceBufferCapture = pMappedDeviceBufferCapture + ((mappedDeviceBufferSizeInFramesCapture - mappedDeviceBufferFramesRemainingCapture ) * bpfCaptureDevice); + pRunningDeviceBufferPlayback = pMappedDeviceBufferPlayback + ((mappedDeviceBufferSizeInFramesPlayback - mappedDeviceBufferFramesRemainingPlayback) * bpfPlaybackDevice); + + /* There may be some data sitting in the converter that needs to be processed first. Once this is exhaused, run the data callback again. */ + if (!pDevice->playback.converter.isPassthrough && outputDataInClientFormatConsumed < outputDataInClientFormatCount) { + ma_uint64 convertedFrameCountClient = (outputDataInClientFormatCount - outputDataInClientFormatConsumed); + ma_uint64 convertedFrameCountDevice = mappedDeviceBufferFramesRemainingPlayback; + void* pConvertedFramesClient = outputDataInClientFormat + (outputDataInClientFormatConsumed * bpfPlaybackClient); + void* pConvertedFramesDevice = pRunningDeviceBufferPlayback; + result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, pConvertedFramesClient, &convertedFrameCountClient, pConvertedFramesDevice, &convertedFrameCountDevice); + if (result != MA_SUCCESS) { + break; + } - hr = ma_IDirectSoundBuffer_Lock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, lockOffsetInBytesPlayback, lockSizeInBytesPlayback, &pMappedBufferPlayback, &mappedSizeInBytesPlayback, NULL, NULL, 0); - if (FAILED(hr)) { - result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from playback device in preparation for writing to the device.", MA_FAILED_TO_MAP_DEVICE_BUFFER); - break; + outputDataInClientFormatConsumed += (ma_uint32)convertedFrameCountClient; /* Safe cast. */ + mappedDeviceBufferFramesRemainingPlayback -= (ma_uint32)convertedFrameCountDevice; /* Safe cast. */ + + if (mappedDeviceBufferFramesRemainingPlayback == 0) { + break; + } } /* - Experiment: If the playback buffer is being starved, pad it with some silence to get it back in sync. This will cause a glitch, but it may prevent - endless glitching due to it constantly running out of data. + Getting here means we need to fire the callback. If format conversion is unnecessary, we can optimize this by passing the pointers to the internal + buffers directly to the callback. */ - if (isPlaybackDeviceStarted) { - DWORD bytesQueuedForPlayback = (pDevice->playback.internalBufferSizeInFrames*bpfPlayback) - availableBytesPlayback; - if (bytesQueuedForPlayback < ((pDevice->playback.internalBufferSizeInFrames/pDevice->playback.internalPeriods)*bpfPlayback)) { - silentPaddingInBytes = ((pDevice->playback.internalBufferSizeInFrames/pDevice->playback.internalPeriods)*2*bpfPlayback) - bytesQueuedForPlayback; - if (silentPaddingInBytes > lockSizeInBytesPlayback) { - silentPaddingInBytes = lockSizeInBytesPlayback; - } + if (pDevice->capture.converter.isPassthrough && pDevice->playback.converter.isPassthrough) { + /* Optimal path. We can pass mapped pointers directly to the callback. */ + framesToProcess = ma_min(mappedDeviceBufferFramesRemainingCapture, mappedDeviceBufferFramesRemainingPlayback); + framesProcessed = framesToProcess; - #ifdef MA_DEBUG_OUTPUT - printf("[DirectSound] (Duplex/Playback) Playback buffer starved. availableBytesPlayback=%d, silentPaddingInBytes=%d\n", availableBytesPlayback, silentPaddingInBytes); - #endif + ma_device__on_data(pDevice, pRunningDeviceBufferPlayback, pRunningDeviceBufferCapture, framesToProcess); + + mappedDeviceBufferFramesRemainingCapture -= framesProcessed; + mappedDeviceBufferFramesRemainingPlayback -= framesProcessed; + + if (mappedDeviceBufferFramesRemainingCapture == 0) { + break; /* Exhausted input data. */ } - } + if (mappedDeviceBufferFramesRemainingPlayback == 0) { + break; /* Exhausted output data. */ + } + } else if (pDevice->capture.converter.isPassthrough) { + /* The input buffer is a passthrough, but the playback buffer requires a conversion. */ + framesToProcess = ma_min(mappedDeviceBufferFramesRemainingCapture, outputDataInClientFormatCap); + framesProcessed = framesToProcess; - /* At this point we have a buffer for output. */ - if (silentPaddingInBytes > 0) { - ma_zero_memory(pMappedBufferPlayback, silentPaddingInBytes); - framesWrittenThisIteration = silentPaddingInBytes/bpfPlayback; + ma_device__on_data(pDevice, outputDataInClientFormat, pRunningDeviceBufferCapture, framesToProcess); + outputDataInClientFormatCount = framesProcessed; + outputDataInClientFormatConsumed = 0; + + mappedDeviceBufferFramesRemainingCapture -= framesProcessed; + if (mappedDeviceBufferFramesRemainingCapture == 0) { + break; /* Exhausted input data. */ + } + } else if (pDevice->playback.converter.isPassthrough) { + /* The input buffer requires conversion, the playback buffer is passthrough. */ + ma_uint64 capturedDeviceFramesToProcess = mappedDeviceBufferFramesRemainingCapture; + ma_uint64 capturedClientFramesToProcess = ma_min(inputDataInClientFormatCap, mappedDeviceBufferFramesRemainingPlayback); + + result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningDeviceBufferCapture, &capturedDeviceFramesToProcess, inputDataInClientFormat, &capturedClientFramesToProcess); + if (result != MA_SUCCESS) { + break; + } + + if (capturedClientFramesToProcess == 0) { + break; + } + + ma_device__on_data(pDevice, pRunningDeviceBufferPlayback, inputDataInClientFormat, (ma_uint32)capturedClientFramesToProcess); /* Safe cast. */ + + mappedDeviceBufferFramesRemainingCapture -= (ma_uint32)capturedDeviceFramesToProcess; + mappedDeviceBufferFramesRemainingPlayback -= (ma_uint32)capturedClientFramesToProcess; } else { - framesWrittenThisIteration = (ma_uint32)ma_pcm_converter_read(&pDevice->playback.converter, pMappedBufferPlayback, mappedSizeInBytesPlayback/bpfPlayback); + ma_uint64 capturedDeviceFramesToProcess = mappedDeviceBufferFramesRemainingCapture; + ma_uint64 capturedClientFramesToProcess = ma_min(inputDataInClientFormatCap, outputDataInClientFormatCap); + + result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningDeviceBufferCapture, &capturedDeviceFramesToProcess, inputDataInClientFormat, &capturedClientFramesToProcess); + if (result != MA_SUCCESS) { + break; + } + + if (capturedClientFramesToProcess == 0) { + break; + } + + ma_device__on_data(pDevice, outputDataInClientFormat, inputDataInClientFormat, (ma_uint32)capturedClientFramesToProcess); + + mappedDeviceBufferFramesRemainingCapture -= (ma_uint32)capturedDeviceFramesToProcess; + outputDataInClientFormatCount = (ma_uint32)capturedClientFramesToProcess; + outputDataInClientFormatConsumed = 0; } - + } + - hr = ma_IDirectSoundBuffer_Unlock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, pMappedBufferPlayback, framesWrittenThisIteration*bpfPlayback, NULL, 0); + /* If at this point we've run out of capture data we need to release the buffer. */ + if (mappedDeviceBufferFramesRemainingCapture == 0 && pMappedDeviceBufferCapture != NULL) { + hr = ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, mappedDeviceBufferSizeInFramesCapture); if (FAILED(hr)) { - result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from playback device after writing to the device.", MA_FAILED_TO_UNMAP_DEVICE_BUFFER); + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to release internal buffer from capture device after reading from the device.", MA_FAILED_TO_UNMAP_DEVICE_BUFFER); + exitLoop = MA_TRUE; break; } - virtualWriteCursorInBytesPlayback += framesWrittenThisIteration*bpfPlayback; - if ((virtualWriteCursorInBytesPlayback/bpfPlayback) == pDevice->playback.internalBufferSizeInFrames) { - virtualWriteCursorInBytesPlayback = 0; - virtualWriteCursorLoopFlagPlayback = !virtualWriteCursorLoopFlagPlayback; - } - - /* - We may need to start the device. We want two full periods to be written before starting the playback device. Having an extra period adds - a bit of a buffer to prevent the playback buffer from getting starved. - */ - framesWrittenToPlaybackDevice += framesWrittenThisIteration; - if (!isPlaybackDeviceStarted && framesWrittenToPlaybackDevice >= ((pDevice->playback.internalBufferSizeInFrames/pDevice->playback.internalPeriods)*2)) { - if (FAILED(ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING))) { - ma_IDirectSoundCaptureBuffer_Stop((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.", MA_FAILED_TO_START_BACKEND_DEVICE); - } - isPlaybackDeviceStarted = MA_TRUE; - } + /*printf("TRACE: Released capture buffer\n");*/ - if (framesWrittenThisIteration < mappedSizeInBytesPlayback/bpfPlayback) { - break; /* We're finished with the output data.*/ - } + pMappedDeviceBufferCapture = NULL; + mappedDeviceBufferFramesRemainingCapture = 0; + mappedDeviceBufferSizeInFramesCapture = 0; } - if (inputFramesInExternalFormatCount < inputFramesInExternalFormatCap) { - break; /* We just consumed every input sample. */ + /* Get out of this loop if we're run out of room in the playback buffer. */ + if (mappedDeviceBufferFramesRemainingPlayback == 0) { + break; } } - /* At this point we're done with the mapped portion of the capture buffer. */ - hr = ma_IDirectSoundCaptureBuffer_Unlock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, pMappedBufferCapture, mappedSizeInBytesCapture, NULL, 0); - if (FAILED(hr)) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from capture device after reading from the device.", MA_FAILED_TO_UNMAP_DEVICE_BUFFER); - } - prevReadCursorInBytesCapture = (lockOffsetInBytesCapture + mappedSizeInBytesCapture); - } break; - + /* If at this point we've run out of data we need to release the buffer. */ + if (mappedDeviceBufferFramesRemainingPlayback == 0 && pMappedDeviceBufferPlayback != NULL) { + hr = ma_IAudioRenderClient_ReleaseBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, mappedDeviceBufferSizeInFramesPlayback, 0); + if (FAILED(hr)) { + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to release internal buffer from playback device after writing to the device.", MA_FAILED_TO_UNMAP_DEVICE_BUFFER); + exitLoop = MA_TRUE; + break; + } + /*printf("TRACE: Released playback buffer\n");*/ + framesWrittenToPlaybackDevice += mappedDeviceBufferSizeInFramesPlayback; - case ma_device_type_capture: - { - DWORD physicalCaptureCursorInBytes; - DWORD physicalReadCursorInBytes; - if (FAILED(ma_IDirectSoundCaptureBuffer_GetCurrentPosition((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, &physicalCaptureCursorInBytes, &physicalReadCursorInBytes))) { - return MA_ERROR; + pMappedDeviceBufferPlayback = NULL; + mappedDeviceBufferFramesRemainingPlayback = 0; + mappedDeviceBufferSizeInFramesPlayback = 0; } - /* If the previous capture position is the same as the current position we need to wait a bit longer. */ - if (prevReadCursorInBytesCapture == physicalReadCursorInBytes) { - ma_sleep(waitTimeInMilliseconds); - continue; - } + if (!pDevice->wasapi.isStartedPlayback) { + ma_uint32 startThreshold = pDevice->playback.internalPeriodSizeInFrames * 1; - /* Getting here means we have capture data available. */ - if (prevReadCursorInBytesCapture < physicalReadCursorInBytes) { - /* The capture position has not looped. This is the simple case. */ - lockOffsetInBytesCapture = prevReadCursorInBytesCapture; - lockSizeInBytesCapture = (physicalReadCursorInBytes - prevReadCursorInBytesCapture); - } else { - /* - The capture position has looped. This is the more complex case. Map to the end of the buffer. If this does not return anything, - do it again from the start. - */ - if (prevReadCursorInBytesCapture < pDevice->capture.internalBufferSizeInFrames*bpfCapture) { - /* Lock up to the end of the buffer. */ - lockOffsetInBytesCapture = prevReadCursorInBytesCapture; - lockSizeInBytesCapture = (pDevice->capture.internalBufferSizeInFrames*bpfCapture) - prevReadCursorInBytesCapture; - } else { - /* Lock starting from the start of the buffer. */ - lockOffsetInBytesCapture = 0; - lockSizeInBytesCapture = physicalReadCursorInBytes; + /* Prevent a deadlock. If we don't clamp against the actual buffer size we'll never end up starting the playback device which will result in a deadlock. */ + if (startThreshold > pDevice->wasapi.actualPeriodSizeInFramesPlayback) { + startThreshold = pDevice->wasapi.actualPeriodSizeInFramesPlayback; + } + + if (pDevice->playback.shareMode == ma_share_mode_exclusive || framesWrittenToPlaybackDevice >= startThreshold) { + hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); + if (FAILED(hr)) { + ma_IAudioClient_Stop((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); + ma_IAudioClient_Reset((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal playback device.", MA_FAILED_TO_START_BACKEND_DEVICE); + } + ma_atomic_exchange_32(&pDevice->wasapi.isStartedPlayback, MA_TRUE); } } + } break; - #ifdef MA_DEBUG_OUTPUT - /*printf("[DirectSound] (Capture) physicalCaptureCursorInBytes=%d, physicalReadCursorInBytes=%d\n", physicalCaptureCursorInBytes, physicalReadCursorInBytes);*/ - /*printf("[DirectSound] (Capture) lockOffsetInBytesCapture=%d, lockSizeInBytesCapture=%d\n", lockOffsetInBytesCapture, lockSizeInBytesCapture);*/ - #endif - if (lockSizeInBytesCapture < (pDevice->capture.internalBufferSizeInFrames/pDevice->capture.internalPeriods)) { - ma_sleep(waitTimeInMilliseconds); - continue; /* Nothing is available in the capture buffer. */ + + case ma_device_type_capture: + case ma_device_type_loopback: + { + ma_uint32 framesAvailableCapture; + DWORD flagsCapture; /* Passed to IAudioCaptureClient_GetBuffer(). */ + + /* Wait for data to become available first. */ + if (WaitForSingleObject(pDevice->wasapi.hEventCapture, INFINITE) == WAIT_FAILED) { + exitLoop = MA_TRUE; + break; /* Wait failed. */ } - hr = ma_IDirectSoundCaptureBuffer_Lock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, lockOffsetInBytesCapture, lockSizeInBytesCapture, &pMappedBufferCapture, &mappedSizeInBytesCapture, NULL, NULL, 0); - if (FAILED(hr)) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from capture device in preparation for writing to the device.", MA_FAILED_TO_MAP_DEVICE_BUFFER); + /* See how many frames are available. Since we waited at the top, I don't think this should ever return 0. I'm checking for this anyway. */ + result = ma_device__get_available_frames__wasapi(pDevice, (ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, &framesAvailableCapture); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; } - #ifdef MA_DEBUG_OUTPUT - if (lockSizeInBytesCapture != mappedSizeInBytesCapture) { - printf("[DirectSound] (Capture) lockSizeInBytesCapture=%d != mappedSizeInBytesCapture=%d\n", lockSizeInBytesCapture, mappedSizeInBytesCapture); + if (framesAvailableCapture < pDevice->wasapi.periodSizeInFramesCapture) { + continue; /* Nothing available. Keep waiting. */ } - #endif - - ma_device__send_frames_to_client(pDevice, mappedSizeInBytesCapture/bpfCapture, pMappedBufferCapture); - hr = ma_IDirectSoundCaptureBuffer_Unlock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, pMappedBufferCapture, mappedSizeInBytesCapture, NULL, 0); + /* Map the data buffer in preparation for sending to the client. */ + mappedDeviceBufferSizeInFramesCapture = framesAvailableCapture; + hr = ma_IAudioCaptureClient_GetBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, (BYTE**)&pMappedDeviceBufferCapture, &mappedDeviceBufferSizeInFramesCapture, &flagsCapture, NULL, NULL); if (FAILED(hr)) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from capture device after reading from the device.", MA_FAILED_TO_UNMAP_DEVICE_BUFFER); + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from capture device in preparation for writing to the device.", MA_FAILED_TO_MAP_DEVICE_BUFFER); + exitLoop = MA_TRUE; + break; } - prevReadCursorInBytesCapture = lockOffsetInBytesCapture + mappedSizeInBytesCapture; - if (prevReadCursorInBytesCapture == (pDevice->capture.internalBufferSizeInFrames*bpfCapture)) { - prevReadCursorInBytesCapture = 0; + /* We should have a buffer at this point. */ + ma_device__send_frames_to_client(pDevice, mappedDeviceBufferSizeInFramesCapture, pMappedDeviceBufferCapture); + + /* At this point we're done with the buffer. */ + hr = ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, mappedDeviceBufferSizeInFramesCapture); + pMappedDeviceBufferCapture = NULL; /* <-- Important. Not doing this can result in an error once we leave this loop because it will use this to know whether or not a final ReleaseBuffer() needs to be called. */ + mappedDeviceBufferSizeInFramesCapture = 0; + if (FAILED(hr)) { + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to release internal buffer from capture device after reading from the device.", MA_FAILED_TO_UNMAP_DEVICE_BUFFER); + exitLoop = MA_TRUE; + break; } } break; @@ -11204,11040 +11374,12043 @@ ma_result ma_device_main_loop__dsound(ma_device* pDevice) case ma_device_type_playback: { - DWORD availableBytesPlayback; - DWORD physicalPlayCursorInBytes; - DWORD physicalWriteCursorInBytes; - if (FAILED(ma_IDirectSoundBuffer_GetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &physicalPlayCursorInBytes, &physicalWriteCursorInBytes))) { - break; - } - - if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) { - physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback; - } - prevPlayCursorInBytesPlayback = physicalPlayCursorInBytes; + ma_uint32 framesAvailablePlayback; - /* If there's any bytes available for writing we can do that now. The space between the virtual cursor position and play cursor. */ - if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) { - /* Same loop iteration. The available bytes wraps all the way around from the virtual write cursor to the physical play cursor. */ - if (physicalPlayCursorInBytes <= virtualWriteCursorInBytesPlayback) { - availableBytesPlayback = (pDevice->playback.internalBufferSizeInFrames*bpfPlayback) - virtualWriteCursorInBytesPlayback; - availableBytesPlayback += physicalPlayCursorInBytes; /* Wrap around. */ - } else { - /* This is an error. */ - #ifdef MA_DEBUG_OUTPUT - printf("[DirectSound] (Playback) WARNING: Play cursor has moved in front of the write cursor (same loop iterations). physicalPlayCursorInBytes=%d, virtualWriteCursorInBytes=%d.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback); - #endif - availableBytesPlayback = 0; - } - } else { - /* Different loop iterations. The available bytes only goes from the virtual write cursor to the physical play cursor. */ - if (physicalPlayCursorInBytes >= virtualWriteCursorInBytesPlayback) { - availableBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback; - } else { - /* This is an error. */ - #ifdef MA_DEBUG_OUTPUT - printf("[DirectSound] (Playback) WARNING: Write cursor has moved behind the play cursor (different loop iterations). physicalPlayCursorInBytes=%d, virtualWriteCursorInBytes=%d.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback); - #endif - availableBytesPlayback = 0; - } + /* Wait for space to become available first. */ + if (WaitForSingleObject(pDevice->wasapi.hEventPlayback, INFINITE) == WAIT_FAILED) { + exitLoop = MA_TRUE; + break; /* Wait failed. */ } - #ifdef MA_DEBUG_OUTPUT - /*printf("[DirectSound] (Playback) physicalPlayCursorInBytes=%d, availableBytesPlayback=%d\n", physicalPlayCursorInBytes, availableBytesPlayback);*/ - #endif - - /* If there's no room available for writing we need to wait for more. */ - if (availableBytesPlayback < (pDevice->playback.internalBufferSizeInFrames/pDevice->playback.internalPeriods)) { - /* If we haven't started the device yet, this will never get beyond 0. In this case we need to get the device started. */ - if (availableBytesPlayback == 0 && !isPlaybackDeviceStarted) { - if (FAILED(ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING))) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.", MA_FAILED_TO_START_BACKEND_DEVICE); - } - isPlaybackDeviceStarted = MA_TRUE; - } else { - ma_sleep(waitTimeInMilliseconds); - continue; - } + /* Check how much space is available. If this returns 0 we just keep waiting. */ + result = ma_device__get_available_frames__wasapi(pDevice, (ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &framesAvailablePlayback); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; } - /* Getting here means there room available somewhere. We limit this to either the end of the buffer or the physical play cursor, whichever is closest. */ - lockOffsetInBytesPlayback = virtualWriteCursorInBytesPlayback; - if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) { - /* Same loop iteration. Go up to the end of the buffer. */ - lockSizeInBytesPlayback = (pDevice->playback.internalBufferSizeInFrames*bpfPlayback) - virtualWriteCursorInBytesPlayback; - } else { - /* Different loop iterations. Go up to the physical play cursor. */ - lockSizeInBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback; + if (framesAvailablePlayback < pDevice->wasapi.periodSizeInFramesPlayback) { + continue; /* No space available. */ } - hr = ma_IDirectSoundBuffer_Lock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, lockOffsetInBytesPlayback, lockSizeInBytesPlayback, &pMappedBufferPlayback, &mappedSizeInBytesPlayback, NULL, NULL, 0); + /* Map a the data buffer in preparation for the callback. */ + hr = ma_IAudioRenderClient_GetBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, framesAvailablePlayback, &pMappedDeviceBufferPlayback); if (FAILED(hr)) { - result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from playback device in preparation for writing to the device.", MA_FAILED_TO_MAP_DEVICE_BUFFER); + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from playback device in preparation for writing to the device.", MA_FAILED_TO_MAP_DEVICE_BUFFER); + exitLoop = MA_TRUE; break; } - /* At this point we have a buffer for output. */ - ma_device__read_frames_from_client(pDevice, (mappedSizeInBytesPlayback/bpfPlayback), pMappedBufferPlayback); + /* We should have a buffer at this point. */ + ma_device__read_frames_from_client(pDevice, framesAvailablePlayback, pMappedDeviceBufferPlayback); + + /* At this point we're done writing to the device and we just need to release the buffer. */ + hr = ma_IAudioRenderClient_ReleaseBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, framesAvailablePlayback, 0); + pMappedDeviceBufferPlayback = NULL; /* <-- Important. Not doing this can result in an error once we leave this loop because it will use this to know whether or not a final ReleaseBuffer() needs to be called. */ + mappedDeviceBufferSizeInFramesPlayback = 0; - hr = ma_IDirectSoundBuffer_Unlock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, pMappedBufferPlayback, mappedSizeInBytesPlayback, NULL, 0); if (FAILED(hr)) { - result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from playback device after writing to the device.", MA_FAILED_TO_UNMAP_DEVICE_BUFFER); + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to release internal buffer from playback device after writing to the device.", MA_FAILED_TO_UNMAP_DEVICE_BUFFER); + exitLoop = MA_TRUE; break; } - virtualWriteCursorInBytesPlayback += mappedSizeInBytesPlayback; - if (virtualWriteCursorInBytesPlayback == pDevice->playback.internalBufferSizeInFrames*bpfPlayback) { - virtualWriteCursorInBytesPlayback = 0; - virtualWriteCursorLoopFlagPlayback = !virtualWriteCursorLoopFlagPlayback; - } - - /* - We may need to start the device. We want two full periods to be written before starting the playback device. Having an extra period adds - a bit of a buffer to prevent the playback buffer from getting starved. - */ - framesWrittenToPlaybackDevice += mappedSizeInBytesPlayback/bpfPlayback; - if (!isPlaybackDeviceStarted && framesWrittenToPlaybackDevice >= (pDevice->playback.internalBufferSizeInFrames/pDevice->playback.internalPeriods)) { - if (FAILED(ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING))) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.", MA_FAILED_TO_START_BACKEND_DEVICE); + framesWrittenToPlaybackDevice += framesAvailablePlayback; + if (!pDevice->wasapi.isStartedPlayback) { + if (pDevice->playback.shareMode == ma_share_mode_exclusive || framesWrittenToPlaybackDevice >= pDevice->playback.internalPeriodSizeInFrames*1) { + hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); + if (FAILED(hr)) { + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal playback device.", MA_FAILED_TO_START_BACKEND_DEVICE); + exitLoop = MA_TRUE; + break; + } + ma_atomic_exchange_32(&pDevice->wasapi.isStartedPlayback, MA_TRUE); } - isPlaybackDeviceStarted = MA_TRUE; } } break; + default: return MA_INVALID_ARGS; + } + } - default: return MA_INVALID_ARGS; /* Invalid device type. */ + /* Here is where the device needs to be stopped. */ + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) { + /* Any mapped buffers need to be released. */ + if (pMappedDeviceBufferCapture != NULL) { + hr = ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, mappedDeviceBufferSizeInFramesCapture); } - if (result != MA_SUCCESS) { - return result; + hr = ma_IAudioClient_Stop((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); + if (FAILED(hr)) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to stop internal capture device.", MA_FAILED_TO_STOP_BACKEND_DEVICE); } - } - /* Getting here means the device is being stopped. */ - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - if (FAILED(ma_IDirectSoundCaptureBuffer_Stop((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer))) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCaptureBuffer_Stop() failed.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + /* The audio client needs to be reset otherwise restarting will fail. */ + hr = ma_IAudioClient_Reset((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture); + if (FAILED(hr)) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to reset internal capture device.", MA_FAILED_TO_STOP_BACKEND_DEVICE); } + + ma_atomic_exchange_32(&pDevice->wasapi.isStartedCapture, MA_FALSE); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - /* The playback device should be drained before stopping. All we do is wait until the available bytes is equal to the size of the buffer. */ - if (isPlaybackDeviceStarted) { - for (;;) { - DWORD availableBytesPlayback = 0; - DWORD physicalPlayCursorInBytes; - DWORD physicalWriteCursorInBytes; - if (FAILED(ma_IDirectSoundBuffer_GetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &physicalPlayCursorInBytes, &physicalWriteCursorInBytes))) { - break; - } + /* Any mapped buffers need to be released. */ + if (pMappedDeviceBufferPlayback != NULL) { + hr = ma_IAudioRenderClient_ReleaseBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, mappedDeviceBufferSizeInFramesPlayback, 0); + } - if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) { - physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback; - } - prevPlayCursorInBytesPlayback = physicalPlayCursorInBytes; + /* + The buffer needs to be drained before stopping the device. Not doing this will result in the last few frames not getting output to + the speakers. This is a problem for very short sounds because it'll result in a significant portion of it not getting played. + */ + if (pDevice->wasapi.isStartedPlayback) { + if (pDevice->playback.shareMode == ma_share_mode_exclusive) { + WaitForSingleObject(pDevice->wasapi.hEventPlayback, INFINITE); + } else { + ma_uint32 prevFramesAvaialablePlayback = (ma_uint32)-1; + ma_uint32 framesAvailablePlayback; + for (;;) { + result = ma_device__get_available_frames__wasapi(pDevice, (ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &framesAvailablePlayback); + if (result != MA_SUCCESS) { + break; + } - if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) { - /* Same loop iteration. The available bytes wraps all the way around from the virtual write cursor to the physical play cursor. */ - if (physicalPlayCursorInBytes <= virtualWriteCursorInBytesPlayback) { - availableBytesPlayback = (pDevice->playback.internalBufferSizeInFrames*bpfPlayback) - virtualWriteCursorInBytesPlayback; - availableBytesPlayback += physicalPlayCursorInBytes; /* Wrap around. */ - } else { + if (framesAvailablePlayback >= pDevice->wasapi.actualPeriodSizeInFramesPlayback) { break; } - } else { - /* Different loop iterations. The available bytes only goes from the virtual write cursor to the physical play cursor. */ - if (physicalPlayCursorInBytes >= virtualWriteCursorInBytesPlayback) { - availableBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback; - } else { + + /* + Just a safety check to avoid an infinite loop. If this iteration results in a situation where the number of available frames + has not changed, get out of the loop. I don't think this should ever happen, but I think it's nice to have just in case. + */ + if (framesAvailablePlayback == prevFramesAvaialablePlayback) { break; } - } + prevFramesAvaialablePlayback = framesAvailablePlayback; - if (availableBytesPlayback >= (pDevice->playback.internalBufferSizeInFrames*bpfPlayback)) { - break; + WaitForSingleObject(pDevice->wasapi.hEventPlayback, INFINITE); + ResetEvent(pDevice->wasapi.hEventPlayback); /* Manual reset. */ } - - ma_sleep(waitTimeInMilliseconds); } } - if (FAILED(ma_IDirectSoundBuffer_Stop((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer))) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Stop() failed.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + hr = ma_IAudioClient_Stop((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); + if (FAILED(hr)) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to stop internal playback device.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + } + + /* The audio client needs to be reset otherwise restarting will fail. */ + hr = ma_IAudioClient_Reset((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback); + if (FAILED(hr)) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to reset internal playback device.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + } + + ma_atomic_exchange_32(&pDevice->wasapi.isStartedPlayback, MA_FALSE); + } + + return MA_SUCCESS; +} + +static ma_result ma_context_uninit__wasapi(ma_context* pContext) +{ + MA_ASSERT(pContext != NULL); + MA_ASSERT(pContext->backend == ma_backend_wasapi); + (void)pContext; + + return MA_SUCCESS; +} + +static ma_result ma_context_init__wasapi(const ma_context_config* pConfig, ma_context* pContext) +{ + ma_result result = MA_SUCCESS; + + MA_ASSERT(pContext != NULL); + + (void)pConfig; + +#ifdef MA_WIN32_DESKTOP + /* + WASAPI is only supported in Vista SP1 and newer. The reason for SP1 and not the base version of Vista is that event-driven + exclusive mode does not work until SP1. + + Unfortunately older compilers don't define these functions so we need to dynamically load them in order to avoid a lin error. + */ + { + ma_OSVERSIONINFOEXW osvi; + ma_handle kernel32DLL; + ma_PFNVerifyVersionInfoW _VerifyVersionInfoW; + ma_PFNVerSetConditionMask _VerSetConditionMask; + + kernel32DLL = ma_dlopen(pContext, "kernel32.dll"); + if (kernel32DLL == NULL) { + return MA_NO_BACKEND; + } + + _VerifyVersionInfoW = (ma_PFNVerifyVersionInfoW)ma_dlsym(pContext, kernel32DLL, "VerifyVersionInfoW"); + _VerSetConditionMask = (ma_PFNVerSetConditionMask)ma_dlsym(pContext, kernel32DLL, "VerSetConditionMask"); + if (_VerifyVersionInfoW == NULL || _VerSetConditionMask == NULL) { + ma_dlclose(pContext, kernel32DLL); + return MA_NO_BACKEND; + } + + MA_ZERO_OBJECT(&osvi); + osvi.dwOSVersionInfoSize = sizeof(osvi); + osvi.dwMajorVersion = HIBYTE(MA_WIN32_WINNT_VISTA); + osvi.dwMinorVersion = LOBYTE(MA_WIN32_WINNT_VISTA); + osvi.wServicePackMajor = 1; + if (_VerifyVersionInfoW(&osvi, MA_VER_MAJORVERSION | MA_VER_MINORVERSION | MA_VER_SERVICEPACKMAJOR, _VerSetConditionMask(_VerSetConditionMask(_VerSetConditionMask(0, MA_VER_MAJORVERSION, MA_VER_GREATER_EQUAL), MA_VER_MINORVERSION, MA_VER_GREATER_EQUAL), MA_VER_SERVICEPACKMAJOR, MA_VER_GREATER_EQUAL))) { + result = MA_SUCCESS; + } else { + result = MA_NO_BACKEND; } - ma_IDirectSoundBuffer_SetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0); + ma_dlclose(pContext, kernel32DLL); } +#endif - return MA_SUCCESS; -} - -ma_result ma_context_uninit__dsound(ma_context* pContext) -{ - ma_assert(pContext != NULL); - ma_assert(pContext->backend == ma_backend_dsound); + if (result != MA_SUCCESS) { + return result; + } - ma_dlclose(pContext, pContext->dsound.hDSoundDLL); + pContext->onUninit = ma_context_uninit__wasapi; + pContext->onDeviceIDEqual = ma_context_is_device_id_equal__wasapi; + pContext->onEnumDevices = ma_context_enumerate_devices__wasapi; + pContext->onGetDeviceInfo = ma_context_get_device_info__wasapi; + pContext->onDeviceInit = ma_device_init__wasapi; + pContext->onDeviceUninit = ma_device_uninit__wasapi; + pContext->onDeviceStart = NULL; /* Not used. Started in onDeviceMainLoop. */ + pContext->onDeviceStop = ma_device_stop__wasapi; /* Required to ensure the capture event is signalled when stopping a loopback device while nothing is playing. */ + pContext->onDeviceMainLoop = ma_device_main_loop__wasapi; - return MA_SUCCESS; + return result; } +#endif -ma_result ma_context_init__dsound(const ma_context_config* pConfig, ma_context* pContext) -{ - ma_assert(pContext != NULL); +/****************************************************************************** - (void)pConfig; +DirectSound Backend - pContext->dsound.hDSoundDLL = ma_dlopen(pContext, "dsound.dll"); - if (pContext->dsound.hDSoundDLL == NULL) { - return MA_API_NOT_FOUND; - } +******************************************************************************/ +#ifdef MA_HAS_DSOUND +/*#include */ - pContext->dsound.DirectSoundCreate = ma_dlsym(pContext, pContext->dsound.hDSoundDLL, "DirectSoundCreate"); - pContext->dsound.DirectSoundEnumerateA = ma_dlsym(pContext, pContext->dsound.hDSoundDLL, "DirectSoundEnumerateA"); - pContext->dsound.DirectSoundCaptureCreate = ma_dlsym(pContext, pContext->dsound.hDSoundDLL, "DirectSoundCaptureCreate"); - pContext->dsound.DirectSoundCaptureEnumerateA = ma_dlsym(pContext, pContext->dsound.hDSoundDLL, "DirectSoundCaptureEnumerateA"); +static const GUID MA_GUID_IID_DirectSoundNotify = {0xb0210783, 0x89cd, 0x11d0, {0xaf, 0x08, 0x00, 0xa0, 0xc9, 0x25, 0xcd, 0x16}}; - pContext->onUninit = ma_context_uninit__dsound; - pContext->onDeviceIDEqual = ma_context_is_device_id_equal__dsound; - pContext->onEnumDevices = ma_context_enumerate_devices__dsound; - pContext->onGetDeviceInfo = ma_context_get_device_info__dsound; - pContext->onDeviceInit = ma_device_init__dsound; - pContext->onDeviceUninit = ma_device_uninit__dsound; - pContext->onDeviceStart = NULL; /* Not used. Started in onDeviceMainLoop. */ - pContext->onDeviceStop = NULL; /* Not used. Stopped in onDeviceMainLoop. */ - pContext->onDeviceMainLoop = ma_device_main_loop__dsound; +/* miniaudio only uses priority or exclusive modes. */ +#define MA_DSSCL_NORMAL 1 +#define MA_DSSCL_PRIORITY 2 +#define MA_DSSCL_EXCLUSIVE 3 +#define MA_DSSCL_WRITEPRIMARY 4 - return MA_SUCCESS; -} -#endif +#define MA_DSCAPS_PRIMARYMONO 0x00000001 +#define MA_DSCAPS_PRIMARYSTEREO 0x00000002 +#define MA_DSCAPS_PRIMARY8BIT 0x00000004 +#define MA_DSCAPS_PRIMARY16BIT 0x00000008 +#define MA_DSCAPS_CONTINUOUSRATE 0x00000010 +#define MA_DSCAPS_EMULDRIVER 0x00000020 +#define MA_DSCAPS_CERTIFIED 0x00000040 +#define MA_DSCAPS_SECONDARYMONO 0x00000100 +#define MA_DSCAPS_SECONDARYSTEREO 0x00000200 +#define MA_DSCAPS_SECONDARY8BIT 0x00000400 +#define MA_DSCAPS_SECONDARY16BIT 0x00000800 +#define MA_DSBCAPS_PRIMARYBUFFER 0x00000001 +#define MA_DSBCAPS_STATIC 0x00000002 +#define MA_DSBCAPS_LOCHARDWARE 0x00000004 +#define MA_DSBCAPS_LOCSOFTWARE 0x00000008 +#define MA_DSBCAPS_CTRL3D 0x00000010 +#define MA_DSBCAPS_CTRLFREQUENCY 0x00000020 +#define MA_DSBCAPS_CTRLPAN 0x00000040 +#define MA_DSBCAPS_CTRLVOLUME 0x00000080 +#define MA_DSBCAPS_CTRLPOSITIONNOTIFY 0x00000100 +#define MA_DSBCAPS_CTRLFX 0x00000200 +#define MA_DSBCAPS_STICKYFOCUS 0x00004000 +#define MA_DSBCAPS_GLOBALFOCUS 0x00008000 +#define MA_DSBCAPS_GETCURRENTPOSITION2 0x00010000 +#define MA_DSBCAPS_MUTE3DATMAXDISTANCE 0x00020000 +#define MA_DSBCAPS_LOCDEFER 0x00040000 +#define MA_DSBCAPS_TRUEPLAYPOSITION 0x00080000 +#define MA_DSBPLAY_LOOPING 0x00000001 +#define MA_DSBPLAY_LOCHARDWARE 0x00000002 +#define MA_DSBPLAY_LOCSOFTWARE 0x00000004 +#define MA_DSBPLAY_TERMINATEBY_TIME 0x00000008 +#define MA_DSBPLAY_TERMINATEBY_DISTANCE 0x00000010 +#define MA_DSBPLAY_TERMINATEBY_PRIORITY 0x00000020 -/****************************************************************************** +#define MA_DSCBSTART_LOOPING 0x00000001 -WinMM Backend +typedef struct +{ + DWORD dwSize; + DWORD dwFlags; + DWORD dwBufferBytes; + DWORD dwReserved; + WAVEFORMATEX* lpwfxFormat; + GUID guid3DAlgorithm; +} MA_DSBUFFERDESC; -******************************************************************************/ -#ifdef MA_HAS_WINMM +typedef struct +{ + DWORD dwSize; + DWORD dwFlags; + DWORD dwBufferBytes; + DWORD dwReserved; + WAVEFORMATEX* lpwfxFormat; + DWORD dwFXCount; + void* lpDSCFXDesc; /* <-- miniaudio doesn't use this, so set to void*. */ +} MA_DSCBUFFERDESC; -/* -Some older compilers don't have WAVEOUTCAPS2A and WAVEINCAPS2A, so we'll need to write this ourselves. These structures -are exactly the same as the older ones but they have a few GUIDs for manufacturer/product/name identification. I'm keeping -the names the same as the Win32 library for consistency, but namespaced to avoid naming conflicts with the Win32 version. -*/ typedef struct { - WORD wMid; - WORD wPid; - MMVERSION vDriverVersion; - CHAR szPname[MAXPNAMELEN]; - DWORD dwFormats; - WORD wChannels; - WORD wReserved1; - DWORD dwSupport; - GUID ManufacturerGuid; - GUID ProductGuid; - GUID NameGuid; -} MA_WAVEOUTCAPS2A; + DWORD dwSize; + DWORD dwFlags; + DWORD dwMinSecondarySampleRate; + DWORD dwMaxSecondarySampleRate; + DWORD dwPrimaryBuffers; + DWORD dwMaxHwMixingAllBuffers; + DWORD dwMaxHwMixingStaticBuffers; + DWORD dwMaxHwMixingStreamingBuffers; + DWORD dwFreeHwMixingAllBuffers; + DWORD dwFreeHwMixingStaticBuffers; + DWORD dwFreeHwMixingStreamingBuffers; + DWORD dwMaxHw3DAllBuffers; + DWORD dwMaxHw3DStaticBuffers; + DWORD dwMaxHw3DStreamingBuffers; + DWORD dwFreeHw3DAllBuffers; + DWORD dwFreeHw3DStaticBuffers; + DWORD dwFreeHw3DStreamingBuffers; + DWORD dwTotalHwMemBytes; + DWORD dwFreeHwMemBytes; + DWORD dwMaxContigFreeHwMemBytes; + DWORD dwUnlockTransferRateHwBuffers; + DWORD dwPlayCpuOverheadSwBuffers; + DWORD dwReserved1; + DWORD dwReserved2; +} MA_DSCAPS; + typedef struct { - WORD wMid; - WORD wPid; - MMVERSION vDriverVersion; - CHAR szPname[MAXPNAMELEN]; + DWORD dwSize; + DWORD dwFlags; + DWORD dwBufferBytes; + DWORD dwUnlockTransferRate; + DWORD dwPlayCpuOverhead; +} MA_DSBCAPS; + +typedef struct +{ + DWORD dwSize; + DWORD dwFlags; DWORD dwFormats; - WORD wChannels; - WORD wReserved1; - GUID ManufacturerGuid; - GUID ProductGuid; - GUID NameGuid; -} MA_WAVEINCAPS2A; + DWORD dwChannels; +} MA_DSCCAPS; -typedef UINT (WINAPI * MA_PFN_waveOutGetNumDevs)(void); -typedef MMRESULT (WINAPI * MA_PFN_waveOutGetDevCapsA)(ma_uintptr uDeviceID, LPWAVEOUTCAPSA pwoc, UINT cbwoc); -typedef MMRESULT (WINAPI * MA_PFN_waveOutOpen)(LPHWAVEOUT phwo, UINT uDeviceID, LPCWAVEFORMATEX pwfx, DWORD_PTR dwCallback, DWORD_PTR dwInstance, DWORD fdwOpen); -typedef MMRESULT (WINAPI * MA_PFN_waveOutClose)(HWAVEOUT hwo); -typedef MMRESULT (WINAPI * MA_PFN_waveOutPrepareHeader)(HWAVEOUT hwo, LPWAVEHDR pwh, UINT cbwh); -typedef MMRESULT (WINAPI * MA_PFN_waveOutUnprepareHeader)(HWAVEOUT hwo, LPWAVEHDR pwh, UINT cbwh); -typedef MMRESULT (WINAPI * MA_PFN_waveOutWrite)(HWAVEOUT hwo, LPWAVEHDR pwh, UINT cbwh); -typedef MMRESULT (WINAPI * MA_PFN_waveOutReset)(HWAVEOUT hwo); -typedef UINT (WINAPI * MA_PFN_waveInGetNumDevs)(void); -typedef MMRESULT (WINAPI * MA_PFN_waveInGetDevCapsA)(ma_uintptr uDeviceID, LPWAVEINCAPSA pwic, UINT cbwic); -typedef MMRESULT (WINAPI * MA_PFN_waveInOpen)(LPHWAVEIN phwi, UINT uDeviceID, LPCWAVEFORMATEX pwfx, DWORD_PTR dwCallback, DWORD_PTR dwInstance, DWORD fdwOpen); -typedef MMRESULT (WINAPI * MA_PFN_waveInClose)(HWAVEIN hwi); -typedef MMRESULT (WINAPI * MA_PFN_waveInPrepareHeader)(HWAVEIN hwi, LPWAVEHDR pwh, UINT cbwh); -typedef MMRESULT (WINAPI * MA_PFN_waveInUnprepareHeader)(HWAVEIN hwi, LPWAVEHDR pwh, UINT cbwh); -typedef MMRESULT (WINAPI * MA_PFN_waveInAddBuffer)(HWAVEIN hwi, LPWAVEHDR pwh, UINT cbwh); -typedef MMRESULT (WINAPI * MA_PFN_waveInStart)(HWAVEIN hwi); -typedef MMRESULT (WINAPI * MA_PFN_waveInReset)(HWAVEIN hwi); +typedef struct +{ + DWORD dwSize; + DWORD dwFlags; + DWORD dwBufferBytes; + DWORD dwReserved; +} MA_DSCBCAPS; -ma_result ma_result_from_MMRESULT(MMRESULT resultMM) +typedef struct { - switch (resultMM) { - case MMSYSERR_NOERROR: return MA_SUCCESS; - case MMSYSERR_BADDEVICEID: return MA_INVALID_ARGS; - case MMSYSERR_INVALHANDLE: return MA_INVALID_ARGS; - case MMSYSERR_NOMEM: return MA_OUT_OF_MEMORY; - case MMSYSERR_INVALFLAG: return MA_INVALID_ARGS; - case MMSYSERR_INVALPARAM: return MA_INVALID_ARGS; - case MMSYSERR_HANDLEBUSY: return MA_DEVICE_BUSY; - case MMSYSERR_ERROR: return MA_ERROR; - default: return MA_ERROR; - } -} + DWORD dwOffset; + HANDLE hEventNotify; +} MA_DSBPOSITIONNOTIFY; + +typedef struct ma_IDirectSound ma_IDirectSound; +typedef struct ma_IDirectSoundBuffer ma_IDirectSoundBuffer; +typedef struct ma_IDirectSoundCapture ma_IDirectSoundCapture; +typedef struct ma_IDirectSoundCaptureBuffer ma_IDirectSoundCaptureBuffer; +typedef struct ma_IDirectSoundNotify ma_IDirectSoundNotify; + + +/* +COM objects. The way these work is that you have a vtable (a list of function pointers, kind of +like how C++ works internally), and then you have a structure with a single member, which is a +pointer to the vtable. The vtable is where the methods of the object are defined. Methods need +to be in a specific order, and parent classes need to have their methods declared first. +*/ + +/* IDirectSound */ +typedef struct +{ + /* IUnknown */ + HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSound* pThis, const IID* const riid, void** ppObject); + ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSound* pThis); + ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSound* pThis); -char* ma_find_last_character(char* str, char ch) + /* IDirectSound */ + HRESULT (STDMETHODCALLTYPE * CreateSoundBuffer) (ma_IDirectSound* pThis, const MA_DSBUFFERDESC* pDSBufferDesc, ma_IDirectSoundBuffer** ppDSBuffer, void* pUnkOuter); + HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSound* pThis, MA_DSCAPS* pDSCaps); + HRESULT (STDMETHODCALLTYPE * DuplicateSoundBuffer)(ma_IDirectSound* pThis, ma_IDirectSoundBuffer* pDSBufferOriginal, ma_IDirectSoundBuffer** ppDSBufferDuplicate); + HRESULT (STDMETHODCALLTYPE * SetCooperativeLevel) (ma_IDirectSound* pThis, HWND hwnd, DWORD dwLevel); + HRESULT (STDMETHODCALLTYPE * Compact) (ma_IDirectSound* pThis); + HRESULT (STDMETHODCALLTYPE * GetSpeakerConfig) (ma_IDirectSound* pThis, DWORD* pSpeakerConfig); + HRESULT (STDMETHODCALLTYPE * SetSpeakerConfig) (ma_IDirectSound* pThis, DWORD dwSpeakerConfig); + HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSound* pThis, const GUID* pGuidDevice); +} ma_IDirectSoundVtbl; +struct ma_IDirectSound { - char* last; - - if (str == NULL) { - return NULL; - } + ma_IDirectSoundVtbl* lpVtbl; +}; +static MA_INLINE HRESULT ma_IDirectSound_QueryInterface(ma_IDirectSound* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } +static MA_INLINE ULONG ma_IDirectSound_AddRef(ma_IDirectSound* pThis) { return pThis->lpVtbl->AddRef(pThis); } +static MA_INLINE ULONG ma_IDirectSound_Release(ma_IDirectSound* pThis) { return pThis->lpVtbl->Release(pThis); } +static MA_INLINE HRESULT ma_IDirectSound_CreateSoundBuffer(ma_IDirectSound* pThis, const MA_DSBUFFERDESC* pDSBufferDesc, ma_IDirectSoundBuffer** ppDSBuffer, void* pUnkOuter) { return pThis->lpVtbl->CreateSoundBuffer(pThis, pDSBufferDesc, ppDSBuffer, pUnkOuter); } +static MA_INLINE HRESULT ma_IDirectSound_GetCaps(ma_IDirectSound* pThis, MA_DSCAPS* pDSCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSCaps); } +static MA_INLINE HRESULT ma_IDirectSound_DuplicateSoundBuffer(ma_IDirectSound* pThis, ma_IDirectSoundBuffer* pDSBufferOriginal, ma_IDirectSoundBuffer** ppDSBufferDuplicate) { return pThis->lpVtbl->DuplicateSoundBuffer(pThis, pDSBufferOriginal, ppDSBufferDuplicate); } +static MA_INLINE HRESULT ma_IDirectSound_SetCooperativeLevel(ma_IDirectSound* pThis, HWND hwnd, DWORD dwLevel) { return pThis->lpVtbl->SetCooperativeLevel(pThis, hwnd, dwLevel); } +static MA_INLINE HRESULT ma_IDirectSound_Compact(ma_IDirectSound* pThis) { return pThis->lpVtbl->Compact(pThis); } +static MA_INLINE HRESULT ma_IDirectSound_GetSpeakerConfig(ma_IDirectSound* pThis, DWORD* pSpeakerConfig) { return pThis->lpVtbl->GetSpeakerConfig(pThis, pSpeakerConfig); } +static MA_INLINE HRESULT ma_IDirectSound_SetSpeakerConfig(ma_IDirectSound* pThis, DWORD dwSpeakerConfig) { return pThis->lpVtbl->SetSpeakerConfig(pThis, dwSpeakerConfig); } +static MA_INLINE HRESULT ma_IDirectSound_Initialize(ma_IDirectSound* pThis, const GUID* pGuidDevice) { return pThis->lpVtbl->Initialize(pThis, pGuidDevice); } - last = NULL; - while (*str != '\0') { - if (*str == ch) { - last = str; - } - str += 1; - } +/* IDirectSoundBuffer */ +typedef struct +{ + /* IUnknown */ + HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundBuffer* pThis, const IID* const riid, void** ppObject); + ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundBuffer* pThis); + ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundBuffer* pThis); - return last; -} + /* IDirectSoundBuffer */ + HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSoundBuffer* pThis, MA_DSBCAPS* pDSBufferCaps); + HRESULT (STDMETHODCALLTYPE * GetCurrentPosition)(ma_IDirectSoundBuffer* pThis, DWORD* pCurrentPlayCursor, DWORD* pCurrentWriteCursor); + HRESULT (STDMETHODCALLTYPE * GetFormat) (ma_IDirectSoundBuffer* pThis, WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten); + HRESULT (STDMETHODCALLTYPE * GetVolume) (ma_IDirectSoundBuffer* pThis, LONG* pVolume); + HRESULT (STDMETHODCALLTYPE * GetPan) (ma_IDirectSoundBuffer* pThis, LONG* pPan); + HRESULT (STDMETHODCALLTYPE * GetFrequency) (ma_IDirectSoundBuffer* pThis, DWORD* pFrequency); + HRESULT (STDMETHODCALLTYPE * GetStatus) (ma_IDirectSoundBuffer* pThis, DWORD* pStatus); + HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSoundBuffer* pThis, ma_IDirectSound* pDirectSound, const MA_DSBUFFERDESC* pDSBufferDesc); + HRESULT (STDMETHODCALLTYPE * Lock) (ma_IDirectSoundBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags); + HRESULT (STDMETHODCALLTYPE * Play) (ma_IDirectSoundBuffer* pThis, DWORD dwReserved1, DWORD dwPriority, DWORD dwFlags); + HRESULT (STDMETHODCALLTYPE * SetCurrentPosition)(ma_IDirectSoundBuffer* pThis, DWORD dwNewPosition); + HRESULT (STDMETHODCALLTYPE * SetFormat) (ma_IDirectSoundBuffer* pThis, const WAVEFORMATEX* pFormat); + HRESULT (STDMETHODCALLTYPE * SetVolume) (ma_IDirectSoundBuffer* pThis, LONG volume); + HRESULT (STDMETHODCALLTYPE * SetPan) (ma_IDirectSoundBuffer* pThis, LONG pan); + HRESULT (STDMETHODCALLTYPE * SetFrequency) (ma_IDirectSoundBuffer* pThis, DWORD dwFrequency); + HRESULT (STDMETHODCALLTYPE * Stop) (ma_IDirectSoundBuffer* pThis); + HRESULT (STDMETHODCALLTYPE * Unlock) (ma_IDirectSoundBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2); + HRESULT (STDMETHODCALLTYPE * Restore) (ma_IDirectSoundBuffer* pThis); +} ma_IDirectSoundBufferVtbl; +struct ma_IDirectSoundBuffer +{ + ma_IDirectSoundBufferVtbl* lpVtbl; +}; +static MA_INLINE HRESULT ma_IDirectSoundBuffer_QueryInterface(ma_IDirectSoundBuffer* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } +static MA_INLINE ULONG ma_IDirectSoundBuffer_AddRef(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->AddRef(pThis); } +static MA_INLINE ULONG ma_IDirectSoundBuffer_Release(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->Release(pThis); } +static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetCaps(ma_IDirectSoundBuffer* pThis, MA_DSBCAPS* pDSBufferCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSBufferCaps); } +static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetCurrentPosition(ma_IDirectSoundBuffer* pThis, DWORD* pCurrentPlayCursor, DWORD* pCurrentWriteCursor) { return pThis->lpVtbl->GetCurrentPosition(pThis, pCurrentPlayCursor, pCurrentWriteCursor); } +static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetFormat(ma_IDirectSoundBuffer* pThis, WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten) { return pThis->lpVtbl->GetFormat(pThis, pFormat, dwSizeAllocated, pSizeWritten); } +static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetVolume(ma_IDirectSoundBuffer* pThis, LONG* pVolume) { return pThis->lpVtbl->GetVolume(pThis, pVolume); } +static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetPan(ma_IDirectSoundBuffer* pThis, LONG* pPan) { return pThis->lpVtbl->GetPan(pThis, pPan); } +static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetFrequency(ma_IDirectSoundBuffer* pThis, DWORD* pFrequency) { return pThis->lpVtbl->GetFrequency(pThis, pFrequency); } +static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetStatus(ma_IDirectSoundBuffer* pThis, DWORD* pStatus) { return pThis->lpVtbl->GetStatus(pThis, pStatus); } +static MA_INLINE HRESULT ma_IDirectSoundBuffer_Initialize(ma_IDirectSoundBuffer* pThis, ma_IDirectSound* pDirectSound, const MA_DSBUFFERDESC* pDSBufferDesc) { return pThis->lpVtbl->Initialize(pThis, pDirectSound, pDSBufferDesc); } +static MA_INLINE HRESULT ma_IDirectSoundBuffer_Lock(ma_IDirectSoundBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags) { return pThis->lpVtbl->Lock(pThis, dwOffset, dwBytes, ppAudioPtr1, pAudioBytes1, ppAudioPtr2, pAudioBytes2, dwFlags); } +static MA_INLINE HRESULT ma_IDirectSoundBuffer_Play(ma_IDirectSoundBuffer* pThis, DWORD dwReserved1, DWORD dwPriority, DWORD dwFlags) { return pThis->lpVtbl->Play(pThis, dwReserved1, dwPriority, dwFlags); } +static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetCurrentPosition(ma_IDirectSoundBuffer* pThis, DWORD dwNewPosition) { return pThis->lpVtbl->SetCurrentPosition(pThis, dwNewPosition); } +static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetFormat(ma_IDirectSoundBuffer* pThis, const WAVEFORMATEX* pFormat) { return pThis->lpVtbl->SetFormat(pThis, pFormat); } +static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetVolume(ma_IDirectSoundBuffer* pThis, LONG volume) { return pThis->lpVtbl->SetVolume(pThis, volume); } +static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetPan(ma_IDirectSoundBuffer* pThis, LONG pan) { return pThis->lpVtbl->SetPan(pThis, pan); } +static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetFrequency(ma_IDirectSoundBuffer* pThis, DWORD dwFrequency) { return pThis->lpVtbl->SetFrequency(pThis, dwFrequency); } +static MA_INLINE HRESULT ma_IDirectSoundBuffer_Stop(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->Stop(pThis); } +static MA_INLINE HRESULT ma_IDirectSoundBuffer_Unlock(ma_IDirectSoundBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2) { return pThis->lpVtbl->Unlock(pThis, pAudioPtr1, dwAudioBytes1, pAudioPtr2, dwAudioBytes2); } +static MA_INLINE HRESULT ma_IDirectSoundBuffer_Restore(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->Restore(pThis); } -/* -Our own "WAVECAPS" structure that contains generic information shared between WAVEOUTCAPS2 and WAVEINCAPS2 so -we can do things generically and typesafely. Names are being kept the same for consistency. -*/ +/* IDirectSoundCapture */ typedef struct { - CHAR szPname[MAXPNAMELEN]; - DWORD dwFormats; - WORD wChannels; - GUID NameGuid; -} MA_WAVECAPSA; + /* IUnknown */ + HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundCapture* pThis, const IID* const riid, void** ppObject); + ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundCapture* pThis); + ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundCapture* pThis); -ma_result ma_get_best_info_from_formats_flags__winmm(DWORD dwFormats, WORD channels, WORD* pBitsPerSample, DWORD* pSampleRate) + /* IDirectSoundCapture */ + HRESULT (STDMETHODCALLTYPE * CreateCaptureBuffer)(ma_IDirectSoundCapture* pThis, const MA_DSCBUFFERDESC* pDSCBufferDesc, ma_IDirectSoundCaptureBuffer** ppDSCBuffer, void* pUnkOuter); + HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSoundCapture* pThis, MA_DSCCAPS* pDSCCaps); + HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSoundCapture* pThis, const GUID* pGuidDevice); +} ma_IDirectSoundCaptureVtbl; +struct ma_IDirectSoundCapture { - WORD bitsPerSample = 0; - DWORD sampleRate = 0; + ma_IDirectSoundCaptureVtbl* lpVtbl; +}; +static MA_INLINE HRESULT ma_IDirectSoundCapture_QueryInterface(ma_IDirectSoundCapture* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } +static MA_INLINE ULONG ma_IDirectSoundCapture_AddRef(ma_IDirectSoundCapture* pThis) { return pThis->lpVtbl->AddRef(pThis); } +static MA_INLINE ULONG ma_IDirectSoundCapture_Release(ma_IDirectSoundCapture* pThis) { return pThis->lpVtbl->Release(pThis); } +static MA_INLINE HRESULT ma_IDirectSoundCapture_CreateCaptureBuffer(ma_IDirectSoundCapture* pThis, const MA_DSCBUFFERDESC* pDSCBufferDesc, ma_IDirectSoundCaptureBuffer** ppDSCBuffer, void* pUnkOuter) { return pThis->lpVtbl->CreateCaptureBuffer(pThis, pDSCBufferDesc, ppDSCBuffer, pUnkOuter); } +static MA_INLINE HRESULT ma_IDirectSoundCapture_GetCaps (ma_IDirectSoundCapture* pThis, MA_DSCCAPS* pDSCCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSCCaps); } +static MA_INLINE HRESULT ma_IDirectSoundCapture_Initialize (ma_IDirectSoundCapture* pThis, const GUID* pGuidDevice) { return pThis->lpVtbl->Initialize(pThis, pGuidDevice); } - if (pBitsPerSample) { - *pBitsPerSample = 0; - } - if (pSampleRate) { - *pSampleRate = 0; - } - if (channels == 1) { - bitsPerSample = 16; - if ((dwFormats & WAVE_FORMAT_48M16) != 0) { - sampleRate = 48000; - } else if ((dwFormats & WAVE_FORMAT_44M16) != 0) { - sampleRate = 44100; - } else if ((dwFormats & WAVE_FORMAT_2M16) != 0) { - sampleRate = 22050; - } else if ((dwFormats & WAVE_FORMAT_1M16) != 0) { - sampleRate = 11025; - } else if ((dwFormats & WAVE_FORMAT_96M16) != 0) { - sampleRate = 96000; - } else { - bitsPerSample = 8; - if ((dwFormats & WAVE_FORMAT_48M08) != 0) { - sampleRate = 48000; - } else if ((dwFormats & WAVE_FORMAT_44M08) != 0) { - sampleRate = 44100; - } else if ((dwFormats & WAVE_FORMAT_2M08) != 0) { - sampleRate = 22050; - } else if ((dwFormats & WAVE_FORMAT_1M08) != 0) { - sampleRate = 11025; - } else if ((dwFormats & WAVE_FORMAT_96M08) != 0) { - sampleRate = 96000; - } else { - return MA_FORMAT_NOT_SUPPORTED; - } - } - } else { - bitsPerSample = 16; - if ((dwFormats & WAVE_FORMAT_48S16) != 0) { - sampleRate = 48000; - } else if ((dwFormats & WAVE_FORMAT_44S16) != 0) { - sampleRate = 44100; - } else if ((dwFormats & WAVE_FORMAT_2S16) != 0) { - sampleRate = 22050; - } else if ((dwFormats & WAVE_FORMAT_1S16) != 0) { - sampleRate = 11025; - } else if ((dwFormats & WAVE_FORMAT_96S16) != 0) { - sampleRate = 96000; - } else { - bitsPerSample = 8; - if ((dwFormats & WAVE_FORMAT_48S08) != 0) { - sampleRate = 48000; - } else if ((dwFormats & WAVE_FORMAT_44S08) != 0) { - sampleRate = 44100; - } else if ((dwFormats & WAVE_FORMAT_2S08) != 0) { - sampleRate = 22050; - } else if ((dwFormats & WAVE_FORMAT_1S08) != 0) { - sampleRate = 11025; - } else if ((dwFormats & WAVE_FORMAT_96S08) != 0) { - sampleRate = 96000; - } else { - return MA_FORMAT_NOT_SUPPORTED; - } - } - } +/* IDirectSoundCaptureBuffer */ +typedef struct +{ + /* IUnknown */ + HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundCaptureBuffer* pThis, const IID* const riid, void** ppObject); + ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundCaptureBuffer* pThis); + ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundCaptureBuffer* pThis); - if (pBitsPerSample) { - *pBitsPerSample = bitsPerSample; - } - if (pSampleRate) { - *pSampleRate = sampleRate; - } + /* IDirectSoundCaptureBuffer */ + HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSoundCaptureBuffer* pThis, MA_DSCBCAPS* pDSCBCaps); + HRESULT (STDMETHODCALLTYPE * GetCurrentPosition)(ma_IDirectSoundCaptureBuffer* pThis, DWORD* pCapturePosition, DWORD* pReadPosition); + HRESULT (STDMETHODCALLTYPE * GetFormat) (ma_IDirectSoundCaptureBuffer* pThis, WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten); + HRESULT (STDMETHODCALLTYPE * GetStatus) (ma_IDirectSoundCaptureBuffer* pThis, DWORD* pStatus); + HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSoundCaptureBuffer* pThis, ma_IDirectSoundCapture* pDirectSoundCapture, const MA_DSCBUFFERDESC* pDSCBufferDesc); + HRESULT (STDMETHODCALLTYPE * Lock) (ma_IDirectSoundCaptureBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags); + HRESULT (STDMETHODCALLTYPE * Start) (ma_IDirectSoundCaptureBuffer* pThis, DWORD dwFlags); + HRESULT (STDMETHODCALLTYPE * Stop) (ma_IDirectSoundCaptureBuffer* pThis); + HRESULT (STDMETHODCALLTYPE * Unlock) (ma_IDirectSoundCaptureBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2); +} ma_IDirectSoundCaptureBufferVtbl; +struct ma_IDirectSoundCaptureBuffer +{ + ma_IDirectSoundCaptureBufferVtbl* lpVtbl; +}; +static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_QueryInterface(ma_IDirectSoundCaptureBuffer* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } +static MA_INLINE ULONG ma_IDirectSoundCaptureBuffer_AddRef(ma_IDirectSoundCaptureBuffer* pThis) { return pThis->lpVtbl->AddRef(pThis); } +static MA_INLINE ULONG ma_IDirectSoundCaptureBuffer_Release(ma_IDirectSoundCaptureBuffer* pThis) { return pThis->lpVtbl->Release(pThis); } +static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_GetCaps(ma_IDirectSoundCaptureBuffer* pThis, MA_DSCBCAPS* pDSCBCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSCBCaps); } +static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_GetCurrentPosition(ma_IDirectSoundCaptureBuffer* pThis, DWORD* pCapturePosition, DWORD* pReadPosition) { return pThis->lpVtbl->GetCurrentPosition(pThis, pCapturePosition, pReadPosition); } +static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_GetFormat(ma_IDirectSoundCaptureBuffer* pThis, WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten) { return pThis->lpVtbl->GetFormat(pThis, pFormat, dwSizeAllocated, pSizeWritten); } +static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_GetStatus(ma_IDirectSoundCaptureBuffer* pThis, DWORD* pStatus) { return pThis->lpVtbl->GetStatus(pThis, pStatus); } +static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Initialize(ma_IDirectSoundCaptureBuffer* pThis, ma_IDirectSoundCapture* pDirectSoundCapture, const MA_DSCBUFFERDESC* pDSCBufferDesc) { return pThis->lpVtbl->Initialize(pThis, pDirectSoundCapture, pDSCBufferDesc); } +static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Lock(ma_IDirectSoundCaptureBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags) { return pThis->lpVtbl->Lock(pThis, dwOffset, dwBytes, ppAudioPtr1, pAudioBytes1, ppAudioPtr2, pAudioBytes2, dwFlags); } +static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Start(ma_IDirectSoundCaptureBuffer* pThis, DWORD dwFlags) { return pThis->lpVtbl->Start(pThis, dwFlags); } +static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Stop(ma_IDirectSoundCaptureBuffer* pThis) { return pThis->lpVtbl->Stop(pThis); } +static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Unlock(ma_IDirectSoundCaptureBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2) { return pThis->lpVtbl->Unlock(pThis, pAudioPtr1, dwAudioBytes1, pAudioPtr2, dwAudioBytes2); } - return MA_SUCCESS; -} -ma_result ma_formats_flags_to_WAVEFORMATEX__winmm(DWORD dwFormats, WORD channels, WAVEFORMATEX* pWF) +/* IDirectSoundNotify */ +typedef struct { - ma_assert(pWF != NULL); + /* IUnknown */ + HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundNotify* pThis, const IID* const riid, void** ppObject); + ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundNotify* pThis); + ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundNotify* pThis); - ma_zero_object(pWF); - pWF->cbSize = sizeof(*pWF); - pWF->wFormatTag = WAVE_FORMAT_PCM; - pWF->nChannels = (WORD)channels; - if (pWF->nChannels > 2) { - pWF->nChannels = 2; - } + /* IDirectSoundNotify */ + HRESULT (STDMETHODCALLTYPE * SetNotificationPositions)(ma_IDirectSoundNotify* pThis, DWORD dwPositionNotifies, const MA_DSBPOSITIONNOTIFY* pPositionNotifies); +} ma_IDirectSoundNotifyVtbl; +struct ma_IDirectSoundNotify +{ + ma_IDirectSoundNotifyVtbl* lpVtbl; +}; +static MA_INLINE HRESULT ma_IDirectSoundNotify_QueryInterface(ma_IDirectSoundNotify* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); } +static MA_INLINE ULONG ma_IDirectSoundNotify_AddRef(ma_IDirectSoundNotify* pThis) { return pThis->lpVtbl->AddRef(pThis); } +static MA_INLINE ULONG ma_IDirectSoundNotify_Release(ma_IDirectSoundNotify* pThis) { return pThis->lpVtbl->Release(pThis); } +static MA_INLINE HRESULT ma_IDirectSoundNotify_SetNotificationPositions(ma_IDirectSoundNotify* pThis, DWORD dwPositionNotifies, const MA_DSBPOSITIONNOTIFY* pPositionNotifies) { return pThis->lpVtbl->SetNotificationPositions(pThis, dwPositionNotifies, pPositionNotifies); } - if (channels == 1) { - pWF->wBitsPerSample = 16; - if ((dwFormats & WAVE_FORMAT_48M16) != 0) { - pWF->nSamplesPerSec = 48000; - } else if ((dwFormats & WAVE_FORMAT_44M16) != 0) { - pWF->nSamplesPerSec = 44100; - } else if ((dwFormats & WAVE_FORMAT_2M16) != 0) { - pWF->nSamplesPerSec = 22050; - } else if ((dwFormats & WAVE_FORMAT_1M16) != 0) { - pWF->nSamplesPerSec = 11025; - } else if ((dwFormats & WAVE_FORMAT_96M16) != 0) { - pWF->nSamplesPerSec = 96000; - } else { - pWF->wBitsPerSample = 8; - if ((dwFormats & WAVE_FORMAT_48M08) != 0) { - pWF->nSamplesPerSec = 48000; - } else if ((dwFormats & WAVE_FORMAT_44M08) != 0) { - pWF->nSamplesPerSec = 44100; - } else if ((dwFormats & WAVE_FORMAT_2M08) != 0) { - pWF->nSamplesPerSec = 22050; - } else if ((dwFormats & WAVE_FORMAT_1M08) != 0) { - pWF->nSamplesPerSec = 11025; - } else if ((dwFormats & WAVE_FORMAT_96M08) != 0) { - pWF->nSamplesPerSec = 96000; - } else { - return MA_FORMAT_NOT_SUPPORTED; - } - } - } else { - pWF->wBitsPerSample = 16; - if ((dwFormats & WAVE_FORMAT_48S16) != 0) { - pWF->nSamplesPerSec = 48000; - } else if ((dwFormats & WAVE_FORMAT_44S16) != 0) { - pWF->nSamplesPerSec = 44100; - } else if ((dwFormats & WAVE_FORMAT_2S16) != 0) { - pWF->nSamplesPerSec = 22050; - } else if ((dwFormats & WAVE_FORMAT_1S16) != 0) { - pWF->nSamplesPerSec = 11025; - } else if ((dwFormats & WAVE_FORMAT_96S16) != 0) { - pWF->nSamplesPerSec = 96000; - } else { - pWF->wBitsPerSample = 8; - if ((dwFormats & WAVE_FORMAT_48S08) != 0) { - pWF->nSamplesPerSec = 48000; - } else if ((dwFormats & WAVE_FORMAT_44S08) != 0) { - pWF->nSamplesPerSec = 44100; - } else if ((dwFormats & WAVE_FORMAT_2S08) != 0) { - pWF->nSamplesPerSec = 22050; - } else if ((dwFormats & WAVE_FORMAT_1S08) != 0) { - pWF->nSamplesPerSec = 11025; - } else if ((dwFormats & WAVE_FORMAT_96S08) != 0) { - pWF->nSamplesPerSec = 96000; - } else { - return MA_FORMAT_NOT_SUPPORTED; + +typedef BOOL (CALLBACK * ma_DSEnumCallbackAProc) (LPGUID pDeviceGUID, LPCSTR pDeviceDescription, LPCSTR pModule, LPVOID pContext); +typedef HRESULT (WINAPI * ma_DirectSoundCreateProc) (const GUID* pcGuidDevice, ma_IDirectSound** ppDS8, LPUNKNOWN pUnkOuter); +typedef HRESULT (WINAPI * ma_DirectSoundEnumerateAProc) (ma_DSEnumCallbackAProc pDSEnumCallback, LPVOID pContext); +typedef HRESULT (WINAPI * ma_DirectSoundCaptureCreateProc) (const GUID* pcGuidDevice, ma_IDirectSoundCapture** ppDSC8, LPUNKNOWN pUnkOuter); +typedef HRESULT (WINAPI * ma_DirectSoundCaptureEnumerateAProc)(ma_DSEnumCallbackAProc pDSEnumCallback, LPVOID pContext); + +static ma_uint32 ma_get_best_sample_rate_within_range(ma_uint32 sampleRateMin, ma_uint32 sampleRateMax) +{ + /* Normalize the range in case we were given something stupid. */ + if (sampleRateMin < MA_MIN_SAMPLE_RATE) { + sampleRateMin = MA_MIN_SAMPLE_RATE; + } + if (sampleRateMax > MA_MAX_SAMPLE_RATE) { + sampleRateMax = MA_MAX_SAMPLE_RATE; + } + if (sampleRateMin > sampleRateMax) { + sampleRateMin = sampleRateMax; + } + + if (sampleRateMin == sampleRateMax) { + return sampleRateMax; + } else { + size_t iStandardRate; + for (iStandardRate = 0; iStandardRate < ma_countof(g_maStandardSampleRatePriorities); ++iStandardRate) { + ma_uint32 standardRate = g_maStandardSampleRatePriorities[iStandardRate]; + if (standardRate >= sampleRateMin && standardRate <= sampleRateMax) { + return standardRate; } } } - pWF->nBlockAlign = (pWF->nChannels * pWF->wBitsPerSample) / 8; - pWF->nAvgBytesPerSec = pWF->nBlockAlign * pWF->nSamplesPerSec; - - return MA_SUCCESS; + /* Should never get here. */ + MA_ASSERT(MA_FALSE); + return 0; } -ma_result ma_context_get_device_info_from_WAVECAPS(ma_context* pContext, MA_WAVECAPSA* pCaps, ma_device_info* pDeviceInfo) +/* +Retrieves the channel count and channel map for the given speaker configuration. If the speaker configuration is unknown, +the channel count and channel map will be left unmodified. +*/ +static void ma_get_channels_from_speaker_config__dsound(DWORD speakerConfig, WORD* pChannelsOut, DWORD* pChannelMapOut) { - WORD bitsPerSample; - DWORD sampleRate; - ma_result result; - - ma_assert(pContext != NULL); - ma_assert(pCaps != NULL); - ma_assert(pDeviceInfo != NULL); + WORD channels; + DWORD channelMap; - /* - Name / Description - - Unfortunately the name specified in WAVE(OUT/IN)CAPS2 is limited to 31 characters. This results in an unprofessional looking - situation where the names of the devices are truncated. To help work around this, we need to look at the name GUID and try - looking in the registry for the full name. If we can't find it there, we need to just fall back to the default name. - */ + channels = 0; + if (pChannelsOut != NULL) { + channels = *pChannelsOut; + } - /* Set the default to begin with. */ - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), pCaps->szPname, (size_t)-1); + channelMap = 0; + if (pChannelMapOut != NULL) { + channelMap = *pChannelMapOut; + } /* - Now try the registry. There's a few things to consider here: - - The name GUID can be null, in which we case we just need to stick to the original 31 characters. - - If the name GUID is not present in the registry we'll also need to stick to the original 31 characters. - - I like consistency, so I want the returned device names to be consistent with those returned by WASAPI and DirectSound. The - problem, however is that WASAPI and DirectSound use " ()" format (such as "Speakers (High Definition Audio)"), - but WinMM does not specificy the component name. From my admittedly limited testing, I've notice the component name seems to - usually fit within the 31 characters of the fixed sized buffer, so what I'm going to do is parse that string for the component - name, and then concatenate the name from the registry. + The speaker configuration is a combination of speaker config and speaker geometry. The lower 8 bits is what we care about. The upper + 16 bits is for the geometry. */ - if (!ma_is_guid_equal(&pCaps->NameGuid, &MA_GUID_NULL)) { - wchar_t guidStrW[256]; - if (((MA_PFN_StringFromGUID2)pContext->win32.StringFromGUID2)(&pCaps->NameGuid, guidStrW, ma_countof(guidStrW)) > 0) { - char guidStr[256]; - char keyStr[1024]; - HKEY hKey; - - WideCharToMultiByte(CP_UTF8, 0, guidStrW, -1, guidStr, sizeof(guidStr), 0, FALSE); + switch ((BYTE)(speakerConfig)) { + case 1 /*DSSPEAKER_HEADPHONE*/: channels = 2; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT; break; + case 2 /*DSSPEAKER_MONO*/: channels = 1; channelMap = SPEAKER_FRONT_CENTER; break; + case 3 /*DSSPEAKER_QUAD*/: channels = 4; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT; break; + case 4 /*DSSPEAKER_STEREO*/: channels = 2; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT; break; + case 5 /*DSSPEAKER_SURROUND*/: channels = 4; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_BACK_CENTER; break; + case 6 /*DSSPEAKER_5POINT1_BACK*/ /*DSSPEAKER_5POINT1*/: channels = 6; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT; break; + case 7 /*DSSPEAKER_7POINT1_WIDE*/ /*DSSPEAKER_7POINT1*/: channels = 8; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_FRONT_LEFT_OF_CENTER | SPEAKER_FRONT_RIGHT_OF_CENTER; break; + case 8 /*DSSPEAKER_7POINT1_SURROUND*/: channels = 8; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT; break; + case 9 /*DSSPEAKER_5POINT1_SURROUND*/: channels = 6; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT; break; + default: break; + } - ma_strcpy_s(keyStr, sizeof(keyStr), "SYSTEM\\CurrentControlSet\\Control\\MediaCategories\\"); - ma_strcat_s(keyStr, sizeof(keyStr), guidStr); + if (pChannelsOut != NULL) { + *pChannelsOut = channels; + } - if (((MA_PFN_RegOpenKeyExA)pContext->win32.RegOpenKeyExA)(HKEY_LOCAL_MACHINE, keyStr, 0, KEY_READ, &hKey) == ERROR_SUCCESS) { - BYTE nameFromReg[512]; - DWORD nameFromRegSize = sizeof(nameFromReg); - result = ((MA_PFN_RegQueryValueExA)pContext->win32.RegQueryValueExA)(hKey, "Name", 0, NULL, (LPBYTE)nameFromReg, (LPDWORD)&nameFromRegSize); - ((MA_PFN_RegCloseKey)pContext->win32.RegCloseKey)(hKey); + if (pChannelMapOut != NULL) { + *pChannelMapOut = channelMap; + } +} - if (result == ERROR_SUCCESS) { - /* We have the value from the registry, so now we need to construct the name string. */ - char name[1024]; - if (ma_strcpy_s(name, sizeof(name), pDeviceInfo->name) == 0) { - char* nameBeg = ma_find_last_character(name, '('); - if (nameBeg != NULL) { - size_t leadingLen = (nameBeg - name); - ma_strncpy_s(nameBeg + 1, sizeof(name) - leadingLen, (const char*)nameFromReg, (size_t)-1); - /* The closing ")", if it can fit. */ - if (leadingLen + nameFromRegSize < sizeof(name)-1) { - ma_strcat_s(name, sizeof(name), ")"); - } +static ma_result ma_context_create_IDirectSound__dsound(ma_context* pContext, ma_share_mode shareMode, const ma_device_id* pDeviceID, ma_IDirectSound** ppDirectSound) +{ + ma_IDirectSound* pDirectSound; + HWND hWnd; - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), name, (size_t)-1); - } - } - } - } - } - } + MA_ASSERT(pContext != NULL); + MA_ASSERT(ppDirectSound != NULL); + *ppDirectSound = NULL; + pDirectSound = NULL; - result = ma_get_best_info_from_formats_flags__winmm(pCaps->dwFormats, pCaps->wChannels, &bitsPerSample, &sampleRate); - if (result != MA_SUCCESS) { - return result; + if (FAILED(((ma_DirectSoundCreateProc)pContext->dsound.DirectSoundCreate)((pDeviceID == NULL) ? NULL : (const GUID*)pDeviceID->dsound, &pDirectSound, NULL))) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[DirectSound] DirectSoundCreate() failed for playback device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); } - pDeviceInfo->minChannels = pCaps->wChannels; - pDeviceInfo->maxChannels = pCaps->wChannels; - pDeviceInfo->minSampleRate = sampleRate; - pDeviceInfo->maxSampleRate = sampleRate; - pDeviceInfo->formatCount = 1; - if (bitsPerSample == 8) { - pDeviceInfo->formats[0] = ma_format_u8; - } else if (bitsPerSample == 16) { - pDeviceInfo->formats[0] = ma_format_s16; - } else if (bitsPerSample == 24) { - pDeviceInfo->formats[0] = ma_format_s24; - } else if (bitsPerSample == 32) { - pDeviceInfo->formats[0] = ma_format_s32; - } else { - return MA_FORMAT_NOT_SUPPORTED; + /* The cooperative level must be set before doing anything else. */ + hWnd = ((MA_PFN_GetForegroundWindow)pContext->win32.GetForegroundWindow)(); + if (hWnd == NULL) { + hWnd = ((MA_PFN_GetDesktopWindow)pContext->win32.GetDesktopWindow)(); + } + if (FAILED(ma_IDirectSound_SetCooperativeLevel(pDirectSound, hWnd, (shareMode == ma_share_mode_exclusive) ? MA_DSSCL_EXCLUSIVE : MA_DSSCL_PRIORITY))) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_SetCooperateiveLevel() failed for playback device.", MA_SHARE_MODE_NOT_SUPPORTED); } + *ppDirectSound = pDirectSound; return MA_SUCCESS; } -ma_result ma_context_get_device_info_from_WAVEOUTCAPS2(ma_context* pContext, MA_WAVEOUTCAPS2A* pCaps, ma_device_info* pDeviceInfo) -{ - MA_WAVECAPSA caps; - - ma_assert(pContext != NULL); - ma_assert(pCaps != NULL); - ma_assert(pDeviceInfo != NULL); - - ma_copy_memory(caps.szPname, pCaps->szPname, sizeof(caps.szPname)); - caps.dwFormats = pCaps->dwFormats; - caps.wChannels = pCaps->wChannels; - caps.NameGuid = pCaps->NameGuid; - return ma_context_get_device_info_from_WAVECAPS(pContext, &caps, pDeviceInfo); -} - -ma_result ma_context_get_device_info_from_WAVEINCAPS2(ma_context* pContext, MA_WAVEINCAPS2A* pCaps, ma_device_info* pDeviceInfo) +static ma_result ma_context_create_IDirectSoundCapture__dsound(ma_context* pContext, ma_share_mode shareMode, const ma_device_id* pDeviceID, ma_IDirectSoundCapture** ppDirectSoundCapture) { - MA_WAVECAPSA caps; + ma_IDirectSoundCapture* pDirectSoundCapture; - ma_assert(pContext != NULL); - ma_assert(pCaps != NULL); - ma_assert(pDeviceInfo != NULL); + MA_ASSERT(pContext != NULL); + MA_ASSERT(ppDirectSoundCapture != NULL); - ma_copy_memory(caps.szPname, pCaps->szPname, sizeof(caps.szPname)); - caps.dwFormats = pCaps->dwFormats; - caps.wChannels = pCaps->wChannels; - caps.NameGuid = pCaps->NameGuid; - return ma_context_get_device_info_from_WAVECAPS(pContext, &caps, pDeviceInfo); -} + /* DirectSound does not support exclusive mode for capture. */ + if (shareMode == ma_share_mode_exclusive) { + return MA_SHARE_MODE_NOT_SUPPORTED; + } + *ppDirectSoundCapture = NULL; + pDirectSoundCapture = NULL; -ma_bool32 ma_context_is_device_id_equal__winmm(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) -{ - ma_assert(pContext != NULL); - ma_assert(pID0 != NULL); - ma_assert(pID1 != NULL); - (void)pContext; + if (FAILED(((ma_DirectSoundCaptureCreateProc)pContext->dsound.DirectSoundCaptureCreate)((pDeviceID == NULL) ? NULL : (const GUID*)pDeviceID->dsound, &pDirectSoundCapture, NULL))) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[DirectSound] DirectSoundCaptureCreate() failed for capture device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - return pID0->winmm == pID1->winmm; + *ppDirectSoundCapture = pDirectSoundCapture; + return MA_SUCCESS; } -ma_result ma_context_enumerate_devices__winmm(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +static ma_result ma_context_get_format_info_for_IDirectSoundCapture__dsound(ma_context* pContext, ma_IDirectSoundCapture* pDirectSoundCapture, WORD* pChannels, WORD* pBitsPerSample, DWORD* pSampleRate) { - UINT playbackDeviceCount; - UINT captureDeviceCount; - UINT iPlaybackDevice; - UINT iCaptureDevice; + MA_DSCCAPS caps; + WORD bitsPerSample; + DWORD sampleRate; - ma_assert(pContext != NULL); - ma_assert(callback != NULL); + MA_ASSERT(pContext != NULL); + MA_ASSERT(pDirectSoundCapture != NULL); - /* Playback. */ - playbackDeviceCount = ((MA_PFN_waveOutGetNumDevs)pContext->winmm.waveOutGetNumDevs)(); - for (iPlaybackDevice = 0; iPlaybackDevice < playbackDeviceCount; ++iPlaybackDevice) { - MMRESULT result; - MA_WAVEOUTCAPS2A caps; + if (pChannels) { + *pChannels = 0; + } + if (pBitsPerSample) { + *pBitsPerSample = 0; + } + if (pSampleRate) { + *pSampleRate = 0; + } - ma_zero_object(&caps); + MA_ZERO_OBJECT(&caps); + caps.dwSize = sizeof(caps); + if (FAILED(ma_IDirectSoundCapture_GetCaps(pDirectSoundCapture, &caps))) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCapture_GetCaps() failed for capture device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - result = ((MA_PFN_waveOutGetDevCapsA)pContext->winmm.waveOutGetDevCapsA)(iPlaybackDevice, (WAVEOUTCAPSA*)&caps, sizeof(caps)); - if (result == MMSYSERR_NOERROR) { - ma_device_info deviceInfo; + if (pChannels) { + *pChannels = (WORD)caps.dwChannels; + } - ma_zero_object(&deviceInfo); - deviceInfo.id.winmm = iPlaybackDevice; + /* The device can support multiple formats. We just go through the different formats in order of priority and pick the first one. This the same type of system as the WinMM backend. */ + bitsPerSample = 16; + sampleRate = 48000; - if (ma_context_get_device_info_from_WAVEOUTCAPS2(pContext, &caps, &deviceInfo) == MA_SUCCESS) { - ma_bool32 cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); - if (cbResult == MA_FALSE) { - return MA_SUCCESS; /* Enumeration was stopped. */ - } + if (caps.dwChannels == 1) { + if ((caps.dwFormats & WAVE_FORMAT_48M16) != 0) { + sampleRate = 48000; + } else if ((caps.dwFormats & WAVE_FORMAT_44M16) != 0) { + sampleRate = 44100; + } else if ((caps.dwFormats & WAVE_FORMAT_2M16) != 0) { + sampleRate = 22050; + } else if ((caps.dwFormats & WAVE_FORMAT_1M16) != 0) { + sampleRate = 11025; + } else if ((caps.dwFormats & WAVE_FORMAT_96M16) != 0) { + sampleRate = 96000; + } else { + bitsPerSample = 8; + if ((caps.dwFormats & WAVE_FORMAT_48M08) != 0) { + sampleRate = 48000; + } else if ((caps.dwFormats & WAVE_FORMAT_44M08) != 0) { + sampleRate = 44100; + } else if ((caps.dwFormats & WAVE_FORMAT_2M08) != 0) { + sampleRate = 22050; + } else if ((caps.dwFormats & WAVE_FORMAT_1M08) != 0) { + sampleRate = 11025; + } else if ((caps.dwFormats & WAVE_FORMAT_96M08) != 0) { + sampleRate = 96000; + } else { + bitsPerSample = 16; /* Didn't find it. Just fall back to 16-bit. */ + } + } + } else if (caps.dwChannels == 2) { + if ((caps.dwFormats & WAVE_FORMAT_48S16) != 0) { + sampleRate = 48000; + } else if ((caps.dwFormats & WAVE_FORMAT_44S16) != 0) { + sampleRate = 44100; + } else if ((caps.dwFormats & WAVE_FORMAT_2S16) != 0) { + sampleRate = 22050; + } else if ((caps.dwFormats & WAVE_FORMAT_1S16) != 0) { + sampleRate = 11025; + } else if ((caps.dwFormats & WAVE_FORMAT_96S16) != 0) { + sampleRate = 96000; + } else { + bitsPerSample = 8; + if ((caps.dwFormats & WAVE_FORMAT_48S08) != 0) { + sampleRate = 48000; + } else if ((caps.dwFormats & WAVE_FORMAT_44S08) != 0) { + sampleRate = 44100; + } else if ((caps.dwFormats & WAVE_FORMAT_2S08) != 0) { + sampleRate = 22050; + } else if ((caps.dwFormats & WAVE_FORMAT_1S08) != 0) { + sampleRate = 11025; + } else if ((caps.dwFormats & WAVE_FORMAT_96S08) != 0) { + sampleRate = 96000; + } else { + bitsPerSample = 16; /* Didn't find it. Just fall back to 16-bit. */ } } } - /* Capture. */ - captureDeviceCount = ((MA_PFN_waveInGetNumDevs)pContext->winmm.waveInGetNumDevs)(); - for (iCaptureDevice = 0; iCaptureDevice < captureDeviceCount; ++iCaptureDevice) { - MMRESULT result; - MA_WAVEINCAPS2A caps; - - ma_zero_object(&caps); - - result = ((MA_PFN_waveInGetDevCapsA)pContext->winmm.waveInGetDevCapsA)(iCaptureDevice, (WAVEINCAPSA*)&caps, sizeof(caps)); - if (result == MMSYSERR_NOERROR) { - ma_device_info deviceInfo; - - ma_zero_object(&deviceInfo); - deviceInfo.id.winmm = iCaptureDevice; - - if (ma_context_get_device_info_from_WAVEINCAPS2(pContext, &caps, &deviceInfo) == MA_SUCCESS) { - ma_bool32 cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); - if (cbResult == MA_FALSE) { - return MA_SUCCESS; /* Enumeration was stopped. */ - } - } - } + if (pBitsPerSample) { + *pBitsPerSample = bitsPerSample; + } + if (pSampleRate) { + *pSampleRate = sampleRate; } return MA_SUCCESS; } -ma_result ma_context_get_device_info__winmm(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) +static ma_bool32 ma_context_is_device_id_equal__dsound(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) { - UINT winMMDeviceID; + MA_ASSERT(pContext != NULL); + MA_ASSERT(pID0 != NULL); + MA_ASSERT(pID1 != NULL); + (void)pContext; - ma_assert(pContext != NULL); + return memcmp(pID0->dsound, pID1->dsound, sizeof(pID0->dsound)) == 0; +} - if (shareMode == ma_share_mode_exclusive) { - return MA_SHARE_MODE_NOT_SUPPORTED; - } - winMMDeviceID = 0; - if (pDeviceID != NULL) { - winMMDeviceID = (UINT)pDeviceID->winmm; - } +typedef struct +{ + ma_context* pContext; + ma_device_type deviceType; + ma_enum_devices_callback_proc callback; + void* pUserData; + ma_bool32 terminated; +} ma_context_enumerate_devices_callback_data__dsound; - pDeviceInfo->id.winmm = winMMDeviceID; +static BOOL CALLBACK ma_context_enumerate_devices_callback__dsound(LPGUID lpGuid, LPCSTR lpcstrDescription, LPCSTR lpcstrModule, LPVOID lpContext) +{ + ma_context_enumerate_devices_callback_data__dsound* pData = (ma_context_enumerate_devices_callback_data__dsound*)lpContext; + ma_device_info deviceInfo; - if (deviceType == ma_device_type_playback) { - MMRESULT result; - MA_WAVEOUTCAPS2A caps; + MA_ZERO_OBJECT(&deviceInfo); - ma_zero_object(&caps); - - result = ((MA_PFN_waveOutGetDevCapsA)pContext->winmm.waveOutGetDevCapsA)(winMMDeviceID, (WAVEOUTCAPSA*)&caps, sizeof(caps)); - if (result == MMSYSERR_NOERROR) { - return ma_context_get_device_info_from_WAVEOUTCAPS2(pContext, &caps, pDeviceInfo); - } + /* ID. */ + if (lpGuid != NULL) { + MA_COPY_MEMORY(deviceInfo.id.dsound, lpGuid, 16); } else { - MMRESULT result; - MA_WAVEINCAPS2A caps; - - ma_zero_object(&caps); - - result = ((MA_PFN_waveInGetDevCapsA)pContext->winmm.waveInGetDevCapsA)(winMMDeviceID, (WAVEINCAPSA*)&caps, sizeof(caps)); - if (result == MMSYSERR_NOERROR) { - return ma_context_get_device_info_from_WAVEINCAPS2(pContext, &caps, pDeviceInfo); - } + MA_ZERO_MEMORY(deviceInfo.id.dsound, 16); } - return MA_NO_DEVICE; -} - - -void ma_device_uninit__winmm(ma_device* pDevice) -{ - ma_assert(pDevice != NULL); + /* Name / Description */ + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), lpcstrDescription, (size_t)-1); - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - ((MA_PFN_waveInClose)pDevice->pContext->winmm.waveInClose)((HWAVEIN)pDevice->winmm.hDeviceCapture); - CloseHandle((HANDLE)pDevice->winmm.hEventCapture); - } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - ((MA_PFN_waveOutReset)pDevice->pContext->winmm.waveOutReset)((HWAVEOUT)pDevice->winmm.hDevicePlayback); - ((MA_PFN_waveOutClose)pDevice->pContext->winmm.waveOutClose)((HWAVEOUT)pDevice->winmm.hDevicePlayback); - CloseHandle((HANDLE)pDevice->winmm.hEventPlayback); + /* Call the callback function, but make sure we stop enumerating if the callee requested so. */ + MA_ASSERT(pData != NULL); + pData->terminated = !pData->callback(pData->pContext, pData->deviceType, &deviceInfo, pData->pUserData); + if (pData->terminated) { + return FALSE; /* Stop enumeration. */ + } else { + return TRUE; /* Continue enumeration. */ } - ma_free(pDevice->winmm._pHeapData); - - ma_zero_object(&pDevice->winmm); /* Safety. */ + (void)lpcstrModule; } -ma_result ma_device_init__winmm(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) +static ma_result ma_context_enumerate_devices__dsound(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { - const char* errorMsg = ""; - ma_result errorCode = MA_ERROR; - ma_result result = MA_SUCCESS; - ma_uint32 heapSize; - UINT winMMDeviceIDPlayback = 0; - UINT winMMDeviceIDCapture = 0; - ma_uint32 bufferSizeInMilliseconds; + ma_context_enumerate_devices_callback_data__dsound data; - ma_assert(pDevice != NULL); - ma_zero_object(&pDevice->winmm); + MA_ASSERT(pContext != NULL); + MA_ASSERT(callback != NULL); - if (pConfig->deviceType == ma_device_type_loopback) { - return MA_DEVICE_TYPE_NOT_SUPPORTED; - } + data.pContext = pContext; + data.callback = callback; + data.pUserData = pUserData; + data.terminated = MA_FALSE; - /* No exlusive mode with WinMM. */ - if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) || - ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) { - return MA_SHARE_MODE_NOT_SUPPORTED; + /* Playback. */ + if (!data.terminated) { + data.deviceType = ma_device_type_playback; + ((ma_DirectSoundEnumerateAProc)pContext->dsound.DirectSoundEnumerateA)(ma_context_enumerate_devices_callback__dsound, &data); } - bufferSizeInMilliseconds = pConfig->bufferSizeInMilliseconds; - if (bufferSizeInMilliseconds == 0) { - bufferSizeInMilliseconds = ma_calculate_buffer_size_in_milliseconds_from_frames(pConfig->bufferSizeInFrames, pConfig->sampleRate); - } - - /* WinMM has horrible latency. */ - if (pDevice->usingDefaultBufferSize) { - if (pConfig->performanceProfile == ma_performance_profile_low_latency) { - bufferSizeInMilliseconds = 40 * pConfig->periods; - } else { - bufferSizeInMilliseconds = 400 * pConfig->periods; - } + /* Capture. */ + if (!data.terminated) { + data.deviceType = ma_device_type_capture; + ((ma_DirectSoundCaptureEnumerateAProc)pContext->dsound.DirectSoundCaptureEnumerateA)(ma_context_enumerate_devices_callback__dsound, &data); } + return MA_SUCCESS; +} - if (pConfig->playback.pDeviceID != NULL) { - winMMDeviceIDPlayback = (UINT)pConfig->playback.pDeviceID->winmm; - } - if (pConfig->capture.pDeviceID != NULL) { - winMMDeviceIDCapture = (UINT)pConfig->capture.pDeviceID->winmm; - } - - /* The capture device needs to be initialized first. */ - if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { - WAVEINCAPSA caps; - WAVEFORMATEX wf; - MMRESULT resultMM; - /* We use an event to know when a new fragment needs to be enqueued. */ - pDevice->winmm.hEventCapture = (ma_handle)CreateEvent(NULL, TRUE, TRUE, NULL); - if (pDevice->winmm.hEventCapture == NULL) { - errorMsg = "[WinMM] Failed to create event for fragment enqueing for the capture device.", errorCode = MA_FAILED_TO_CREATE_EVENT; - goto on_error; - } +typedef struct +{ + const ma_device_id* pDeviceID; + ma_device_info* pDeviceInfo; + ma_bool32 found; +} ma_context_get_device_info_callback_data__dsound; - /* The format should be based on the device's actual format. */ - if (((MA_PFN_waveInGetDevCapsA)pContext->winmm.waveInGetDevCapsA)(winMMDeviceIDCapture, &caps, sizeof(caps)) != MMSYSERR_NOERROR) { - errorMsg = "[WinMM] Failed to retrieve internal device caps.", errorCode = MA_FORMAT_NOT_SUPPORTED; - goto on_error; - } +static BOOL CALLBACK ma_context_get_device_info_callback__dsound(LPGUID lpGuid, LPCSTR lpcstrDescription, LPCSTR lpcstrModule, LPVOID lpContext) +{ + ma_context_get_device_info_callback_data__dsound* pData = (ma_context_get_device_info_callback_data__dsound*)lpContext; + MA_ASSERT(pData != NULL); - result = ma_formats_flags_to_WAVEFORMATEX__winmm(caps.dwFormats, caps.wChannels, &wf); - if (result != MA_SUCCESS) { - errorMsg = "[WinMM] Could not find appropriate format for internal device.", errorCode = result; - goto on_error; + if ((pData->pDeviceID == NULL || ma_is_guid_equal(pData->pDeviceID->dsound, &MA_GUID_NULL)) && (lpGuid == NULL || ma_is_guid_equal(lpGuid, &MA_GUID_NULL))) { + /* Default device. */ + ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), lpcstrDescription, (size_t)-1); + pData->found = MA_TRUE; + return FALSE; /* Stop enumeration. */ + } else { + /* Not the default device. */ + if (lpGuid != NULL && pData->pDeviceID != NULL) { + if (memcmp(pData->pDeviceID->dsound, lpGuid, sizeof(pData->pDeviceID->dsound)) == 0) { + ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), lpcstrDescription, (size_t)-1); + pData->found = MA_TRUE; + return FALSE; /* Stop enumeration. */ + } } + } - resultMM = ((MA_PFN_waveInOpen)pDevice->pContext->winmm.waveInOpen)((LPHWAVEIN)&pDevice->winmm.hDeviceCapture, winMMDeviceIDCapture, &wf, (DWORD_PTR)pDevice->winmm.hEventCapture, (DWORD_PTR)pDevice, CALLBACK_EVENT | WAVE_ALLOWSYNC); - if (resultMM != MMSYSERR_NOERROR) { - errorMsg = "[WinMM] Failed to open capture device.", errorCode = MA_FAILED_TO_OPEN_BACKEND_DEVICE; - goto on_error; - } + (void)lpcstrModule; + return TRUE; +} - pDevice->capture.internalFormat = ma_format_from_WAVEFORMATEX(&wf); - pDevice->capture.internalChannels = wf.nChannels; - pDevice->capture.internalSampleRate = wf.nSamplesPerSec; - ma_get_standard_channel_map(ma_standard_channel_map_microsoft, pDevice->capture.internalChannels, pDevice->capture.internalChannelMap); - pDevice->capture.internalPeriods = pConfig->periods; - pDevice->capture.internalBufferSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(bufferSizeInMilliseconds, pDevice->capture.internalSampleRate); +static ma_result ma_context_get_device_info__dsound(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) +{ + /* Exclusive mode and capture not supported with DirectSound. */ + if (deviceType == ma_device_type_capture && shareMode == ma_share_mode_exclusive) { + return MA_SHARE_MODE_NOT_SUPPORTED; } - if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { - WAVEOUTCAPSA caps; - WAVEFORMATEX wf; - MMRESULT resultMM; + if (pDeviceID != NULL) { + ma_context_get_device_info_callback_data__dsound data; - /* We use an event to know when a new fragment needs to be enqueued. */ - pDevice->winmm.hEventPlayback = (ma_handle)CreateEvent(NULL, TRUE, TRUE, NULL); - if (pDevice->winmm.hEventPlayback == NULL) { - errorMsg = "[WinMM] Failed to create event for fragment enqueing for the playback device.", errorCode = MA_FAILED_TO_CREATE_EVENT; - goto on_error; - } + /* ID. */ + MA_COPY_MEMORY(pDeviceInfo->id.dsound, pDeviceID->dsound, 16); - /* The format should be based on the device's actual format. */ - if (((MA_PFN_waveOutGetDevCapsA)pContext->winmm.waveOutGetDevCapsA)(winMMDeviceIDPlayback, &caps, sizeof(caps)) != MMSYSERR_NOERROR) { - errorMsg = "[WinMM] Failed to retrieve internal device caps.", errorCode = MA_FORMAT_NOT_SUPPORTED; - goto on_error; + /* Name / Description. This is retrieved by enumerating over each device until we find that one that matches the input ID. */ + data.pDeviceID = pDeviceID; + data.pDeviceInfo = pDeviceInfo; + data.found = MA_FALSE; + if (deviceType == ma_device_type_playback) { + ((ma_DirectSoundEnumerateAProc)pContext->dsound.DirectSoundEnumerateA)(ma_context_get_device_info_callback__dsound, &data); + } else { + ((ma_DirectSoundCaptureEnumerateAProc)pContext->dsound.DirectSoundCaptureEnumerateA)(ma_context_get_device_info_callback__dsound, &data); } - result = ma_formats_flags_to_WAVEFORMATEX__winmm(caps.dwFormats, caps.wChannels, &wf); - if (result != MA_SUCCESS) { - errorMsg = "[WinMM] Could not find appropriate format for internal device.", errorCode = result; - goto on_error; + if (!data.found) { + return MA_NO_DEVICE; } + } else { + /* I don't think there's a way to get the name of the default device with DirectSound. In this case we just need to use defaults. */ - resultMM = ((MA_PFN_waveOutOpen)pContext->winmm.waveOutOpen)((LPHWAVEOUT)&pDevice->winmm.hDevicePlayback, winMMDeviceIDPlayback, &wf, (DWORD_PTR)pDevice->winmm.hEventPlayback, (DWORD_PTR)pDevice, CALLBACK_EVENT | WAVE_ALLOWSYNC); - if (resultMM != MMSYSERR_NOERROR) { - errorMsg = "[WinMM] Failed to open playback device.", errorCode = MA_FAILED_TO_OPEN_BACKEND_DEVICE; - goto on_error; - } + /* ID */ + MA_ZERO_MEMORY(pDeviceInfo->id.dsound, 16); - pDevice->playback.internalFormat = ma_format_from_WAVEFORMATEX(&wf); - pDevice->playback.internalChannels = wf.nChannels; - pDevice->playback.internalSampleRate = wf.nSamplesPerSec; - ma_get_standard_channel_map(ma_standard_channel_map_microsoft, pDevice->playback.internalChannels, pDevice->playback.internalChannelMap); - pDevice->playback.internalPeriods = pConfig->periods; - pDevice->playback.internalBufferSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(bufferSizeInMilliseconds, pDevice->playback.internalSampleRate); + /* Name / Description */ + if (deviceType == ma_device_type_playback) { + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); + } else { + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); + } } - /* - The heap allocated data is allocated like so: - - [Capture WAVEHDRs][Playback WAVEHDRs][Capture Intermediary Buffer][Playback Intermediary Buffer] - */ - heapSize = 0; - if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { - heapSize += sizeof(WAVEHDR)*pDevice->capture.internalPeriods + (pDevice->capture.internalBufferSizeInFrames*ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); - } - if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { - heapSize += sizeof(WAVEHDR)*pDevice->playback.internalPeriods + (pDevice->playback.internalBufferSizeInFrames*ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); - } + /* Retrieving detailed information is slightly different depending on the device type. */ + if (deviceType == ma_device_type_playback) { + /* Playback. */ + ma_IDirectSound* pDirectSound; + ma_result result; + MA_DSCAPS caps; + ma_uint32 iFormat; - pDevice->winmm._pHeapData = (ma_uint8*)ma_malloc(heapSize); - if (pDevice->winmm._pHeapData == NULL) { - errorMsg = "[WinMM] Failed to allocate memory for the intermediary buffer.", errorCode = MA_OUT_OF_MEMORY; - goto on_error; - } + result = ma_context_create_IDirectSound__dsound(pContext, shareMode, pDeviceID, &pDirectSound); + if (result != MA_SUCCESS) { + return result; + } + + MA_ZERO_OBJECT(&caps); + caps.dwSize = sizeof(caps); + if (FAILED(ma_IDirectSound_GetCaps(pDirectSound, &caps))) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_GetCaps() failed for playback device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - ma_zero_memory(pDevice->winmm._pHeapData, heapSize); + if ((caps.dwFlags & MA_DSCAPS_PRIMARYSTEREO) != 0) { + /* It supports at least stereo, but could support more. */ + WORD channels = 2; - if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { - ma_uint32 iPeriod; + /* Look at the speaker configuration to get a better idea on the channel count. */ + DWORD speakerConfig; + if (SUCCEEDED(ma_IDirectSound_GetSpeakerConfig(pDirectSound, &speakerConfig))) { + ma_get_channels_from_speaker_config__dsound(speakerConfig, &channels, NULL); + } - if (pConfig->deviceType == ma_device_type_capture) { - pDevice->winmm.pWAVEHDRCapture = pDevice->winmm._pHeapData; - pDevice->winmm.pIntermediaryBufferCapture = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDevice->capture.internalPeriods)); + pDeviceInfo->minChannels = channels; + pDeviceInfo->maxChannels = channels; } else { - pDevice->winmm.pWAVEHDRCapture = pDevice->winmm._pHeapData; - pDevice->winmm.pIntermediaryBufferCapture = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDevice->capture.internalPeriods + pDevice->playback.internalPeriods)); + /* It does not support stereo, which means we are stuck with mono. */ + pDeviceInfo->minChannels = 1; + pDeviceInfo->maxChannels = 1; } - /* Prepare headers. */ - for (iPeriod = 0; iPeriod < pDevice->capture.internalPeriods; ++iPeriod) { - ma_uint32 fragmentSizeInBytes = ma_get_fragment_size_in_bytes(pDevice->capture.internalBufferSizeInFrames, pDevice->capture.internalPeriods, pDevice->capture.internalFormat, pDevice->capture.internalChannels); - - ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].lpData = (LPSTR)(pDevice->winmm.pIntermediaryBufferCapture + (fragmentSizeInBytes*iPeriod)); - ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwBufferLength = fragmentSizeInBytes; - ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwFlags = 0L; - ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwLoops = 0L; - ((MA_PFN_waveInPrepareHeader)pContext->winmm.waveInPrepareHeader)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod], sizeof(WAVEHDR)); + /* Sample rate. */ + if ((caps.dwFlags & MA_DSCAPS_CONTINUOUSRATE) != 0) { + pDeviceInfo->minSampleRate = caps.dwMinSecondarySampleRate; + pDeviceInfo->maxSampleRate = caps.dwMaxSecondarySampleRate; /* - The user data of the WAVEHDR structure is a single flag the controls whether or not it is ready for writing. Consider it to be named "isLocked". A value of 0 means - it's unlocked and available for writing. A value of 1 means it's locked. + On my machine the min and max sample rates can return 100 and 200000 respectively. I'd rather these be within + the range of our standard sample rates so I'm clamping. */ - ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwUser = 0; + if (caps.dwMinSecondarySampleRate < MA_MIN_SAMPLE_RATE && caps.dwMaxSecondarySampleRate >= MA_MIN_SAMPLE_RATE) { + pDeviceInfo->minSampleRate = MA_MIN_SAMPLE_RATE; + } + if (caps.dwMaxSecondarySampleRate > MA_MAX_SAMPLE_RATE && caps.dwMinSecondarySampleRate <= MA_MAX_SAMPLE_RATE) { + pDeviceInfo->maxSampleRate = MA_MAX_SAMPLE_RATE; + } + } else { + /* Only supports a single sample rate. Set both min an max to the same thing. Do not clamp within the standard rates. */ + pDeviceInfo->minSampleRate = caps.dwMaxSecondarySampleRate; + pDeviceInfo->maxSampleRate = caps.dwMaxSecondarySampleRate; } - } - if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { - ma_uint32 iPeriod; - if (pConfig->deviceType == ma_device_type_playback) { - pDevice->winmm.pWAVEHDRPlayback = pDevice->winmm._pHeapData; - pDevice->winmm.pIntermediaryBufferPlayback = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*pDevice->playback.internalPeriods); - } else { - pDevice->winmm.pWAVEHDRPlayback = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDevice->capture.internalPeriods)); - pDevice->winmm.pIntermediaryBufferPlayback = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDevice->capture.internalPeriods + pDevice->playback.internalPeriods)) + (pDevice->playback.internalBufferSizeInFrames*ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); + /* DirectSound can support all formats. */ + pDeviceInfo->formatCount = ma_format_count - 1; /* Minus one because we don't want to include ma_format_unknown. */ + for (iFormat = 0; iFormat < pDeviceInfo->formatCount; ++iFormat) { + pDeviceInfo->formats[iFormat] = (ma_format)(iFormat + 1); /* +1 to skip over ma_format_unknown. */ } - /* Prepare headers. */ - for (iPeriod = 0; iPeriod < pDevice->playback.internalPeriods; ++iPeriod) { - ma_uint32 fragmentSizeInBytes = ma_get_fragment_size_in_bytes(pDevice->playback.internalBufferSizeInFrames, pDevice->playback.internalPeriods, pDevice->playback.internalFormat, pDevice->playback.internalChannels); + ma_IDirectSound_Release(pDirectSound); + } else { + /* + Capture. This is a little different to playback due to the say the supported formats are reported. Technically capture + devices can support a number of different formats, but for simplicity and consistency with ma_device_init() I'm just + reporting the best format. + */ + ma_IDirectSoundCapture* pDirectSoundCapture; + ma_result result; + WORD channels; + WORD bitsPerSample; + DWORD sampleRate; - ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].lpData = (LPSTR)(pDevice->winmm.pIntermediaryBufferPlayback + (fragmentSizeInBytes*iPeriod)); - ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwBufferLength = fragmentSizeInBytes; - ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwFlags = 0L; - ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwLoops = 0L; - ((MA_PFN_waveOutPrepareHeader)pContext->winmm.waveOutPrepareHeader)((HWAVEOUT)pDevice->winmm.hDevicePlayback, &((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod], sizeof(WAVEHDR)); + result = ma_context_create_IDirectSoundCapture__dsound(pContext, shareMode, pDeviceID, &pDirectSoundCapture); + if (result != MA_SUCCESS) { + return result; + } - /* - The user data of the WAVEHDR structure is a single flag the controls whether or not it is ready for writing. Consider it to be named "isLocked". A value of 0 means - it's unlocked and available for writing. A value of 1 means it's locked. - */ - ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwUser = 0; + result = ma_context_get_format_info_for_IDirectSoundCapture__dsound(pContext, pDirectSoundCapture, &channels, &bitsPerSample, &sampleRate); + if (result != MA_SUCCESS) { + ma_IDirectSoundCapture_Release(pDirectSoundCapture); + return result; + } + + pDeviceInfo->minChannels = channels; + pDeviceInfo->maxChannels = channels; + pDeviceInfo->minSampleRate = sampleRate; + pDeviceInfo->maxSampleRate = sampleRate; + pDeviceInfo->formatCount = 1; + if (bitsPerSample == 8) { + pDeviceInfo->formats[0] = ma_format_u8; + } else if (bitsPerSample == 16) { + pDeviceInfo->formats[0] = ma_format_s16; + } else if (bitsPerSample == 24) { + pDeviceInfo->formats[0] = ma_format_s24; + } else if (bitsPerSample == 32) { + pDeviceInfo->formats[0] = ma_format_s32; + } else { + ma_IDirectSoundCapture_Release(pDirectSoundCapture); + return MA_FORMAT_NOT_SUPPORTED; } + + ma_IDirectSoundCapture_Release(pDirectSoundCapture); } return MA_SUCCESS; +} -on_error: - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - if (pDevice->winmm.pWAVEHDRCapture != NULL) { - ma_uint32 iPeriod; - for (iPeriod = 0; iPeriod < pDevice->capture.internalPeriods; ++iPeriod) { - ((MA_PFN_waveInUnprepareHeader)pContext->winmm.waveInUnprepareHeader)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod], sizeof(WAVEHDR)); - } - } - ((MA_PFN_waveInClose)pContext->winmm.waveInClose)((HWAVEIN)pDevice->winmm.hDeviceCapture); - } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - if (pDevice->winmm.pWAVEHDRCapture != NULL) { - ma_uint32 iPeriod; - for (iPeriod = 0; iPeriod < pDevice->playback.internalPeriods; ++iPeriod) { - ((MA_PFN_waveOutUnprepareHeader)pContext->winmm.waveOutUnprepareHeader)((HWAVEOUT)pDevice->winmm.hDevicePlayback, &((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod], sizeof(WAVEHDR)); - } - } +static void ma_device_uninit__dsound(ma_device* pDevice) +{ + MA_ASSERT(pDevice != NULL); - ((MA_PFN_waveOutClose)pContext->winmm.waveOutClose)((HWAVEOUT)pDevice->winmm.hDevicePlayback); + if (pDevice->dsound.pCaptureBuffer != NULL) { + ma_IDirectSoundCaptureBuffer_Release((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer); + } + if (pDevice->dsound.pCapture != NULL) { + ma_IDirectSoundCapture_Release((ma_IDirectSoundCapture*)pDevice->dsound.pCapture); } - ma_free(pDevice->winmm._pHeapData); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, errorMsg, errorCode); + if (pDevice->dsound.pPlaybackBuffer != NULL) { + ma_IDirectSoundBuffer_Release((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer); + } + if (pDevice->dsound.pPlaybackPrimaryBuffer != NULL) { + ma_IDirectSoundBuffer_Release((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer); + } + if (pDevice->dsound.pPlayback != NULL) { + ma_IDirectSound_Release((ma_IDirectSound*)pDevice->dsound.pPlayback); + } } -ma_result ma_device_stop__winmm(ma_device* pDevice) +static ma_result ma_config_to_WAVEFORMATEXTENSIBLE(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, const ma_channel* pChannelMap, WAVEFORMATEXTENSIBLE* pWF) { - MMRESULT resultMM; + GUID subformat; - ma_assert(pDevice != NULL); + switch (format) + { + case ma_format_u8: + case ma_format_s16: + case ma_format_s24: + /*case ma_format_s24_32:*/ + case ma_format_s32: + { + subformat = MA_GUID_KSDATAFORMAT_SUBTYPE_PCM; + } break; - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - if (pDevice->winmm.hDeviceCapture == NULL) { - return MA_INVALID_ARGS; - } + case ma_format_f32: + { + subformat = MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT; + } break; - resultMM = ((MA_PFN_waveInReset)pDevice->pContext->winmm.waveInReset)((HWAVEIN)pDevice->winmm.hDeviceCapture); - if (resultMM != MMSYSERR_NOERROR) { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WinMM] WARNING: Failed to reset capture device.", ma_result_from_MMRESULT(resultMM)); - } + default: + return MA_FORMAT_NOT_SUPPORTED; } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - if (pDevice->winmm.hDevicePlayback == NULL) { - return MA_INVALID_ARGS; - } - - resultMM = ((MA_PFN_waveOutReset)pDevice->pContext->winmm.waveOutReset)((HWAVEOUT)pDevice->winmm.hDevicePlayback); - if (resultMM != MMSYSERR_NOERROR) { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WinMM] WARNING: Failed to reset playback device.", ma_result_from_MMRESULT(resultMM)); - } - } + MA_ZERO_OBJECT(pWF); + pWF->Format.cbSize = sizeof(*pWF); + pWF->Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE; + pWF->Format.nChannels = (WORD)channels; + pWF->Format.nSamplesPerSec = (DWORD)sampleRate; + pWF->Format.wBitsPerSample = (WORD)ma_get_bytes_per_sample(format)*8; + pWF->Format.nBlockAlign = (pWF->Format.nChannels * pWF->Format.wBitsPerSample) / 8; + pWF->Format.nAvgBytesPerSec = pWF->Format.nBlockAlign * pWF->Format.nSamplesPerSec; + pWF->Samples.wValidBitsPerSample = pWF->Format.wBitsPerSample; + pWF->dwChannelMask = ma_channel_map_to_channel_mask__win32(pChannelMap, channels); + pWF->SubFormat = subformat; return MA_SUCCESS; } -ma_result ma_device_write__winmm(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) +static ma_result ma_device_init__dsound(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) { - ma_result result = MA_SUCCESS; - MMRESULT resultMM; - ma_uint32 totalFramesWritten; - WAVEHDR* pWAVEHDR; + ma_result result; + ma_uint32 periodSizeInMilliseconds; - ma_assert(pDevice != NULL); - ma_assert(pPCMFrames != NULL); + MA_ASSERT(pDevice != NULL); + MA_ZERO_OBJECT(&pDevice->dsound); - if (pFramesWritten != NULL) { - *pFramesWritten = 0; + if (pConfig->deviceType == ma_device_type_loopback) { + return MA_DEVICE_TYPE_NOT_SUPPORTED; } - pWAVEHDR = (WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback; - - /* Keep processing as much data as possible. */ - totalFramesWritten = 0; - while (totalFramesWritten < frameCount) { - /* If the current header has some space available we need to write part of it. */ - if (pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwUser == 0) { /* 0 = unlocked. */ - /* - This header has room in it. We copy as much of it as we can. If we end up fully consuming the buffer we need to - write it out and move on to the next iteration. - */ - ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - ma_uint32 framesRemainingInHeader = (pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwBufferLength/bpf) - pDevice->winmm.headerFramesConsumedPlayback; + periodSizeInMilliseconds = pConfig->periodSizeInMilliseconds; + if (periodSizeInMilliseconds == 0) { + periodSizeInMilliseconds = ma_calculate_buffer_size_in_milliseconds_from_frames(pConfig->periodSizeInFrames, pConfig->sampleRate); + } + + /* DirectSound should use a latency of about 20ms per period for low latency mode. */ + if (pDevice->usingDefaultBufferSize) { + if (pConfig->performanceProfile == ma_performance_profile_low_latency) { + periodSizeInMilliseconds = 20; + } else { + periodSizeInMilliseconds = 200; + } + } - ma_uint32 framesToCopy = ma_min(framesRemainingInHeader, (frameCount - totalFramesWritten)); - const void* pSrc = ma_offset_ptr(pPCMFrames, totalFramesWritten*bpf); - void* pDst = ma_offset_ptr(pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].lpData, pDevice->winmm.headerFramesConsumedPlayback*bpf); - ma_copy_memory(pDst, pSrc, framesToCopy*bpf); + /* DirectSound breaks down with tiny buffer sizes (bad glitching and silent output). I am therefore restricting the size of the buffer to a minimum of 20 milliseconds. */ + if (periodSizeInMilliseconds < 20) { + periodSizeInMilliseconds = 20; + } - pDevice->winmm.headerFramesConsumedPlayback += framesToCopy; - totalFramesWritten += framesToCopy; + /* + Unfortunately DirectSound uses different APIs and data structures for playback and catpure devices. We need to initialize + the capture device first because we'll want to match it's buffer size and period count on the playback side if we're using + full-duplex mode. + */ + if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { + WAVEFORMATEXTENSIBLE wf; + MA_DSCBUFFERDESC descDS; + ma_uint32 periodSizeInFrames; + char rawdata[1024]; /* <-- Ugly hack to avoid a malloc() due to a crappy DirectSound API. */ + WAVEFORMATEXTENSIBLE* pActualFormat; - /* If we've consumed the buffer entirely we need to write it out to the device. */ - if (pDevice->winmm.headerFramesConsumedPlayback == (pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwBufferLength/bpf)) { - pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwUser = 1; /* 1 = locked. */ - pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwFlags &= ~WHDR_DONE; /* <-- Need to make sure the WHDR_DONE flag is unset. */ + result = ma_config_to_WAVEFORMATEXTENSIBLE(pConfig->capture.format, pConfig->capture.channels, pConfig->sampleRate, pConfig->capture.channelMap, &wf); + if (result != MA_SUCCESS) { + return result; + } - /* Make sure the event is reset to a non-signaled state to ensure we don't prematurely return from WaitForSingleObject(). */ - ResetEvent((HANDLE)pDevice->winmm.hEventPlayback); + result = ma_context_create_IDirectSoundCapture__dsound(pContext, pConfig->capture.shareMode, pConfig->capture.pDeviceID, (ma_IDirectSoundCapture**)&pDevice->dsound.pCapture); + if (result != MA_SUCCESS) { + ma_device_uninit__dsound(pDevice); + return result; + } - /* The device will be started here. */ - resultMM = ((MA_PFN_waveOutWrite)pDevice->pContext->winmm.waveOutWrite)((HWAVEOUT)pDevice->winmm.hDevicePlayback, &pWAVEHDR[pDevice->winmm.iNextHeaderPlayback], sizeof(WAVEHDR)); - if (resultMM != MMSYSERR_NOERROR) { - result = ma_result_from_MMRESULT(resultMM); - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WinMM] waveOutWrite() failed.", result); - break; - } + result = ma_context_get_format_info_for_IDirectSoundCapture__dsound(pContext, (ma_IDirectSoundCapture*)pDevice->dsound.pCapture, &wf.Format.nChannels, &wf.Format.wBitsPerSample, &wf.Format.nSamplesPerSec); + if (result != MA_SUCCESS) { + ma_device_uninit__dsound(pDevice); + return result; + } - /* Make sure we move to the next header. */ - pDevice->winmm.iNextHeaderPlayback = (pDevice->winmm.iNextHeaderPlayback + 1) % pDevice->playback.internalPeriods; - pDevice->winmm.headerFramesConsumedPlayback = 0; - } + wf.Format.nBlockAlign = (wf.Format.nChannels * wf.Format.wBitsPerSample) / 8; + wf.Format.nAvgBytesPerSec = wf.Format.nBlockAlign * wf.Format.nSamplesPerSec; + wf.Samples.wValidBitsPerSample = wf.Format.wBitsPerSample; + wf.SubFormat = MA_GUID_KSDATAFORMAT_SUBTYPE_PCM; - /* If at this point we have consumed the entire input buffer we can return. */ - ma_assert(totalFramesWritten <= frameCount); - if (totalFramesWritten == frameCount) { - break; - } + /* The size of the buffer must be a clean multiple of the period count. */ + periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(periodSizeInMilliseconds, wf.Format.nSamplesPerSec); - /* Getting here means there's more to process. */ - continue; + MA_ZERO_OBJECT(&descDS); + descDS.dwSize = sizeof(descDS); + descDS.dwFlags = 0; + descDS.dwBufferBytes = periodSizeInFrames * pConfig->periods * ma_get_bytes_per_frame(pDevice->capture.internalFormat, wf.Format.nChannels); + descDS.lpwfxFormat = (WAVEFORMATEX*)&wf; + if (FAILED(ma_IDirectSoundCapture_CreateCaptureBuffer((ma_IDirectSoundCapture*)pDevice->dsound.pCapture, &descDS, (ma_IDirectSoundCaptureBuffer**)&pDevice->dsound.pCaptureBuffer, NULL))) { + ma_device_uninit__dsound(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCapture_CreateCaptureBuffer() failed for capture device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); } - /* Getting here means there isn't enough room in the buffer and we need to wait for one to become available. */ - if (WaitForSingleObject((HANDLE)pDevice->winmm.hEventPlayback, INFINITE) != WAIT_OBJECT_0) { - result = MA_ERROR; - break; + /* Get the _actual_ properties of the buffer. */ + pActualFormat = (WAVEFORMATEXTENSIBLE*)rawdata; + if (FAILED(ma_IDirectSoundCaptureBuffer_GetFormat((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, (WAVEFORMATEX*)pActualFormat, sizeof(rawdata), NULL))) { + ma_device_uninit__dsound(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to retrieve the actual format of the capture device's buffer.", MA_FORMAT_NOT_SUPPORTED); } - /* Something happened. If the next buffer has been marked as done we need to reset a bit of state. */ - if ((pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwFlags & WHDR_DONE) != 0) { - pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwUser = 0; /* 0 = unlocked (make it available for writing). */ - pDevice->winmm.headerFramesConsumedPlayback = 0; - } + pDevice->capture.internalFormat = ma_format_from_WAVEFORMATEX((WAVEFORMATEX*)pActualFormat); + pDevice->capture.internalChannels = pActualFormat->Format.nChannels; + pDevice->capture.internalSampleRate = pActualFormat->Format.nSamplesPerSec; - /* If the device has been stopped we need to break. */ - if (ma_device__get_state(pDevice) != MA_STATE_STARTED) { - break; + /* Get the internal channel map based on the channel mask. */ + if (pActualFormat->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE) { + ma_channel_mask_to_channel_map__win32(pActualFormat->dwChannelMask, pDevice->capture.internalChannels, pDevice->capture.internalChannelMap); + } else { + ma_channel_mask_to_channel_map__win32(wf.dwChannelMask, pDevice->capture.internalChannels, pDevice->capture.internalChannelMap); } - } - if (pFramesWritten != NULL) { - *pFramesWritten = totalFramesWritten; - } + /* + After getting the actual format the size of the buffer in frames may have actually changed. However, we want this to be as close to what the + user has asked for as possible, so let's go ahead and release the old capture buffer and create a new one in this case. + */ + if (periodSizeInFrames != (descDS.dwBufferBytes / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels) / pConfig->periods)) { + descDS.dwBufferBytes = periodSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.internalFormat, wf.Format.nChannels) * pConfig->periods; + ma_IDirectSoundCaptureBuffer_Release((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer); - return result; -} + if (FAILED(ma_IDirectSoundCapture_CreateCaptureBuffer((ma_IDirectSoundCapture*)pDevice->dsound.pCapture, &descDS, (ma_IDirectSoundCaptureBuffer**)&pDevice->dsound.pCaptureBuffer, NULL))) { + ma_device_uninit__dsound(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Second attempt at IDirectSoundCapture_CreateCaptureBuffer() failed for capture device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } + } -ma_result ma_device_read__winmm(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead) -{ - ma_result result = MA_SUCCESS; - MMRESULT resultMM; - ma_uint32 totalFramesRead; - WAVEHDR* pWAVEHDR; + /* DirectSound should give us a buffer exactly the size we asked for. */ + pDevice->capture.internalPeriodSizeInFrames = periodSizeInFrames; + pDevice->capture.internalPeriods = pConfig->periods; + } - ma_assert(pDevice != NULL); - ma_assert(pPCMFrames != NULL); + if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { + WAVEFORMATEXTENSIBLE wf; + MA_DSBUFFERDESC descDSPrimary; + MA_DSCAPS caps; + char rawdata[1024]; /* <-- Ugly hack to avoid a malloc() due to a crappy DirectSound API. */ + WAVEFORMATEXTENSIBLE* pActualFormat; + ma_uint32 periodSizeInFrames; + MA_DSBUFFERDESC descDS; - if (pFramesRead != NULL) { - *pFramesRead = 0; - } + result = ma_config_to_WAVEFORMATEXTENSIBLE(pConfig->playback.format, pConfig->playback.channels, pConfig->sampleRate, pConfig->playback.channelMap, &wf); + if (result != MA_SUCCESS) { + return result; + } - pWAVEHDR = (WAVEHDR*)pDevice->winmm.pWAVEHDRCapture; + result = ma_context_create_IDirectSound__dsound(pContext, pConfig->playback.shareMode, pConfig->playback.pDeviceID, (ma_IDirectSound**)&pDevice->dsound.pPlayback); + if (result != MA_SUCCESS) { + ma_device_uninit__dsound(pDevice); + return result; + } - /* Keep processing as much data as possible. */ - totalFramesRead = 0; - while (totalFramesRead < frameCount) { - /* If the current header has some space available we need to write part of it. */ - if (pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwUser == 0) { /* 0 = unlocked. */ - /* The buffer is available for reading. If we fully consume it we need to add it back to the buffer. */ - ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 framesRemainingInHeader = (pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwBufferLength/bpf) - pDevice->winmm.headerFramesConsumedCapture; + MA_ZERO_OBJECT(&descDSPrimary); + descDSPrimary.dwSize = sizeof(MA_DSBUFFERDESC); + descDSPrimary.dwFlags = MA_DSBCAPS_PRIMARYBUFFER | MA_DSBCAPS_CTRLVOLUME; + if (FAILED(ma_IDirectSound_CreateSoundBuffer((ma_IDirectSound*)pDevice->dsound.pPlayback, &descDSPrimary, (ma_IDirectSoundBuffer**)&pDevice->dsound.pPlaybackPrimaryBuffer, NULL))) { + ma_device_uninit__dsound(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_CreateSoundBuffer() failed for playback device's primary buffer.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - ma_uint32 framesToCopy = ma_min(framesRemainingInHeader, (frameCount - totalFramesRead)); - const void* pSrc = ma_offset_ptr(pWAVEHDR[pDevice->winmm.iNextHeaderCapture].lpData, pDevice->winmm.headerFramesConsumedCapture*bpf); - void* pDst = ma_offset_ptr(pPCMFrames, totalFramesRead*bpf); - ma_copy_memory(pDst, pSrc, framesToCopy*bpf); - pDevice->winmm.headerFramesConsumedCapture += framesToCopy; - totalFramesRead += framesToCopy; + /* We may want to make some adjustments to the format if we are using defaults. */ + MA_ZERO_OBJECT(&caps); + caps.dwSize = sizeof(caps); + if (FAILED(ma_IDirectSound_GetCaps((ma_IDirectSound*)pDevice->dsound.pPlayback, &caps))) { + ma_device_uninit__dsound(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_GetCaps() failed for playback device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - /* If we've consumed the buffer entirely we need to add it back to the device. */ - if (pDevice->winmm.headerFramesConsumedCapture == (pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwBufferLength/bpf)) { - pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwUser = 1; /* 1 = locked. */ - pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwFlags &= ~WHDR_DONE; /* <-- Need to make sure the WHDR_DONE flag is unset. */ + if (pDevice->playback.usingDefaultChannels) { + if ((caps.dwFlags & MA_DSCAPS_PRIMARYSTEREO) != 0) { + DWORD speakerConfig; - /* Make sure the event is reset to a non-signaled state to ensure we don't prematurely return from WaitForSingleObject(). */ - ResetEvent((HANDLE)pDevice->winmm.hEventCapture); + /* It supports at least stereo, but could support more. */ + wf.Format.nChannels = 2; - /* The device will be started here. */ - resultMM = ((MA_PFN_waveInAddBuffer)pDevice->pContext->winmm.waveInAddBuffer)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((LPWAVEHDR)pDevice->winmm.pWAVEHDRCapture)[pDevice->winmm.iNextHeaderCapture], sizeof(WAVEHDR)); - if (resultMM != MMSYSERR_NOERROR) { - result = ma_result_from_MMRESULT(resultMM); - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WinMM] waveInAddBuffer() failed.", result); - break; + /* Look at the speaker configuration to get a better idea on the channel count. */ + if (SUCCEEDED(ma_IDirectSound_GetSpeakerConfig((ma_IDirectSound*)pDevice->dsound.pPlayback, &speakerConfig))) { + ma_get_channels_from_speaker_config__dsound(speakerConfig, &wf.Format.nChannels, &wf.dwChannelMask); } - - /* Make sure we move to the next header. */ - pDevice->winmm.iNextHeaderCapture = (pDevice->winmm.iNextHeaderCapture + 1) % pDevice->capture.internalPeriods; - pDevice->winmm.headerFramesConsumedCapture = 0; + } else { + /* It does not support stereo, which means we are stuck with mono. */ + wf.Format.nChannels = 1; } + } - /* If at this point we have filled the entire input buffer we can return. */ - ma_assert(totalFramesRead <= frameCount); - if (totalFramesRead == frameCount) { - break; + if (pDevice->usingDefaultSampleRate) { + /* We base the sample rate on the values returned by GetCaps(). */ + if ((caps.dwFlags & MA_DSCAPS_CONTINUOUSRATE) != 0) { + wf.Format.nSamplesPerSec = ma_get_best_sample_rate_within_range(caps.dwMinSecondarySampleRate, caps.dwMaxSecondarySampleRate); + } else { + wf.Format.nSamplesPerSec = caps.dwMaxSecondarySampleRate; } - - /* Getting here means there's more to process. */ - continue; } - /* Getting here means there isn't enough any data left to send to the client which means we need to wait for more. */ - if (WaitForSingleObject((HANDLE)pDevice->winmm.hEventCapture, INFINITE) != WAIT_OBJECT_0) { - result = MA_ERROR; - break; - } + wf.Format.nBlockAlign = (wf.Format.nChannels * wf.Format.wBitsPerSample) / 8; + wf.Format.nAvgBytesPerSec = wf.Format.nBlockAlign * wf.Format.nSamplesPerSec; - /* Something happened. If the next buffer has been marked as done we need to reset a bit of state. */ - if ((pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwFlags & WHDR_DONE) != 0) { - pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwUser = 0; /* 0 = unlocked (make it available for reading). */ - pDevice->winmm.headerFramesConsumedCapture = 0; + /* + From MSDN: + + The method succeeds even if the hardware does not support the requested format; DirectSound sets the buffer to the closest + supported format. To determine whether this has happened, an application can call the GetFormat method for the primary buffer + and compare the result with the format that was requested with the SetFormat method. + */ + if (FAILED(ma_IDirectSoundBuffer_SetFormat((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer, (WAVEFORMATEX*)&wf))) { + ma_device_uninit__dsound(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to set format of playback device's primary buffer.", MA_FORMAT_NOT_SUPPORTED); } - /* If the device has been stopped we need to break. */ - if (ma_device__get_state(pDevice) != MA_STATE_STARTED) { - break; + /* Get the _actual_ properties of the buffer. */ + pActualFormat = (WAVEFORMATEXTENSIBLE*)rawdata; + if (FAILED(ma_IDirectSoundBuffer_GetFormat((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer, (WAVEFORMATEX*)pActualFormat, sizeof(rawdata), NULL))) { + ma_device_uninit__dsound(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to retrieve the actual format of the playback device's primary buffer.", MA_FORMAT_NOT_SUPPORTED); } - } - if (pFramesRead != NULL) { - *pFramesRead = totalFramesRead; - } + pDevice->playback.internalFormat = ma_format_from_WAVEFORMATEX((WAVEFORMATEX*)pActualFormat); + pDevice->playback.internalChannels = pActualFormat->Format.nChannels; + pDevice->playback.internalSampleRate = pActualFormat->Format.nSamplesPerSec; - return result; -} + /* Get the internal channel map based on the channel mask. */ + if (pActualFormat->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE) { + ma_channel_mask_to_channel_map__win32(pActualFormat->dwChannelMask, pDevice->playback.internalChannels, pDevice->playback.internalChannelMap); + } else { + ma_channel_mask_to_channel_map__win32(wf.dwChannelMask, pDevice->playback.internalChannels, pDevice->playback.internalChannelMap); + } -ma_result ma_device_main_loop__winmm(ma_device* pDevice) -{ - ma_result result = MA_SUCCESS; - ma_bool32 exitLoop = MA_FALSE; - - ma_assert(pDevice != NULL); + /* The size of the buffer must be a clean multiple of the period count. */ + periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(periodSizeInMilliseconds, pDevice->playback.internalSampleRate); - /* The capture device needs to be started immediately. */ - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - MMRESULT resultMM; - WAVEHDR* pWAVEHDR; - ma_uint32 iPeriod; + /* + Meaning of dwFlags (from MSDN): + + DSBCAPS_CTRLPOSITIONNOTIFY + The buffer has position notification capability. + + DSBCAPS_GLOBALFOCUS + With this flag set, an application using DirectSound can continue to play its buffers if the user switches focus to + another application, even if the new application uses DirectSound. + + DSBCAPS_GETCURRENTPOSITION2 + In the first version of DirectSound, the play cursor was significantly ahead of the actual playing sound on emulated + sound cards; it was directly behind the write cursor. Now, if the DSBCAPS_GETCURRENTPOSITION2 flag is specified, the + application can get a more accurate play cursor. + */ + MA_ZERO_OBJECT(&descDS); + descDS.dwSize = sizeof(descDS); + descDS.dwFlags = MA_DSBCAPS_CTRLPOSITIONNOTIFY | MA_DSBCAPS_GLOBALFOCUS | MA_DSBCAPS_GETCURRENTPOSITION2; + descDS.dwBufferBytes = periodSizeInFrames * pConfig->periods * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + descDS.lpwfxFormat = (WAVEFORMATEX*)&wf; + if (FAILED(ma_IDirectSound_CreateSoundBuffer((ma_IDirectSound*)pDevice->dsound.pPlayback, &descDS, (ma_IDirectSoundBuffer**)&pDevice->dsound.pPlaybackBuffer, NULL))) { + ma_device_uninit__dsound(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_CreateSoundBuffer() failed for playback device's secondary buffer.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } + + /* DirectSound should give us a buffer exactly the size we asked for. */ + pDevice->playback.internalPeriodSizeInFrames = periodSizeInFrames; + pDevice->playback.internalPeriods = pConfig->periods; + } - pWAVEHDR = (WAVEHDR*)pDevice->winmm.pWAVEHDRCapture; + (void)pContext; + return MA_SUCCESS; +} - /* Make sure the event is reset to a non-signaled state to ensure we don't prematurely return from WaitForSingleObject(). */ - ResetEvent((HANDLE)pDevice->winmm.hEventCapture); - /* To start the device we attach all of the buffers and then start it. As the buffers are filled with data we will get notifications. */ - for (iPeriod = 0; iPeriod < pDevice->capture.internalPeriods; ++iPeriod) { - resultMM = ((MA_PFN_waveInAddBuffer)pDevice->pContext->winmm.waveInAddBuffer)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((LPWAVEHDR)pDevice->winmm.pWAVEHDRCapture)[iPeriod], sizeof(WAVEHDR)); - if (resultMM != MMSYSERR_NOERROR) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WinMM] Failed to attach input buffers to capture device in preparation for capture.", ma_result_from_MMRESULT(resultMM)); - } +static ma_result ma_device_main_loop__dsound(ma_device* pDevice) +{ + ma_result result = MA_SUCCESS; + ma_uint32 bpfDeviceCapture = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + ma_uint32 bpfDevicePlayback = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + HRESULT hr; + DWORD lockOffsetInBytesCapture; + DWORD lockSizeInBytesCapture; + DWORD mappedSizeInBytesCapture; + DWORD mappedDeviceFramesProcessedCapture; + void* pMappedDeviceBufferCapture; + DWORD lockOffsetInBytesPlayback; + DWORD lockSizeInBytesPlayback; + DWORD mappedSizeInBytesPlayback; + void* pMappedDeviceBufferPlayback; + DWORD prevReadCursorInBytesCapture = 0; + DWORD prevPlayCursorInBytesPlayback = 0; + ma_bool32 physicalPlayCursorLoopFlagPlayback = 0; + DWORD virtualWriteCursorInBytesPlayback = 0; + ma_bool32 virtualWriteCursorLoopFlagPlayback = 0; + ma_bool32 isPlaybackDeviceStarted = MA_FALSE; + ma_uint32 framesWrittenToPlaybackDevice = 0; /* For knowing whether or not the playback device needs to be started. */ + ma_uint32 waitTimeInMilliseconds = 1; - /* Make sure all of the buffers start out locked. We don't want to access them until the backend tells us we can. */ - pWAVEHDR[iPeriod].dwUser = 1; /* 1 = locked. */ - } + MA_ASSERT(pDevice != NULL); - /* Capture devices need to be explicitly started, unlike playback devices. */ - resultMM = ((MA_PFN_waveInStart)pDevice->pContext->winmm.waveInStart)((HWAVEIN)pDevice->winmm.hDeviceCapture); - if (resultMM != MMSYSERR_NOERROR) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WinMM] Failed to start backend device.", ma_result_from_MMRESULT(resultMM)); + /* The first thing to do is start the capture device. The playback device is only started after the first period is written. */ + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + if (FAILED(ma_IDirectSoundCaptureBuffer_Start((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, MA_DSCBSTART_LOOPING))) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCaptureBuffer_Start() failed.", MA_FAILED_TO_START_BACKEND_DEVICE); } } - - - while (ma_device__get_state(pDevice) == MA_STATE_STARTED && !exitLoop) { + + while (ma_device__get_state(pDevice) == MA_STATE_STARTED) { switch (pDevice->type) { case ma_device_type_duplex: { - /* The process is: device_read -> convert -> callback -> convert -> device_write */ - ma_uint8 capturedDeviceData[8192]; - ma_uint8 playbackDeviceData[8192]; - ma_uint32 capturedDeviceDataCapInFrames = sizeof(capturedDeviceData) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 playbackDeviceDataCapInFrames = sizeof(playbackDeviceData) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + DWORD physicalCaptureCursorInBytes; + DWORD physicalReadCursorInBytes; + if (FAILED(ma_IDirectSoundCaptureBuffer_GetCurrentPosition((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, &physicalCaptureCursorInBytes, &physicalReadCursorInBytes))) { + return MA_ERROR; + } - ma_uint32 totalFramesProcessed = 0; - ma_uint32 periodSizeInFrames = ma_min(pDevice->capture.internalBufferSizeInFrames/pDevice->capture.internalPeriods, pDevice->playback.internalBufferSizeInFrames/pDevice->playback.internalPeriods); - - while (totalFramesProcessed < periodSizeInFrames) { - ma_uint32 framesRemaining = periodSizeInFrames - totalFramesProcessed; - ma_uint32 framesProcessed; - ma_uint32 framesToProcess = framesRemaining; - if (framesToProcess > capturedDeviceDataCapInFrames) { - framesToProcess = capturedDeviceDataCapInFrames; + /* If nothing is available we just sleep for a bit and return from this iteration. */ + if (physicalReadCursorInBytes == prevReadCursorInBytesCapture) { + ma_sleep(waitTimeInMilliseconds); + continue; /* Nothing is available in the capture buffer. */ + } + + /* + The current position has moved. We need to map all of the captured samples and write them to the playback device, making sure + we don't return until every frame has been copied over. + */ + if (prevReadCursorInBytesCapture < physicalReadCursorInBytes) { + /* The capture position has not looped. This is the simple case. */ + lockOffsetInBytesCapture = prevReadCursorInBytesCapture; + lockSizeInBytesCapture = (physicalReadCursorInBytes - prevReadCursorInBytesCapture); + } else { + /* + The capture position has looped. This is the more complex case. Map to the end of the buffer. If this does not return anything, + do it again from the start. + */ + if (prevReadCursorInBytesCapture < pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) { + /* Lock up to the end of the buffer. */ + lockOffsetInBytesCapture = prevReadCursorInBytesCapture; + lockSizeInBytesCapture = (pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) - prevReadCursorInBytesCapture; + } else { + /* Lock starting from the start of the buffer. */ + lockOffsetInBytesCapture = 0; + lockSizeInBytesCapture = physicalReadCursorInBytes; } + } + + if (lockSizeInBytesCapture == 0) { + ma_sleep(waitTimeInMilliseconds); + continue; /* Nothing is available in the capture buffer. */ + } + + hr = ma_IDirectSoundCaptureBuffer_Lock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, lockOffsetInBytesCapture, lockSizeInBytesCapture, &pMappedDeviceBufferCapture, &mappedSizeInBytesCapture, NULL, NULL, 0); + if (FAILED(hr)) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from capture device in preparation for writing to the device.", MA_FAILED_TO_MAP_DEVICE_BUFFER); + } + + + /* At this point we have some input data that we need to output. We do not return until every mapped frame of the input data is written to the playback device. */ + mappedDeviceFramesProcessedCapture = 0; - result = ma_device_read__winmm(pDevice, capturedDeviceData, framesToProcess, &framesProcessed); + for (;;) { /* Keep writing to the playback device. */ + ma_uint8 inputFramesInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 inputFramesInClientFormatCap = sizeof(inputFramesInClientFormat) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); + ma_uint8 outputFramesInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 outputFramesInClientFormatCap = sizeof(outputFramesInClientFormat) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); + ma_uint32 outputFramesInClientFormatCount; + ma_uint32 outputFramesInClientFormatConsumed = 0; + ma_uint64 clientCapturedFramesToProcess = ma_min(inputFramesInClientFormatCap, outputFramesInClientFormatCap); + ma_uint64 deviceCapturedFramesToProcess = (mappedSizeInBytesCapture / bpfDeviceCapture) - mappedDeviceFramesProcessedCapture; + void* pRunningMappedDeviceBufferCapture = ma_offset_ptr(pMappedDeviceBufferCapture, mappedDeviceFramesProcessedCapture * bpfDeviceCapture); + + result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningMappedDeviceBufferCapture, &deviceCapturedFramesToProcess, inputFramesInClientFormat, &clientCapturedFramesToProcess); if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; break; } - pDevice->capture._dspFrameCount = framesToProcess; - pDevice->capture._dspFrames = capturedDeviceData; + outputFramesInClientFormatCount = (ma_uint32)clientCapturedFramesToProcess; + mappedDeviceFramesProcessedCapture += (ma_uint32)deviceCapturedFramesToProcess; + + ma_device__on_data(pDevice, outputFramesInClientFormat, inputFramesInClientFormat, (ma_uint32)clientCapturedFramesToProcess); + /* At this point we have input and output data in client format. All we need to do now is convert it to the output device format. This may take a few passes. */ for (;;) { - ma_uint8 capturedData[8192]; - ma_uint8 playbackData[8192]; - ma_uint32 capturedDataCapInFrames = sizeof(capturedData) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); - ma_uint32 playbackDataCapInFrames = sizeof(playbackData) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); - - ma_uint32 capturedFramesToTryProcessing = ma_min(capturedDataCapInFrames, playbackDataCapInFrames); - ma_uint32 capturedFramesToProcess = (ma_uint32)ma_pcm_converter_read(&pDevice->capture.converter, capturedData, capturedFramesToTryProcessing); - if (capturedFramesToProcess == 0) { - break; /* Don't fire the data callback with zero frames. */ + ma_uint32 framesWrittenThisIteration; + DWORD physicalPlayCursorInBytes; + DWORD physicalWriteCursorInBytes; + DWORD availableBytesPlayback; + DWORD silentPaddingInBytes = 0; /* <-- Must be initialized to 0. */ + + /* We need the physical play and write cursors. */ + if (FAILED(ma_IDirectSoundBuffer_GetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &physicalPlayCursorInBytes, &physicalWriteCursorInBytes))) { + break; } - ma_device__on_data(pDevice, playbackData, capturedData, capturedFramesToProcess); + if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) { + physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback; + } + prevPlayCursorInBytesPlayback = physicalPlayCursorInBytes; - /* At this point the playbackData buffer should be holding data that needs to be written to the device. */ - pDevice->playback._dspFrameCount = capturedFramesToProcess; - pDevice->playback._dspFrames = playbackData; - for (;;) { - ma_uint32 playbackDeviceFramesCount = (ma_uint32)ma_pcm_converter_read(&pDevice->playback.converter, playbackDeviceData, playbackDeviceDataCapInFrames); - if (playbackDeviceFramesCount == 0) { - break; + /* If there's any bytes available for writing we can do that now. The space between the virtual cursor position and play cursor. */ + if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) { + /* Same loop iteration. The available bytes wraps all the way around from the virtual write cursor to the physical play cursor. */ + if (physicalPlayCursorInBytes <= virtualWriteCursorInBytesPlayback) { + availableBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback; + availableBytesPlayback += physicalPlayCursorInBytes; /* Wrap around. */ + } else { + /* This is an error. */ + #ifdef MA_DEBUG_OUTPUT + printf("[DirectSound] (Duplex/Playback) WARNING: Play cursor has moved in front of the write cursor (same loop iterations). physicalPlayCursorInBytes=%d, virtualWriteCursorInBytes=%d.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback); + #endif + availableBytesPlayback = 0; + } + } else { + /* Different loop iterations. The available bytes only goes from the virtual write cursor to the physical play cursor. */ + if (physicalPlayCursorInBytes >= virtualWriteCursorInBytesPlayback) { + availableBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback; + } else { + /* This is an error. */ + #ifdef MA_DEBUG_OUTPUT + printf("[DirectSound] (Duplex/Playback) WARNING: Write cursor has moved behind the play cursor (different loop iterations). physicalPlayCursorInBytes=%d, virtualWriteCursorInBytes=%d.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback); + #endif + availableBytesPlayback = 0; } + } - result = ma_device_write__winmm(pDevice, playbackDeviceData, playbackDeviceFramesCount, NULL); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; + #ifdef MA_DEBUG_OUTPUT + /*printf("[DirectSound] (Duplex/Playback) physicalPlayCursorInBytes=%d, availableBytesPlayback=%d\n", physicalPlayCursorInBytes, availableBytesPlayback);*/ + #endif + + /* If there's no room available for writing we need to wait for more. */ + if (availableBytesPlayback == 0) { + /* If we haven't started the device yet, this will never get beyond 0. In this case we need to get the device started. */ + if (!isPlaybackDeviceStarted) { + if (FAILED(ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING))) { + ma_IDirectSoundCaptureBuffer_Stop((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.", MA_FAILED_TO_START_BACKEND_DEVICE); + } + isPlaybackDeviceStarted = MA_TRUE; + } else { + ma_sleep(waitTimeInMilliseconds); + continue; + } + } + + + /* Getting here means there room available somewhere. We limit this to either the end of the buffer or the physical play cursor, whichever is closest. */ + lockOffsetInBytesPlayback = virtualWriteCursorInBytesPlayback; + if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) { + /* Same loop iteration. Go up to the end of the buffer. */ + lockSizeInBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback; + } else { + /* Different loop iterations. Go up to the physical play cursor. */ + lockSizeInBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback; + } + + hr = ma_IDirectSoundBuffer_Lock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, lockOffsetInBytesPlayback, lockSizeInBytesPlayback, &pMappedDeviceBufferPlayback, &mappedSizeInBytesPlayback, NULL, NULL, 0); + if (FAILED(hr)) { + result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from playback device in preparation for writing to the device.", MA_FAILED_TO_MAP_DEVICE_BUFFER); + break; + } + + /* + Experiment: If the playback buffer is being starved, pad it with some silence to get it back in sync. This will cause a glitch, but it may prevent + endless glitching due to it constantly running out of data. + */ + if (isPlaybackDeviceStarted) { + DWORD bytesQueuedForPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - availableBytesPlayback; + if (bytesQueuedForPlayback < (pDevice->playback.internalPeriodSizeInFrames*bpfDevicePlayback)) { + silentPaddingInBytes = (pDevice->playback.internalPeriodSizeInFrames*2*bpfDevicePlayback) - bytesQueuedForPlayback; + if (silentPaddingInBytes > lockSizeInBytesPlayback) { + silentPaddingInBytes = lockSizeInBytesPlayback; + } + + #ifdef MA_DEBUG_OUTPUT + printf("[DirectSound] (Duplex/Playback) Playback buffer starved. availableBytesPlayback=%d, silentPaddingInBytes=%d\n", availableBytesPlayback, silentPaddingInBytes); + #endif } + } - if (playbackDeviceFramesCount < playbackDeviceDataCapInFrames) { + /* At this point we have a buffer for output. */ + if (silentPaddingInBytes > 0) { + MA_ZERO_MEMORY(pMappedDeviceBufferPlayback, silentPaddingInBytes); + framesWrittenThisIteration = silentPaddingInBytes/bpfDevicePlayback; + } else { + ma_uint64 convertedFrameCountIn = (outputFramesInClientFormatCount - outputFramesInClientFormatConsumed); + ma_uint64 convertedFrameCountOut = mappedSizeInBytesPlayback/bpfDevicePlayback; + void* pConvertedFramesIn = ma_offset_ptr(outputFramesInClientFormat, outputFramesInClientFormatConsumed * bpfDevicePlayback); + void* pConvertedFramesOut = pMappedDeviceBufferPlayback; + + result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, pConvertedFramesIn, &convertedFrameCountIn, pConvertedFramesOut, &convertedFrameCountOut); + if (result != MA_SUCCESS) { break; } + + outputFramesInClientFormatConsumed += (ma_uint32)convertedFrameCountOut; + framesWrittenThisIteration = (ma_uint32)convertedFrameCountOut; } + - if (capturedFramesToProcess < capturedFramesToTryProcessing) { + hr = ma_IDirectSoundBuffer_Unlock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, pMappedDeviceBufferPlayback, framesWrittenThisIteration*bpfDevicePlayback, NULL, 0); + if (FAILED(hr)) { + result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from playback device after writing to the device.", MA_FAILED_TO_UNMAP_DEVICE_BUFFER); break; } - /* In case an error happened from ma_device_write2__alsa()... */ - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; + virtualWriteCursorInBytesPlayback += framesWrittenThisIteration*bpfDevicePlayback; + if ((virtualWriteCursorInBytesPlayback/bpfDevicePlayback) == pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods) { + virtualWriteCursorInBytesPlayback = 0; + virtualWriteCursorLoopFlagPlayback = !virtualWriteCursorLoopFlagPlayback; + } + + /* + We may need to start the device. We want two full periods to be written before starting the playback device. Having an extra period adds + a bit of a buffer to prevent the playback buffer from getting starved. + */ + framesWrittenToPlaybackDevice += framesWrittenThisIteration; + if (!isPlaybackDeviceStarted && framesWrittenToPlaybackDevice >= (pDevice->playback.internalPeriodSizeInFrames*2)) { + if (FAILED(ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING))) { + ma_IDirectSoundCaptureBuffer_Stop((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.", MA_FAILED_TO_START_BACKEND_DEVICE); + } + isPlaybackDeviceStarted = MA_TRUE; + } + + if (framesWrittenThisIteration < mappedSizeInBytesPlayback/bpfDevicePlayback) { + break; /* We're finished with the output data.*/ } } - - totalFramesProcessed += framesProcessed; + + if (clientCapturedFramesToProcess == 0) { + break; /* We just consumed every input sample. */ + } + } + + + /* At this point we're done with the mapped portion of the capture buffer. */ + hr = ma_IDirectSoundCaptureBuffer_Unlock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, pMappedDeviceBufferCapture, mappedSizeInBytesCapture, NULL, 0); + if (FAILED(hr)) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from capture device after reading from the device.", MA_FAILED_TO_UNMAP_DEVICE_BUFFER); + } + prevReadCursorInBytesCapture = (lockOffsetInBytesCapture + mappedSizeInBytesCapture); + } break; + + + + case ma_device_type_capture: + { + DWORD physicalCaptureCursorInBytes; + DWORD physicalReadCursorInBytes; + if (FAILED(ma_IDirectSoundCaptureBuffer_GetCurrentPosition((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, &physicalCaptureCursorInBytes, &physicalReadCursorInBytes))) { + return MA_ERROR; + } + + /* If the previous capture position is the same as the current position we need to wait a bit longer. */ + if (prevReadCursorInBytesCapture == physicalReadCursorInBytes) { + ma_sleep(waitTimeInMilliseconds); + continue; + } + + /* Getting here means we have capture data available. */ + if (prevReadCursorInBytesCapture < physicalReadCursorInBytes) { + /* The capture position has not looped. This is the simple case. */ + lockOffsetInBytesCapture = prevReadCursorInBytesCapture; + lockSizeInBytesCapture = (physicalReadCursorInBytes - prevReadCursorInBytesCapture); + } else { + /* + The capture position has looped. This is the more complex case. Map to the end of the buffer. If this does not return anything, + do it again from the start. + */ + if (prevReadCursorInBytesCapture < pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) { + /* Lock up to the end of the buffer. */ + lockOffsetInBytesCapture = prevReadCursorInBytesCapture; + lockSizeInBytesCapture = (pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) - prevReadCursorInBytesCapture; + } else { + /* Lock starting from the start of the buffer. */ + lockOffsetInBytesCapture = 0; + lockSizeInBytesCapture = physicalReadCursorInBytes; + } } - } break; - - case ma_device_type_capture: - { - /* We read in chunks of the period size, but use a stack allocated buffer for the intermediary. */ - ma_uint8 intermediaryBuffer[8192]; - ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 periodSizeInFrames = pDevice->capture.internalBufferSizeInFrames / pDevice->capture.internalPeriods; - ma_uint32 framesReadThisPeriod = 0; - while (framesReadThisPeriod < periodSizeInFrames) { - ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesReadThisPeriod; - ma_uint32 framesProcessed; - ma_uint32 framesToReadThisIteration = framesRemainingInPeriod; - if (framesToReadThisIteration > intermediaryBufferSizeInFrames) { - framesToReadThisIteration = intermediaryBufferSizeInFrames; - } - result = ma_device_read__winmm(pDevice, intermediaryBuffer, framesToReadThisIteration, &framesProcessed); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } + #ifdef MA_DEBUG_OUTPUT + /*printf("[DirectSound] (Capture) physicalCaptureCursorInBytes=%d, physicalReadCursorInBytes=%d\n", physicalCaptureCursorInBytes, physicalReadCursorInBytes);*/ + /*printf("[DirectSound] (Capture) lockOffsetInBytesCapture=%d, lockSizeInBytesCapture=%d\n", lockOffsetInBytesCapture, lockSizeInBytesCapture);*/ + #endif - ma_device__send_frames_to_client(pDevice, framesProcessed, intermediaryBuffer); + if (lockSizeInBytesCapture < pDevice->capture.internalPeriodSizeInFrames) { + ma_sleep(waitTimeInMilliseconds); + continue; /* Nothing is available in the capture buffer. */ + } - framesReadThisPeriod += framesProcessed; + hr = ma_IDirectSoundCaptureBuffer_Lock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, lockOffsetInBytesCapture, lockSizeInBytesCapture, &pMappedDeviceBufferCapture, &mappedSizeInBytesCapture, NULL, NULL, 0); + if (FAILED(hr)) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from capture device in preparation for writing to the device.", MA_FAILED_TO_MAP_DEVICE_BUFFER); } - } break; - case ma_device_type_playback: - { - /* We write in chunks of the period size, but use a stack allocated buffer for the intermediary. */ - ma_uint8 intermediaryBuffer[8192]; - ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - ma_uint32 periodSizeInFrames = pDevice->playback.internalBufferSizeInFrames / pDevice->playback.internalPeriods; - ma_uint32 framesWrittenThisPeriod = 0; - while (framesWrittenThisPeriod < periodSizeInFrames) { - ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesWrittenThisPeriod; - ma_uint32 framesProcessed; - ma_uint32 framesToWriteThisIteration = framesRemainingInPeriod; - if (framesToWriteThisIteration > intermediaryBufferSizeInFrames) { - framesToWriteThisIteration = intermediaryBufferSizeInFrames; - } + #ifdef MA_DEBUG_OUTPUT + if (lockSizeInBytesCapture != mappedSizeInBytesCapture) { + printf("[DirectSound] (Capture) lockSizeInBytesCapture=%d != mappedSizeInBytesCapture=%d\n", lockSizeInBytesCapture, mappedSizeInBytesCapture); + } + #endif - ma_device__read_frames_from_client(pDevice, framesToWriteThisIteration, intermediaryBuffer); + ma_device__send_frames_to_client(pDevice, mappedSizeInBytesCapture/bpfDeviceCapture, pMappedDeviceBufferCapture); - result = ma_device_write__winmm(pDevice, intermediaryBuffer, framesToWriteThisIteration, &framesProcessed); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } + hr = ma_IDirectSoundCaptureBuffer_Unlock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, pMappedDeviceBufferCapture, mappedSizeInBytesCapture, NULL, 0); + if (FAILED(hr)) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from capture device after reading from the device.", MA_FAILED_TO_UNMAP_DEVICE_BUFFER); + } + prevReadCursorInBytesCapture = lockOffsetInBytesCapture + mappedSizeInBytesCapture; - framesWrittenThisPeriod += framesProcessed; + if (prevReadCursorInBytesCapture == (pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture)) { + prevReadCursorInBytesCapture = 0; } } break; - /* To silence a warning. Will never hit this. */ - case ma_device_type_loopback: - default: break; - } - } - /* Here is where the device is started. */ - ma_device_stop__winmm(pDevice); + case ma_device_type_playback: + { + DWORD availableBytesPlayback; + DWORD physicalPlayCursorInBytes; + DWORD physicalWriteCursorInBytes; + if (FAILED(ma_IDirectSoundBuffer_GetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &physicalPlayCursorInBytes, &physicalWriteCursorInBytes))) { + break; + } - return result; -} + if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) { + physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback; + } + prevPlayCursorInBytesPlayback = physicalPlayCursorInBytes; -ma_result ma_context_uninit__winmm(ma_context* pContext) -{ - ma_assert(pContext != NULL); - ma_assert(pContext->backend == ma_backend_winmm); + /* If there's any bytes available for writing we can do that now. The space between the virtual cursor position and play cursor. */ + if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) { + /* Same loop iteration. The available bytes wraps all the way around from the virtual write cursor to the physical play cursor. */ + if (physicalPlayCursorInBytes <= virtualWriteCursorInBytesPlayback) { + availableBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback; + availableBytesPlayback += physicalPlayCursorInBytes; /* Wrap around. */ + } else { + /* This is an error. */ + #ifdef MA_DEBUG_OUTPUT + printf("[DirectSound] (Playback) WARNING: Play cursor has moved in front of the write cursor (same loop iterations). physicalPlayCursorInBytes=%d, virtualWriteCursorInBytes=%d.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback); + #endif + availableBytesPlayback = 0; + } + } else { + /* Different loop iterations. The available bytes only goes from the virtual write cursor to the physical play cursor. */ + if (physicalPlayCursorInBytes >= virtualWriteCursorInBytesPlayback) { + availableBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback; + } else { + /* This is an error. */ + #ifdef MA_DEBUG_OUTPUT + printf("[DirectSound] (Playback) WARNING: Write cursor has moved behind the play cursor (different loop iterations). physicalPlayCursorInBytes=%d, virtualWriteCursorInBytes=%d.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback); + #endif + availableBytesPlayback = 0; + } + } - ma_dlclose(pContext, pContext->winmm.hWinMM); - return MA_SUCCESS; -} + #ifdef MA_DEBUG_OUTPUT + /*printf("[DirectSound] (Playback) physicalPlayCursorInBytes=%d, availableBytesPlayback=%d\n", physicalPlayCursorInBytes, availableBytesPlayback);*/ + #endif -ma_result ma_context_init__winmm(const ma_context_config* pConfig, ma_context* pContext) -{ - ma_assert(pContext != NULL); + /* If there's no room available for writing we need to wait for more. */ + if (availableBytesPlayback < pDevice->playback.internalPeriodSizeInFrames) { + /* If we haven't started the device yet, this will never get beyond 0. In this case we need to get the device started. */ + if (availableBytesPlayback == 0 && !isPlaybackDeviceStarted) { + if (FAILED(ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING))) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.", MA_FAILED_TO_START_BACKEND_DEVICE); + } + isPlaybackDeviceStarted = MA_TRUE; + } else { + ma_sleep(waitTimeInMilliseconds); + continue; + } + } - (void)pConfig; + /* Getting here means there room available somewhere. We limit this to either the end of the buffer or the physical play cursor, whichever is closest. */ + lockOffsetInBytesPlayback = virtualWriteCursorInBytesPlayback; + if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) { + /* Same loop iteration. Go up to the end of the buffer. */ + lockSizeInBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback; + } else { + /* Different loop iterations. Go up to the physical play cursor. */ + lockSizeInBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback; + } - pContext->winmm.hWinMM = ma_dlopen(pContext, "winmm.dll"); - if (pContext->winmm.hWinMM == NULL) { - return MA_NO_BACKEND; - } + hr = ma_IDirectSoundBuffer_Lock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, lockOffsetInBytesPlayback, lockSizeInBytesPlayback, &pMappedDeviceBufferPlayback, &mappedSizeInBytesPlayback, NULL, NULL, 0); + if (FAILED(hr)) { + result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from playback device in preparation for writing to the device.", MA_FAILED_TO_MAP_DEVICE_BUFFER); + break; + } - pContext->winmm.waveOutGetNumDevs = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutGetNumDevs"); - pContext->winmm.waveOutGetDevCapsA = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutGetDevCapsA"); - pContext->winmm.waveOutOpen = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutOpen"); - pContext->winmm.waveOutClose = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutClose"); - pContext->winmm.waveOutPrepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutPrepareHeader"); - pContext->winmm.waveOutUnprepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutUnprepareHeader"); - pContext->winmm.waveOutWrite = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutWrite"); - pContext->winmm.waveOutReset = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutReset"); - pContext->winmm.waveInGetNumDevs = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInGetNumDevs"); - pContext->winmm.waveInGetDevCapsA = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInGetDevCapsA"); - pContext->winmm.waveInOpen = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInOpen"); - pContext->winmm.waveInClose = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInClose"); - pContext->winmm.waveInPrepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInPrepareHeader"); - pContext->winmm.waveInUnprepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInUnprepareHeader"); - pContext->winmm.waveInAddBuffer = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInAddBuffer"); - pContext->winmm.waveInStart = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInStart"); - pContext->winmm.waveInReset = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInReset"); + /* At this point we have a buffer for output. */ + ma_device__read_frames_from_client(pDevice, (mappedSizeInBytesPlayback/bpfDevicePlayback), pMappedDeviceBufferPlayback); - pContext->onUninit = ma_context_uninit__winmm; - pContext->onDeviceIDEqual = ma_context_is_device_id_equal__winmm; - pContext->onEnumDevices = ma_context_enumerate_devices__winmm; - pContext->onGetDeviceInfo = ma_context_get_device_info__winmm; - pContext->onDeviceInit = ma_device_init__winmm; - pContext->onDeviceUninit = ma_device_uninit__winmm; - pContext->onDeviceStart = NULL; /* Not used with synchronous backends. */ - pContext->onDeviceStop = NULL; /* Not used with synchronous backends. */ - pContext->onDeviceMainLoop = ma_device_main_loop__winmm; + hr = ma_IDirectSoundBuffer_Unlock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, pMappedDeviceBufferPlayback, mappedSizeInBytesPlayback, NULL, 0); + if (FAILED(hr)) { + result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from playback device after writing to the device.", MA_FAILED_TO_UNMAP_DEVICE_BUFFER); + break; + } - return MA_SUCCESS; -} -#endif + virtualWriteCursorInBytesPlayback += mappedSizeInBytesPlayback; + if (virtualWriteCursorInBytesPlayback == pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) { + virtualWriteCursorInBytesPlayback = 0; + virtualWriteCursorLoopFlagPlayback = !virtualWriteCursorLoopFlagPlayback; + } + + /* + We may need to start the device. We want two full periods to be written before starting the playback device. Having an extra period adds + a bit of a buffer to prevent the playback buffer from getting starved. + */ + framesWrittenToPlaybackDevice += mappedSizeInBytesPlayback/bpfDevicePlayback; + if (!isPlaybackDeviceStarted && framesWrittenToPlaybackDevice >= pDevice->playback.internalPeriodSizeInFrames) { + if (FAILED(ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING))) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.", MA_FAILED_TO_START_BACKEND_DEVICE); + } + isPlaybackDeviceStarted = MA_TRUE; + } + } break; + default: return MA_INVALID_ARGS; /* Invalid device type. */ + } + if (result != MA_SUCCESS) { + return result; + } + } -/****************************************************************************** + /* Getting here means the device is being stopped. */ + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + if (FAILED(ma_IDirectSoundCaptureBuffer_Stop((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer))) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCaptureBuffer_Stop() failed.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + } + } -ALSA Backend + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + /* The playback device should be drained before stopping. All we do is wait until the available bytes is equal to the size of the buffer. */ + if (isPlaybackDeviceStarted) { + for (;;) { + DWORD availableBytesPlayback = 0; + DWORD physicalPlayCursorInBytes; + DWORD physicalWriteCursorInBytes; + if (FAILED(ma_IDirectSoundBuffer_GetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &physicalPlayCursorInBytes, &physicalWriteCursorInBytes))) { + break; + } -******************************************************************************/ -#ifdef MA_HAS_ALSA + if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) { + physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback; + } + prevPlayCursorInBytesPlayback = physicalPlayCursorInBytes; -#ifdef MA_NO_RUNTIME_LINKING -#include -typedef snd_pcm_uframes_t ma_snd_pcm_uframes_t; -typedef snd_pcm_sframes_t ma_snd_pcm_sframes_t; -typedef snd_pcm_stream_t ma_snd_pcm_stream_t; -typedef snd_pcm_format_t ma_snd_pcm_format_t; -typedef snd_pcm_access_t ma_snd_pcm_access_t; -typedef snd_pcm_t ma_snd_pcm_t; -typedef snd_pcm_hw_params_t ma_snd_pcm_hw_params_t; -typedef snd_pcm_sw_params_t ma_snd_pcm_sw_params_t; -typedef snd_pcm_format_mask_t ma_snd_pcm_format_mask_t; -typedef snd_pcm_info_t ma_snd_pcm_info_t; -typedef snd_pcm_channel_area_t ma_snd_pcm_channel_area_t; -typedef snd_pcm_chmap_t ma_snd_pcm_chmap_t; + if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) { + /* Same loop iteration. The available bytes wraps all the way around from the virtual write cursor to the physical play cursor. */ + if (physicalPlayCursorInBytes <= virtualWriteCursorInBytesPlayback) { + availableBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback; + availableBytesPlayback += physicalPlayCursorInBytes; /* Wrap around. */ + } else { + break; + } + } else { + /* Different loop iterations. The available bytes only goes from the virtual write cursor to the physical play cursor. */ + if (physicalPlayCursorInBytes >= virtualWriteCursorInBytesPlayback) { + availableBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback; + } else { + break; + } + } -/* snd_pcm_stream_t */ -#define MA_SND_PCM_STREAM_PLAYBACK SND_PCM_STREAM_PLAYBACK -#define MA_SND_PCM_STREAM_CAPTURE SND_PCM_STREAM_CAPTURE + if (availableBytesPlayback >= (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback)) { + break; + } -/* snd_pcm_format_t */ -#define MA_SND_PCM_FORMAT_UNKNOWN SND_PCM_FORMAT_UNKNOWN -#define MA_SND_PCM_FORMAT_U8 SND_PCM_FORMAT_U8 -#define MA_SND_PCM_FORMAT_S16_LE SND_PCM_FORMAT_S16_LE -#define MA_SND_PCM_FORMAT_S16_BE SND_PCM_FORMAT_S16_BE -#define MA_SND_PCM_FORMAT_S24_LE SND_PCM_FORMAT_S24_LE -#define MA_SND_PCM_FORMAT_S24_BE SND_PCM_FORMAT_S24_BE -#define MA_SND_PCM_FORMAT_S32_LE SND_PCM_FORMAT_S32_LE -#define MA_SND_PCM_FORMAT_S32_BE SND_PCM_FORMAT_S32_BE -#define MA_SND_PCM_FORMAT_FLOAT_LE SND_PCM_FORMAT_FLOAT_LE -#define MA_SND_PCM_FORMAT_FLOAT_BE SND_PCM_FORMAT_FLOAT_BE -#define MA_SND_PCM_FORMAT_FLOAT64_LE SND_PCM_FORMAT_FLOAT64_LE -#define MA_SND_PCM_FORMAT_FLOAT64_BE SND_PCM_FORMAT_FLOAT64_BE -#define MA_SND_PCM_FORMAT_MU_LAW SND_PCM_FORMAT_MU_LAW -#define MA_SND_PCM_FORMAT_A_LAW SND_PCM_FORMAT_A_LAW -#define MA_SND_PCM_FORMAT_S24_3LE SND_PCM_FORMAT_S24_3LE -#define MA_SND_PCM_FORMAT_S24_3BE SND_PCM_FORMAT_S24_3BE + ma_sleep(waitTimeInMilliseconds); + } + } -/* ma_snd_pcm_access_t */ -#define MA_SND_PCM_ACCESS_MMAP_INTERLEAVED SND_PCM_ACCESS_MMAP_INTERLEAVED -#define MA_SND_PCM_ACCESS_MMAP_NONINTERLEAVED SND_PCM_ACCESS_MMAP_NONINTERLEAVED -#define MA_SND_PCM_ACCESS_MMAP_COMPLEX SND_PCM_ACCESS_MMAP_COMPLEX -#define MA_SND_PCM_ACCESS_RW_INTERLEAVED SND_PCM_ACCESS_RW_INTERLEAVED -#define MA_SND_PCM_ACCESS_RW_NONINTERLEAVED SND_PCM_ACCESS_RW_NONINTERLEAVED + if (FAILED(ma_IDirectSoundBuffer_Stop((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer))) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Stop() failed.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + } -/* Channel positions. */ -#define MA_SND_CHMAP_UNKNOWN SND_CHMAP_UNKNOWN -#define MA_SND_CHMAP_NA SND_CHMAP_NA -#define MA_SND_CHMAP_MONO SND_CHMAP_MONO -#define MA_SND_CHMAP_FL SND_CHMAP_FL -#define MA_SND_CHMAP_FR SND_CHMAP_FR -#define MA_SND_CHMAP_RL SND_CHMAP_RL -#define MA_SND_CHMAP_RR SND_CHMAP_RR -#define MA_SND_CHMAP_FC SND_CHMAP_FC -#define MA_SND_CHMAP_LFE SND_CHMAP_LFE -#define MA_SND_CHMAP_SL SND_CHMAP_SL -#define MA_SND_CHMAP_SR SND_CHMAP_SR -#define MA_SND_CHMAP_RC SND_CHMAP_RC -#define MA_SND_CHMAP_FLC SND_CHMAP_FLC -#define MA_SND_CHMAP_FRC SND_CHMAP_FRC -#define MA_SND_CHMAP_RLC SND_CHMAP_RLC -#define MA_SND_CHMAP_RRC SND_CHMAP_RRC -#define MA_SND_CHMAP_FLW SND_CHMAP_FLW -#define MA_SND_CHMAP_FRW SND_CHMAP_FRW -#define MA_SND_CHMAP_FLH SND_CHMAP_FLH -#define MA_SND_CHMAP_FCH SND_CHMAP_FCH -#define MA_SND_CHMAP_FRH SND_CHMAP_FRH -#define MA_SND_CHMAP_TC SND_CHMAP_TC -#define MA_SND_CHMAP_TFL SND_CHMAP_TFL -#define MA_SND_CHMAP_TFR SND_CHMAP_TFR -#define MA_SND_CHMAP_TFC SND_CHMAP_TFC -#define MA_SND_CHMAP_TRL SND_CHMAP_TRL -#define MA_SND_CHMAP_TRR SND_CHMAP_TRR -#define MA_SND_CHMAP_TRC SND_CHMAP_TRC -#define MA_SND_CHMAP_TFLC SND_CHMAP_TFLC -#define MA_SND_CHMAP_TFRC SND_CHMAP_TFRC -#define MA_SND_CHMAP_TSL SND_CHMAP_TSL -#define MA_SND_CHMAP_TSR SND_CHMAP_TSR -#define MA_SND_CHMAP_LLFE SND_CHMAP_LLFE -#define MA_SND_CHMAP_RLFE SND_CHMAP_RLFE -#define MA_SND_CHMAP_BC SND_CHMAP_BC -#define MA_SND_CHMAP_BLC SND_CHMAP_BLC -#define MA_SND_CHMAP_BRC SND_CHMAP_BRC + ma_IDirectSoundBuffer_SetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0); + } -/* Open mode flags. */ -#define MA_SND_PCM_NO_AUTO_RESAMPLE SND_PCM_NO_AUTO_RESAMPLE -#define MA_SND_PCM_NO_AUTO_CHANNELS SND_PCM_NO_AUTO_CHANNELS -#define MA_SND_PCM_NO_AUTO_FORMAT SND_PCM_NO_AUTO_FORMAT -#else -#include /* For EPIPE, etc. */ -typedef unsigned long ma_snd_pcm_uframes_t; -typedef long ma_snd_pcm_sframes_t; -typedef int ma_snd_pcm_stream_t; -typedef int ma_snd_pcm_format_t; -typedef int ma_snd_pcm_access_t; -typedef struct ma_snd_pcm_t ma_snd_pcm_t; -typedef struct ma_snd_pcm_hw_params_t ma_snd_pcm_hw_params_t; -typedef struct ma_snd_pcm_sw_params_t ma_snd_pcm_sw_params_t; -typedef struct ma_snd_pcm_format_mask_t ma_snd_pcm_format_mask_t; -typedef struct ma_snd_pcm_info_t ma_snd_pcm_info_t; -typedef struct -{ - void* addr; - unsigned int first; - unsigned int step; -} ma_snd_pcm_channel_area_t; -typedef struct + return MA_SUCCESS; +} + +static ma_result ma_context_uninit__dsound(ma_context* pContext) { - unsigned int channels; - unsigned int pos[1]; -} ma_snd_pcm_chmap_t; + MA_ASSERT(pContext != NULL); + MA_ASSERT(pContext->backend == ma_backend_dsound); -/* snd_pcm_state_t */ -#define MA_SND_PCM_STATE_OPEN 0 -#define MA_SND_PCM_STATE_SETUP 1 -#define MA_SND_PCM_STATE_PREPARED 2 -#define MA_SND_PCM_STATE_RUNNING 3 -#define MA_SND_PCM_STATE_XRUN 4 -#define MA_SND_PCM_STATE_DRAINING 5 -#define MA_SND_PCM_STATE_PAUSED 6 -#define MA_SND_PCM_STATE_SUSPENDED 7 -#define MA_SND_PCM_STATE_DISCONNECTED 8 + ma_dlclose(pContext, pContext->dsound.hDSoundDLL); -/* snd_pcm_stream_t */ -#define MA_SND_PCM_STREAM_PLAYBACK 0 -#define MA_SND_PCM_STREAM_CAPTURE 1 + return MA_SUCCESS; +} -/* snd_pcm_format_t */ -#define MA_SND_PCM_FORMAT_UNKNOWN -1 -#define MA_SND_PCM_FORMAT_U8 1 -#define MA_SND_PCM_FORMAT_S16_LE 2 -#define MA_SND_PCM_FORMAT_S16_BE 3 -#define MA_SND_PCM_FORMAT_S24_LE 6 -#define MA_SND_PCM_FORMAT_S24_BE 7 -#define MA_SND_PCM_FORMAT_S32_LE 10 -#define MA_SND_PCM_FORMAT_S32_BE 11 -#define MA_SND_PCM_FORMAT_FLOAT_LE 14 -#define MA_SND_PCM_FORMAT_FLOAT_BE 15 -#define MA_SND_PCM_FORMAT_FLOAT64_LE 16 -#define MA_SND_PCM_FORMAT_FLOAT64_BE 17 -#define MA_SND_PCM_FORMAT_MU_LAW 20 -#define MA_SND_PCM_FORMAT_A_LAW 21 -#define MA_SND_PCM_FORMAT_S24_3LE 32 -#define MA_SND_PCM_FORMAT_S24_3BE 33 +static ma_result ma_context_init__dsound(const ma_context_config* pConfig, ma_context* pContext) +{ + MA_ASSERT(pContext != NULL); -/* snd_pcm_access_t */ -#define MA_SND_PCM_ACCESS_MMAP_INTERLEAVED 0 -#define MA_SND_PCM_ACCESS_MMAP_NONINTERLEAVED 1 -#define MA_SND_PCM_ACCESS_MMAP_COMPLEX 2 -#define MA_SND_PCM_ACCESS_RW_INTERLEAVED 3 -#define MA_SND_PCM_ACCESS_RW_NONINTERLEAVED 4 + (void)pConfig; -/* Channel positions. */ -#define MA_SND_CHMAP_UNKNOWN 0 -#define MA_SND_CHMAP_NA 1 -#define MA_SND_CHMAP_MONO 2 -#define MA_SND_CHMAP_FL 3 -#define MA_SND_CHMAP_FR 4 -#define MA_SND_CHMAP_RL 5 -#define MA_SND_CHMAP_RR 6 -#define MA_SND_CHMAP_FC 7 -#define MA_SND_CHMAP_LFE 8 -#define MA_SND_CHMAP_SL 9 -#define MA_SND_CHMAP_SR 10 -#define MA_SND_CHMAP_RC 11 -#define MA_SND_CHMAP_FLC 12 -#define MA_SND_CHMAP_FRC 13 -#define MA_SND_CHMAP_RLC 14 -#define MA_SND_CHMAP_RRC 15 -#define MA_SND_CHMAP_FLW 16 -#define MA_SND_CHMAP_FRW 17 -#define MA_SND_CHMAP_FLH 18 -#define MA_SND_CHMAP_FCH 19 -#define MA_SND_CHMAP_FRH 20 -#define MA_SND_CHMAP_TC 21 -#define MA_SND_CHMAP_TFL 22 -#define MA_SND_CHMAP_TFR 23 -#define MA_SND_CHMAP_TFC 24 -#define MA_SND_CHMAP_TRL 25 -#define MA_SND_CHMAP_TRR 26 -#define MA_SND_CHMAP_TRC 27 -#define MA_SND_CHMAP_TFLC 28 -#define MA_SND_CHMAP_TFRC 29 -#define MA_SND_CHMAP_TSL 30 -#define MA_SND_CHMAP_TSR 31 -#define MA_SND_CHMAP_LLFE 32 -#define MA_SND_CHMAP_RLFE 33 -#define MA_SND_CHMAP_BC 34 -#define MA_SND_CHMAP_BLC 35 -#define MA_SND_CHMAP_BRC 36 + pContext->dsound.hDSoundDLL = ma_dlopen(pContext, "dsound.dll"); + if (pContext->dsound.hDSoundDLL == NULL) { + return MA_API_NOT_FOUND; + } -/* Open mode flags. */ -#define MA_SND_PCM_NO_AUTO_RESAMPLE 0x00010000 -#define MA_SND_PCM_NO_AUTO_CHANNELS 0x00020000 -#define MA_SND_PCM_NO_AUTO_FORMAT 0x00040000 -#endif + pContext->dsound.DirectSoundCreate = ma_dlsym(pContext, pContext->dsound.hDSoundDLL, "DirectSoundCreate"); + pContext->dsound.DirectSoundEnumerateA = ma_dlsym(pContext, pContext->dsound.hDSoundDLL, "DirectSoundEnumerateA"); + pContext->dsound.DirectSoundCaptureCreate = ma_dlsym(pContext, pContext->dsound.hDSoundDLL, "DirectSoundCaptureCreate"); + pContext->dsound.DirectSoundCaptureEnumerateA = ma_dlsym(pContext, pContext->dsound.hDSoundDLL, "DirectSoundCaptureEnumerateA"); -typedef int (* ma_snd_pcm_open_proc) (ma_snd_pcm_t **pcm, const char *name, ma_snd_pcm_stream_t stream, int mode); -typedef int (* ma_snd_pcm_close_proc) (ma_snd_pcm_t *pcm); -typedef size_t (* ma_snd_pcm_hw_params_sizeof_proc) (void); -typedef int (* ma_snd_pcm_hw_params_any_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params); -typedef int (* ma_snd_pcm_hw_params_set_format_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t val); -typedef int (* ma_snd_pcm_hw_params_set_format_first_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t *format); -typedef void (* ma_snd_pcm_hw_params_get_format_mask_proc) (ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_mask_t *mask); -typedef int (* ma_snd_pcm_hw_params_set_channels_near_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *val); -typedef int (* ma_snd_pcm_hw_params_set_rate_resample_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val); -typedef int (* ma_snd_pcm_hw_params_set_rate_near_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *val, int *dir); -typedef int (* ma_snd_pcm_hw_params_set_buffer_size_near_proc)(ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_uframes_t *val); -typedef int (* ma_snd_pcm_hw_params_set_periods_near_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *val, int *dir); -typedef int (* ma_snd_pcm_hw_params_set_access_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_access_t _access); -typedef int (* ma_snd_pcm_hw_params_get_format_proc) (const ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t *format); -typedef int (* ma_snd_pcm_hw_params_get_channels_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val); -typedef int (* ma_snd_pcm_hw_params_get_channels_min_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val); -typedef int (* ma_snd_pcm_hw_params_get_channels_max_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val); -typedef int (* ma_snd_pcm_hw_params_get_rate_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *rate, int *dir); -typedef int (* ma_snd_pcm_hw_params_get_rate_min_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *rate, int *dir); -typedef int (* ma_snd_pcm_hw_params_get_rate_max_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *rate, int *dir); -typedef int (* ma_snd_pcm_hw_params_get_buffer_size_proc) (const ma_snd_pcm_hw_params_t *params, ma_snd_pcm_uframes_t *val); -typedef int (* ma_snd_pcm_hw_params_get_periods_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val, int *dir); -typedef int (* ma_snd_pcm_hw_params_get_access_proc) (const ma_snd_pcm_hw_params_t *params, ma_snd_pcm_access_t *_access); -typedef int (* ma_snd_pcm_hw_params_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params); -typedef size_t (* ma_snd_pcm_sw_params_sizeof_proc) (void); -typedef int (* ma_snd_pcm_sw_params_current_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params); -typedef int (* ma_snd_pcm_sw_params_get_boundary_proc) (ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t* val); -typedef int (* ma_snd_pcm_sw_params_set_avail_min_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t val); -typedef int (* ma_snd_pcm_sw_params_set_start_threshold_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t val); -typedef int (* ma_snd_pcm_sw_params_set_stop_threshold_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t val); -typedef int (* ma_snd_pcm_sw_params_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params); -typedef size_t (* ma_snd_pcm_format_mask_sizeof_proc) (void); -typedef int (* ma_snd_pcm_format_mask_test_proc) (const ma_snd_pcm_format_mask_t *mask, ma_snd_pcm_format_t val); -typedef ma_snd_pcm_chmap_t * (* ma_snd_pcm_get_chmap_proc) (ma_snd_pcm_t *pcm); -typedef int (* ma_snd_pcm_state_proc) (ma_snd_pcm_t *pcm); -typedef int (* ma_snd_pcm_prepare_proc) (ma_snd_pcm_t *pcm); -typedef int (* ma_snd_pcm_start_proc) (ma_snd_pcm_t *pcm); -typedef int (* ma_snd_pcm_drop_proc) (ma_snd_pcm_t *pcm); -typedef int (* ma_snd_pcm_drain_proc) (ma_snd_pcm_t *pcm); -typedef int (* ma_snd_device_name_hint_proc) (int card, const char *iface, void ***hints); -typedef char * (* ma_snd_device_name_get_hint_proc) (const void *hint, const char *id); -typedef int (* ma_snd_card_get_index_proc) (const char *name); -typedef int (* ma_snd_device_name_free_hint_proc) (void **hints); -typedef int (* ma_snd_pcm_mmap_begin_proc) (ma_snd_pcm_t *pcm, const ma_snd_pcm_channel_area_t **areas, ma_snd_pcm_uframes_t *offset, ma_snd_pcm_uframes_t *frames); -typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_mmap_commit_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_uframes_t offset, ma_snd_pcm_uframes_t frames); -typedef int (* ma_snd_pcm_recover_proc) (ma_snd_pcm_t *pcm, int err, int silent); -typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_readi_proc) (ma_snd_pcm_t *pcm, void *buffer, ma_snd_pcm_uframes_t size); -typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_writei_proc) (ma_snd_pcm_t *pcm, const void *buffer, ma_snd_pcm_uframes_t size); -typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_avail_proc) (ma_snd_pcm_t *pcm); -typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_avail_update_proc) (ma_snd_pcm_t *pcm); -typedef int (* ma_snd_pcm_wait_proc) (ma_snd_pcm_t *pcm, int timeout); -typedef int (* ma_snd_pcm_info_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_info_t* info); -typedef size_t (* ma_snd_pcm_info_sizeof_proc) (); -typedef const char* (* ma_snd_pcm_info_get_name_proc) (const ma_snd_pcm_info_t* info); -typedef int (* ma_snd_config_update_free_global_proc) (); + pContext->onUninit = ma_context_uninit__dsound; + pContext->onDeviceIDEqual = ma_context_is_device_id_equal__dsound; + pContext->onEnumDevices = ma_context_enumerate_devices__dsound; + pContext->onGetDeviceInfo = ma_context_get_device_info__dsound; + pContext->onDeviceInit = ma_device_init__dsound; + pContext->onDeviceUninit = ma_device_uninit__dsound; + pContext->onDeviceStart = NULL; /* Not used. Started in onDeviceMainLoop. */ + pContext->onDeviceStop = NULL; /* Not used. Stopped in onDeviceMainLoop. */ + pContext->onDeviceMainLoop = ma_device_main_loop__dsound; -/* This array specifies each of the common devices that can be used for both playback and capture. */ -const char* g_maCommonDeviceNamesALSA[] = { - "default", - "null", - "pulse", - "jack" -}; + return MA_SUCCESS; +} +#endif -/* This array allows us to blacklist specific playback devices. */ -const char* g_maBlacklistedPlaybackDeviceNamesALSA[] = { - "" -}; -/* This array allows us to blacklist specific capture devices. */ -const char* g_maBlacklistedCaptureDeviceNamesALSA[] = { - "" -}; +/****************************************************************************** + +WinMM Backend + +******************************************************************************/ +#ifdef MA_HAS_WINMM /* -This array allows miniaudio to control device-specific default buffer sizes. This uses a scaling factor. Order is important. If -any part of the string is present in the device's name, the associated scale will be used. +Some older compilers don't have WAVEOUTCAPS2A and WAVEINCAPS2A, so we'll need to write this ourselves. These structures +are exactly the same as the older ones but they have a few GUIDs for manufacturer/product/name identification. I'm keeping +the names the same as the Win32 library for consistency, but namespaced to avoid naming conflicts with the Win32 version. */ -static struct +typedef struct { - const char* name; - float scale; -} g_maDefaultBufferSizeScalesALSA[] = { - {"bcm2835 IEC958/HDMI", 2.0f}, - {"bcm2835 ALSA", 2.0f} -}; - -float ma_find_default_buffer_size_scale__alsa(const char* deviceName) + WORD wMid; + WORD wPid; + MMVERSION vDriverVersion; + CHAR szPname[MAXPNAMELEN]; + DWORD dwFormats; + WORD wChannels; + WORD wReserved1; + DWORD dwSupport; + GUID ManufacturerGuid; + GUID ProductGuid; + GUID NameGuid; +} MA_WAVEOUTCAPS2A; +typedef struct { - size_t i; + WORD wMid; + WORD wPid; + MMVERSION vDriverVersion; + CHAR szPname[MAXPNAMELEN]; + DWORD dwFormats; + WORD wChannels; + WORD wReserved1; + GUID ManufacturerGuid; + GUID ProductGuid; + GUID NameGuid; +} MA_WAVEINCAPS2A; - if (deviceName == NULL) { - return 1; - } +typedef UINT (WINAPI * MA_PFN_waveOutGetNumDevs)(void); +typedef MMRESULT (WINAPI * MA_PFN_waveOutGetDevCapsA)(ma_uintptr uDeviceID, LPWAVEOUTCAPSA pwoc, UINT cbwoc); +typedef MMRESULT (WINAPI * MA_PFN_waveOutOpen)(LPHWAVEOUT phwo, UINT uDeviceID, LPCWAVEFORMATEX pwfx, DWORD_PTR dwCallback, DWORD_PTR dwInstance, DWORD fdwOpen); +typedef MMRESULT (WINAPI * MA_PFN_waveOutClose)(HWAVEOUT hwo); +typedef MMRESULT (WINAPI * MA_PFN_waveOutPrepareHeader)(HWAVEOUT hwo, LPWAVEHDR pwh, UINT cbwh); +typedef MMRESULT (WINAPI * MA_PFN_waveOutUnprepareHeader)(HWAVEOUT hwo, LPWAVEHDR pwh, UINT cbwh); +typedef MMRESULT (WINAPI * MA_PFN_waveOutWrite)(HWAVEOUT hwo, LPWAVEHDR pwh, UINT cbwh); +typedef MMRESULT (WINAPI * MA_PFN_waveOutReset)(HWAVEOUT hwo); +typedef UINT (WINAPI * MA_PFN_waveInGetNumDevs)(void); +typedef MMRESULT (WINAPI * MA_PFN_waveInGetDevCapsA)(ma_uintptr uDeviceID, LPWAVEINCAPSA pwic, UINT cbwic); +typedef MMRESULT (WINAPI * MA_PFN_waveInOpen)(LPHWAVEIN phwi, UINT uDeviceID, LPCWAVEFORMATEX pwfx, DWORD_PTR dwCallback, DWORD_PTR dwInstance, DWORD fdwOpen); +typedef MMRESULT (WINAPI * MA_PFN_waveInClose)(HWAVEIN hwi); +typedef MMRESULT (WINAPI * MA_PFN_waveInPrepareHeader)(HWAVEIN hwi, LPWAVEHDR pwh, UINT cbwh); +typedef MMRESULT (WINAPI * MA_PFN_waveInUnprepareHeader)(HWAVEIN hwi, LPWAVEHDR pwh, UINT cbwh); +typedef MMRESULT (WINAPI * MA_PFN_waveInAddBuffer)(HWAVEIN hwi, LPWAVEHDR pwh, UINT cbwh); +typedef MMRESULT (WINAPI * MA_PFN_waveInStart)(HWAVEIN hwi); +typedef MMRESULT (WINAPI * MA_PFN_waveInReset)(HWAVEIN hwi); - for (i = 0; i < ma_countof(g_maDefaultBufferSizeScalesALSA); ++i) { - if (strstr(g_maDefaultBufferSizeScalesALSA[i].name, deviceName) != NULL) { - return g_maDefaultBufferSizeScalesALSA[i].scale; - } +static ma_result ma_result_from_MMRESULT(MMRESULT resultMM) +{ + switch (resultMM) { + case MMSYSERR_NOERROR: return MA_SUCCESS; + case MMSYSERR_BADDEVICEID: return MA_INVALID_ARGS; + case MMSYSERR_INVALHANDLE: return MA_INVALID_ARGS; + case MMSYSERR_NOMEM: return MA_OUT_OF_MEMORY; + case MMSYSERR_INVALFLAG: return MA_INVALID_ARGS; + case MMSYSERR_INVALPARAM: return MA_INVALID_ARGS; + case MMSYSERR_HANDLEBUSY: return MA_DEVICE_BUSY; + case MMSYSERR_ERROR: return MA_ERROR; + default: return MA_ERROR; } - - return 1; } -ma_snd_pcm_format_t ma_convert_ma_format_to_alsa_format(ma_format format) +static char* ma_find_last_character(char* str, char ch) { - ma_snd_pcm_format_t ALSAFormats[] = { - MA_SND_PCM_FORMAT_UNKNOWN, /* ma_format_unknown */ - MA_SND_PCM_FORMAT_U8, /* ma_format_u8 */ - MA_SND_PCM_FORMAT_S16_LE, /* ma_format_s16 */ - MA_SND_PCM_FORMAT_S24_3LE, /* ma_format_s24 */ - MA_SND_PCM_FORMAT_S32_LE, /* ma_format_s32 */ - MA_SND_PCM_FORMAT_FLOAT_LE /* ma_format_f32 */ - }; + char* last; - if (ma_is_big_endian()) { - ALSAFormats[0] = MA_SND_PCM_FORMAT_UNKNOWN; - ALSAFormats[1] = MA_SND_PCM_FORMAT_U8; - ALSAFormats[2] = MA_SND_PCM_FORMAT_S16_BE; - ALSAFormats[3] = MA_SND_PCM_FORMAT_S24_3BE; - ALSAFormats[4] = MA_SND_PCM_FORMAT_S32_BE; - ALSAFormats[5] = MA_SND_PCM_FORMAT_FLOAT_BE; + if (str == NULL) { + return NULL; } - return ALSAFormats[format]; -} - -ma_format ma_format_from_alsa(ma_snd_pcm_format_t formatALSA) -{ - if (ma_is_little_endian()) { - switch (formatALSA) { - case MA_SND_PCM_FORMAT_S16_LE: return ma_format_s16; - case MA_SND_PCM_FORMAT_S24_3LE: return ma_format_s24; - case MA_SND_PCM_FORMAT_S32_LE: return ma_format_s32; - case MA_SND_PCM_FORMAT_FLOAT_LE: return ma_format_f32; - default: break; - } - } else { - switch (formatALSA) { - case MA_SND_PCM_FORMAT_S16_BE: return ma_format_s16; - case MA_SND_PCM_FORMAT_S24_3BE: return ma_format_s24; - case MA_SND_PCM_FORMAT_S32_BE: return ma_format_s32; - case MA_SND_PCM_FORMAT_FLOAT_BE: return ma_format_f32; - default: break; + last = NULL; + while (*str != '\0') { + if (*str == ch) { + last = str; } - } - /* Endian agnostic. */ - switch (formatALSA) { - case MA_SND_PCM_FORMAT_U8: return ma_format_u8; - default: return ma_format_unknown; + str += 1; } + + return last; } -ma_channel ma_convert_alsa_channel_position_to_ma_channel(unsigned int alsaChannelPos) +static ma_uint32 ma_get_period_size_in_bytes(ma_uint32 periodSizeInFrames, ma_format format, ma_uint32 channels) { - switch (alsaChannelPos) - { - case MA_SND_CHMAP_MONO: return MA_CHANNEL_MONO; - case MA_SND_CHMAP_FL: return MA_CHANNEL_FRONT_LEFT; - case MA_SND_CHMAP_FR: return MA_CHANNEL_FRONT_RIGHT; - case MA_SND_CHMAP_RL: return MA_CHANNEL_BACK_LEFT; - case MA_SND_CHMAP_RR: return MA_CHANNEL_BACK_RIGHT; - case MA_SND_CHMAP_FC: return MA_CHANNEL_FRONT_CENTER; - case MA_SND_CHMAP_LFE: return MA_CHANNEL_LFE; - case MA_SND_CHMAP_SL: return MA_CHANNEL_SIDE_LEFT; - case MA_SND_CHMAP_SR: return MA_CHANNEL_SIDE_RIGHT; - case MA_SND_CHMAP_RC: return MA_CHANNEL_BACK_CENTER; - case MA_SND_CHMAP_FLC: return MA_CHANNEL_FRONT_LEFT_CENTER; - case MA_SND_CHMAP_FRC: return MA_CHANNEL_FRONT_RIGHT_CENTER; - case MA_SND_CHMAP_RLC: return 0; - case MA_SND_CHMAP_RRC: return 0; - case MA_SND_CHMAP_FLW: return 0; - case MA_SND_CHMAP_FRW: return 0; - case MA_SND_CHMAP_FLH: return 0; - case MA_SND_CHMAP_FCH: return 0; - case MA_SND_CHMAP_FRH: return 0; - case MA_SND_CHMAP_TC: return MA_CHANNEL_TOP_CENTER; - case MA_SND_CHMAP_TFL: return MA_CHANNEL_TOP_FRONT_LEFT; - case MA_SND_CHMAP_TFR: return MA_CHANNEL_TOP_FRONT_RIGHT; - case MA_SND_CHMAP_TFC: return MA_CHANNEL_TOP_FRONT_CENTER; - case MA_SND_CHMAP_TRL: return MA_CHANNEL_TOP_BACK_LEFT; - case MA_SND_CHMAP_TRR: return MA_CHANNEL_TOP_BACK_RIGHT; - case MA_SND_CHMAP_TRC: return MA_CHANNEL_TOP_BACK_CENTER; - default: break; - } - - return 0; + return periodSizeInFrames * ma_get_bytes_per_frame(format, channels); } -ma_bool32 ma_is_common_device_name__alsa(const char* name) + +/* +Our own "WAVECAPS" structure that contains generic information shared between WAVEOUTCAPS2 and WAVEINCAPS2 so +we can do things generically and typesafely. Names are being kept the same for consistency. +*/ +typedef struct { - size_t iName; - for (iName = 0; iName < ma_countof(g_maCommonDeviceNamesALSA); ++iName) { - if (ma_strcmp(name, g_maCommonDeviceNamesALSA[iName]) == 0) { - return MA_TRUE; + CHAR szPname[MAXPNAMELEN]; + DWORD dwFormats; + WORD wChannels; + GUID NameGuid; +} MA_WAVECAPSA; + +static ma_result ma_get_best_info_from_formats_flags__winmm(DWORD dwFormats, WORD channels, WORD* pBitsPerSample, DWORD* pSampleRate) +{ + WORD bitsPerSample = 0; + DWORD sampleRate = 0; + + if (pBitsPerSample) { + *pBitsPerSample = 0; + } + if (pSampleRate) { + *pSampleRate = 0; + } + + if (channels == 1) { + bitsPerSample = 16; + if ((dwFormats & WAVE_FORMAT_48M16) != 0) { + sampleRate = 48000; + } else if ((dwFormats & WAVE_FORMAT_44M16) != 0) { + sampleRate = 44100; + } else if ((dwFormats & WAVE_FORMAT_2M16) != 0) { + sampleRate = 22050; + } else if ((dwFormats & WAVE_FORMAT_1M16) != 0) { + sampleRate = 11025; + } else if ((dwFormats & WAVE_FORMAT_96M16) != 0) { + sampleRate = 96000; + } else { + bitsPerSample = 8; + if ((dwFormats & WAVE_FORMAT_48M08) != 0) { + sampleRate = 48000; + } else if ((dwFormats & WAVE_FORMAT_44M08) != 0) { + sampleRate = 44100; + } else if ((dwFormats & WAVE_FORMAT_2M08) != 0) { + sampleRate = 22050; + } else if ((dwFormats & WAVE_FORMAT_1M08) != 0) { + sampleRate = 11025; + } else if ((dwFormats & WAVE_FORMAT_96M08) != 0) { + sampleRate = 96000; + } else { + return MA_FORMAT_NOT_SUPPORTED; + } + } + } else { + bitsPerSample = 16; + if ((dwFormats & WAVE_FORMAT_48S16) != 0) { + sampleRate = 48000; + } else if ((dwFormats & WAVE_FORMAT_44S16) != 0) { + sampleRate = 44100; + } else if ((dwFormats & WAVE_FORMAT_2S16) != 0) { + sampleRate = 22050; + } else if ((dwFormats & WAVE_FORMAT_1S16) != 0) { + sampleRate = 11025; + } else if ((dwFormats & WAVE_FORMAT_96S16) != 0) { + sampleRate = 96000; + } else { + bitsPerSample = 8; + if ((dwFormats & WAVE_FORMAT_48S08) != 0) { + sampleRate = 48000; + } else if ((dwFormats & WAVE_FORMAT_44S08) != 0) { + sampleRate = 44100; + } else if ((dwFormats & WAVE_FORMAT_2S08) != 0) { + sampleRate = 22050; + } else if ((dwFormats & WAVE_FORMAT_1S08) != 0) { + sampleRate = 11025; + } else if ((dwFormats & WAVE_FORMAT_96S08) != 0) { + sampleRate = 96000; + } else { + return MA_FORMAT_NOT_SUPPORTED; + } } } - return MA_FALSE; -} - - -ma_bool32 ma_is_playback_device_blacklisted__alsa(const char* name) -{ - size_t iName; - for (iName = 0; iName < ma_countof(g_maBlacklistedPlaybackDeviceNamesALSA); ++iName) { - if (ma_strcmp(name, g_maBlacklistedPlaybackDeviceNamesALSA[iName]) == 0) { - return MA_TRUE; - } + if (pBitsPerSample) { + *pBitsPerSample = bitsPerSample; } - - return MA_FALSE; -} - -ma_bool32 ma_is_capture_device_blacklisted__alsa(const char* name) -{ - size_t iName; - for (iName = 0; iName < ma_countof(g_maBlacklistedCaptureDeviceNamesALSA); ++iName) { - if (ma_strcmp(name, g_maBlacklistedCaptureDeviceNamesALSA[iName]) == 0) { - return MA_TRUE; - } + if (pSampleRate) { + *pSampleRate = sampleRate; } - return MA_FALSE; + return MA_SUCCESS; } -ma_bool32 ma_is_device_blacklisted__alsa(ma_device_type deviceType, const char* name) +static ma_result ma_formats_flags_to_WAVEFORMATEX__winmm(DWORD dwFormats, WORD channels, WAVEFORMATEX* pWF) { - if (deviceType == ma_device_type_playback) { - return ma_is_playback_device_blacklisted__alsa(name); - } else { - return ma_is_capture_device_blacklisted__alsa(name); - } -} + MA_ASSERT(pWF != NULL); + MA_ZERO_OBJECT(pWF); + pWF->cbSize = sizeof(*pWF); + pWF->wFormatTag = WAVE_FORMAT_PCM; + pWF->nChannels = (WORD)channels; + if (pWF->nChannels > 2) { + pWF->nChannels = 2; + } -const char* ma_find_char(const char* str, char c, int* index) -{ - int i = 0; - for (;;) { - if (str[i] == '\0') { - if (index) *index = -1; - return NULL; + if (channels == 1) { + pWF->wBitsPerSample = 16; + if ((dwFormats & WAVE_FORMAT_48M16) != 0) { + pWF->nSamplesPerSec = 48000; + } else if ((dwFormats & WAVE_FORMAT_44M16) != 0) { + pWF->nSamplesPerSec = 44100; + } else if ((dwFormats & WAVE_FORMAT_2M16) != 0) { + pWF->nSamplesPerSec = 22050; + } else if ((dwFormats & WAVE_FORMAT_1M16) != 0) { + pWF->nSamplesPerSec = 11025; + } else if ((dwFormats & WAVE_FORMAT_96M16) != 0) { + pWF->nSamplesPerSec = 96000; + } else { + pWF->wBitsPerSample = 8; + if ((dwFormats & WAVE_FORMAT_48M08) != 0) { + pWF->nSamplesPerSec = 48000; + } else if ((dwFormats & WAVE_FORMAT_44M08) != 0) { + pWF->nSamplesPerSec = 44100; + } else if ((dwFormats & WAVE_FORMAT_2M08) != 0) { + pWF->nSamplesPerSec = 22050; + } else if ((dwFormats & WAVE_FORMAT_1M08) != 0) { + pWF->nSamplesPerSec = 11025; + } else if ((dwFormats & WAVE_FORMAT_96M08) != 0) { + pWF->nSamplesPerSec = 96000; + } else { + return MA_FORMAT_NOT_SUPPORTED; + } } - - if (str[i] == c) { - if (index) *index = i; - return str + i; + } else { + pWF->wBitsPerSample = 16; + if ((dwFormats & WAVE_FORMAT_48S16) != 0) { + pWF->nSamplesPerSec = 48000; + } else if ((dwFormats & WAVE_FORMAT_44S16) != 0) { + pWF->nSamplesPerSec = 44100; + } else if ((dwFormats & WAVE_FORMAT_2S16) != 0) { + pWF->nSamplesPerSec = 22050; + } else if ((dwFormats & WAVE_FORMAT_1S16) != 0) { + pWF->nSamplesPerSec = 11025; + } else if ((dwFormats & WAVE_FORMAT_96S16) != 0) { + pWF->nSamplesPerSec = 96000; + } else { + pWF->wBitsPerSample = 8; + if ((dwFormats & WAVE_FORMAT_48S08) != 0) { + pWF->nSamplesPerSec = 48000; + } else if ((dwFormats & WAVE_FORMAT_44S08) != 0) { + pWF->nSamplesPerSec = 44100; + } else if ((dwFormats & WAVE_FORMAT_2S08) != 0) { + pWF->nSamplesPerSec = 22050; + } else if ((dwFormats & WAVE_FORMAT_1S08) != 0) { + pWF->nSamplesPerSec = 11025; + } else if ((dwFormats & WAVE_FORMAT_96S08) != 0) { + pWF->nSamplesPerSec = 96000; + } else { + return MA_FORMAT_NOT_SUPPORTED; + } } - - i += 1; } - /* Should never get here, but treat it as though the character was not found to make me feel better inside. */ - if (index) *index = -1; - return NULL; + pWF->nBlockAlign = (pWF->nChannels * pWF->wBitsPerSample) / 8; + pWF->nAvgBytesPerSec = pWF->nBlockAlign * pWF->nSamplesPerSec; + + return MA_SUCCESS; } -ma_bool32 ma_is_device_name_in_hw_format__alsa(const char* hwid) +static ma_result ma_context_get_device_info_from_WAVECAPS(ma_context* pContext, MA_WAVECAPSA* pCaps, ma_device_info* pDeviceInfo) { - /* This function is just checking whether or not hwid is in "hw:%d,%d" format. */ - - int commaPos; - const char* dev; - int i; - - if (hwid == NULL) { - return MA_FALSE; - } - - if (hwid[0] != 'h' || hwid[1] != 'w' || hwid[2] != ':') { - return MA_FALSE; - } + WORD bitsPerSample; + DWORD sampleRate; + ma_result result; - hwid += 3; + MA_ASSERT(pContext != NULL); + MA_ASSERT(pCaps != NULL); + MA_ASSERT(pDeviceInfo != NULL); - dev = ma_find_char(hwid, ',', &commaPos); - if (dev == NULL) { - return MA_FALSE; - } else { - dev += 1; /* Skip past the ",". */ - } + /* + Name / Description + + Unfortunately the name specified in WAVE(OUT/IN)CAPS2 is limited to 31 characters. This results in an unprofessional looking + situation where the names of the devices are truncated. To help work around this, we need to look at the name GUID and try + looking in the registry for the full name. If we can't find it there, we need to just fall back to the default name. + */ - /* Check if the part between the ":" and the "," contains only numbers. If not, return false. */ - for (i = 0; i < commaPos; ++i) { - if (hwid[i] < '0' || hwid[i] > '9') { - return MA_FALSE; - } - } + /* Set the default to begin with. */ + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), pCaps->szPname, (size_t)-1); - /* Check if everything after the "," is numeric. If not, return false. */ - i = 0; - while (dev[i] != '\0') { - if (dev[i] < '0' || dev[i] > '9') { - return MA_FALSE; - } - i += 1; - } + /* + Now try the registry. There's a few things to consider here: + - The name GUID can be null, in which we case we just need to stick to the original 31 characters. + - If the name GUID is not present in the registry we'll also need to stick to the original 31 characters. + - I like consistency, so I want the returned device names to be consistent with those returned by WASAPI and DirectSound. The + problem, however is that WASAPI and DirectSound use " ()" format (such as "Speakers (High Definition Audio)"), + but WinMM does not specificy the component name. From my admittedly limited testing, I've notice the component name seems to + usually fit within the 31 characters of the fixed sized buffer, so what I'm going to do is parse that string for the component + name, and then concatenate the name from the registry. + */ + if (!ma_is_guid_equal(&pCaps->NameGuid, &MA_GUID_NULL)) { + wchar_t guidStrW[256]; + if (((MA_PFN_StringFromGUID2)pContext->win32.StringFromGUID2)(&pCaps->NameGuid, guidStrW, ma_countof(guidStrW)) > 0) { + char guidStr[256]; + char keyStr[1024]; + HKEY hKey; - return MA_TRUE; -} + WideCharToMultiByte(CP_UTF8, 0, guidStrW, -1, guidStr, sizeof(guidStr), 0, FALSE); -int ma_convert_device_name_to_hw_format__alsa(ma_context* pContext, char* dst, size_t dstSize, const char* src) /* Returns 0 on success, non-0 on error. */ -{ - /* src should look something like this: "hw:CARD=I82801AAICH,DEV=0" */ + ma_strcpy_s(keyStr, sizeof(keyStr), "SYSTEM\\CurrentControlSet\\Control\\MediaCategories\\"); + ma_strcat_s(keyStr, sizeof(keyStr), guidStr); - int colonPos; - int commaPos; - char card[256]; - const char* dev; - int cardIndex; + if (((MA_PFN_RegOpenKeyExA)pContext->win32.RegOpenKeyExA)(HKEY_LOCAL_MACHINE, keyStr, 0, KEY_READ, &hKey) == ERROR_SUCCESS) { + BYTE nameFromReg[512]; + DWORD nameFromRegSize = sizeof(nameFromReg); + result = ((MA_PFN_RegQueryValueExA)pContext->win32.RegQueryValueExA)(hKey, "Name", 0, NULL, (LPBYTE)nameFromReg, (LPDWORD)&nameFromRegSize); + ((MA_PFN_RegCloseKey)pContext->win32.RegCloseKey)(hKey); - if (dst == NULL) { - return -1; - } - if (dstSize < 7) { - return -1; /* Absolute minimum size of the output buffer is 7 bytes. */ - } + if (result == ERROR_SUCCESS) { + /* We have the value from the registry, so now we need to construct the name string. */ + char name[1024]; + if (ma_strcpy_s(name, sizeof(name), pDeviceInfo->name) == 0) { + char* nameBeg = ma_find_last_character(name, '('); + if (nameBeg != NULL) { + size_t leadingLen = (nameBeg - name); + ma_strncpy_s(nameBeg + 1, sizeof(name) - leadingLen, (const char*)nameFromReg, (size_t)-1); - *dst = '\0'; /* Safety. */ - if (src == NULL) { - return -1; - } + /* The closing ")", if it can fit. */ + if (leadingLen + nameFromRegSize < sizeof(name)-1) { + ma_strcat_s(name, sizeof(name), ")"); + } - /* If the input name is already in "hw:%d,%d" format, just return that verbatim. */ - if (ma_is_device_name_in_hw_format__alsa(src)) { - return ma_strcpy_s(dst, dstSize, src); + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), name, (size_t)-1); + } + } + } + } + } } - src = ma_find_char(src, ':', &colonPos); - if (src == NULL) { - return -1; /* Couldn't find a colon */ - } - dev = ma_find_char(src, ',', &commaPos); - if (dev == NULL) { - dev = "0"; - ma_strncpy_s(card, sizeof(card), src+6, (size_t)-1); /* +6 = ":CARD=" */ - } else { - dev = dev + 5; /* +5 = ",DEV=" */ - ma_strncpy_s(card, sizeof(card), src+6, commaPos-6); /* +6 = ":CARD=" */ + result = ma_get_best_info_from_formats_flags__winmm(pCaps->dwFormats, pCaps->wChannels, &bitsPerSample, &sampleRate); + if (result != MA_SUCCESS) { + return result; } - cardIndex = ((ma_snd_card_get_index_proc)pContext->alsa.snd_card_get_index)(card); - if (cardIndex < 0) { - return -2; /* Failed to retrieve the card index. */ + pDeviceInfo->minChannels = pCaps->wChannels; + pDeviceInfo->maxChannels = pCaps->wChannels; + pDeviceInfo->minSampleRate = sampleRate; + pDeviceInfo->maxSampleRate = sampleRate; + pDeviceInfo->formatCount = 1; + if (bitsPerSample == 8) { + pDeviceInfo->formats[0] = ma_format_u8; + } else if (bitsPerSample == 16) { + pDeviceInfo->formats[0] = ma_format_s16; + } else if (bitsPerSample == 24) { + pDeviceInfo->formats[0] = ma_format_s24; + } else if (bitsPerSample == 32) { + pDeviceInfo->formats[0] = ma_format_s32; + } else { + return MA_FORMAT_NOT_SUPPORTED; } - /*printf("TESTING: CARD=%s,DEV=%s\n", card, dev); */ + return MA_SUCCESS; +} +static ma_result ma_context_get_device_info_from_WAVEOUTCAPS2(ma_context* pContext, MA_WAVEOUTCAPS2A* pCaps, ma_device_info* pDeviceInfo) +{ + MA_WAVECAPSA caps; - /* Construction. */ - dst[0] = 'h'; dst[1] = 'w'; dst[2] = ':'; - if (ma_itoa_s(cardIndex, dst+3, dstSize-3, 10) != 0) { - return -3; - } - if (ma_strcat_s(dst, dstSize, ",") != 0) { - return -3; - } - if (ma_strcat_s(dst, dstSize, dev) != 0) { - return -3; - } + MA_ASSERT(pContext != NULL); + MA_ASSERT(pCaps != NULL); + MA_ASSERT(pDeviceInfo != NULL); - return 0; + MA_COPY_MEMORY(caps.szPname, pCaps->szPname, sizeof(caps.szPname)); + caps.dwFormats = pCaps->dwFormats; + caps.wChannels = pCaps->wChannels; + caps.NameGuid = pCaps->NameGuid; + return ma_context_get_device_info_from_WAVECAPS(pContext, &caps, pDeviceInfo); } -ma_bool32 ma_does_id_exist_in_list__alsa(ma_device_id* pUniqueIDs, ma_uint32 count, const char* pHWID) +static ma_result ma_context_get_device_info_from_WAVEINCAPS2(ma_context* pContext, MA_WAVEINCAPS2A* pCaps, ma_device_info* pDeviceInfo) { - ma_uint32 i; - - ma_assert(pHWID != NULL); + MA_WAVECAPSA caps; - for (i = 0; i < count; ++i) { - if (ma_strcmp(pUniqueIDs[i].alsa, pHWID) == 0) { - return MA_TRUE; - } - } + MA_ASSERT(pContext != NULL); + MA_ASSERT(pCaps != NULL); + MA_ASSERT(pDeviceInfo != NULL); - return MA_FALSE; + MA_COPY_MEMORY(caps.szPname, pCaps->szPname, sizeof(caps.szPname)); + caps.dwFormats = pCaps->dwFormats; + caps.wChannels = pCaps->wChannels; + caps.NameGuid = pCaps->NameGuid; + return ma_context_get_device_info_from_WAVECAPS(pContext, &caps, pDeviceInfo); } -ma_result ma_context_open_pcm__alsa(ma_context* pContext, ma_share_mode shareMode, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_snd_pcm_t** ppPCM) +static ma_bool32 ma_context_is_device_id_equal__winmm(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) { - ma_snd_pcm_t* pPCM; - ma_snd_pcm_stream_t stream; - int openMode; + MA_ASSERT(pContext != NULL); + MA_ASSERT(pID0 != NULL); + MA_ASSERT(pID1 != NULL); + (void)pContext; - ma_assert(pContext != NULL); - ma_assert(ppPCM != NULL); + return pID0->winmm == pID1->winmm; +} - *ppPCM = NULL; - pPCM = NULL; +static ma_result ma_context_enumerate_devices__winmm(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +{ + UINT playbackDeviceCount; + UINT captureDeviceCount; + UINT iPlaybackDevice; + UINT iCaptureDevice; - stream = (deviceType == ma_device_type_playback) ? MA_SND_PCM_STREAM_PLAYBACK : MA_SND_PCM_STREAM_CAPTURE; - openMode = MA_SND_PCM_NO_AUTO_RESAMPLE | MA_SND_PCM_NO_AUTO_CHANNELS | MA_SND_PCM_NO_AUTO_FORMAT; + MA_ASSERT(pContext != NULL); + MA_ASSERT(callback != NULL); - if (pDeviceID == NULL) { - ma_bool32 isDeviceOpen; - size_t i; + /* Playback. */ + playbackDeviceCount = ((MA_PFN_waveOutGetNumDevs)pContext->winmm.waveOutGetNumDevs)(); + for (iPlaybackDevice = 0; iPlaybackDevice < playbackDeviceCount; ++iPlaybackDevice) { + MMRESULT result; + MA_WAVEOUTCAPS2A caps; - /* - We're opening the default device. I don't know if trying anything other than "default" is necessary, but it makes - me feel better to try as hard as we can get to get _something_ working. - */ - const char* defaultDeviceNames[] = { - "default", - NULL, - NULL, - NULL, - NULL, - NULL, - NULL - }; + MA_ZERO_OBJECT(&caps); - if (shareMode == ma_share_mode_exclusive) { - defaultDeviceNames[1] = "hw"; - defaultDeviceNames[2] = "hw:0"; - defaultDeviceNames[3] = "hw:0,0"; - } else { - if (deviceType == ma_device_type_playback) { - defaultDeviceNames[1] = "dmix"; - defaultDeviceNames[2] = "dmix:0"; - defaultDeviceNames[3] = "dmix:0,0"; - } else { - defaultDeviceNames[1] = "dsnoop"; - defaultDeviceNames[2] = "dsnoop:0"; - defaultDeviceNames[3] = "dsnoop:0,0"; - } - defaultDeviceNames[4] = "hw"; - defaultDeviceNames[5] = "hw:0"; - defaultDeviceNames[6] = "hw:0,0"; - } + result = ((MA_PFN_waveOutGetDevCapsA)pContext->winmm.waveOutGetDevCapsA)(iPlaybackDevice, (WAVEOUTCAPSA*)&caps, sizeof(caps)); + if (result == MMSYSERR_NOERROR) { + ma_device_info deviceInfo; - isDeviceOpen = MA_FALSE; - for (i = 0; i < ma_countof(defaultDeviceNames); ++i) { - if (defaultDeviceNames[i] != NULL && defaultDeviceNames[i][0] != '\0') { - if (((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, defaultDeviceNames[i], stream, openMode) == 0) { - isDeviceOpen = MA_TRUE; - break; + MA_ZERO_OBJECT(&deviceInfo); + deviceInfo.id.winmm = iPlaybackDevice; + + if (ma_context_get_device_info_from_WAVEOUTCAPS2(pContext, &caps, &deviceInfo) == MA_SUCCESS) { + ma_bool32 cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); + if (cbResult == MA_FALSE) { + return MA_SUCCESS; /* Enumeration was stopped. */ } } } + } - if (!isDeviceOpen) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_open() failed when trying to open an appropriate default device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } - } else { - /* - We're trying to open a specific device. There's a few things to consider here: - - miniaudio recongnizes a special format of device id that excludes the "hw", "dmix", etc. prefix. It looks like this: ":0,0", ":0,1", etc. When - an ID of this format is specified, it indicates to miniaudio that it can try different combinations of plugins ("hw", "dmix", etc.) until it - finds an appropriate one that works. This comes in very handy when trying to open a device in shared mode ("dmix"), vs exclusive mode ("hw"). - */ - - /* May end up needing to make small adjustments to the ID, so make a copy. */ - ma_device_id deviceID = *pDeviceID; - ma_bool32 isDeviceOpen = MA_FALSE; - - if (deviceID.alsa[0] != ':') { - /* The ID is not in ":0,0" format. Use the ID exactly as-is. */ - if (((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, deviceID.alsa, stream, openMode) == 0) { - isDeviceOpen = MA_TRUE; - } - } else { - char hwid[256]; + /* Capture. */ + captureDeviceCount = ((MA_PFN_waveInGetNumDevs)pContext->winmm.waveInGetNumDevs)(); + for (iCaptureDevice = 0; iCaptureDevice < captureDeviceCount; ++iCaptureDevice) { + MMRESULT result; + MA_WAVEINCAPS2A caps; - /* The ID is in ":0,0" format. Try different plugins depending on the shared mode. */ - if (deviceID.alsa[1] == '\0') { - deviceID.alsa[0] = '\0'; /* An ID of ":" should be converted to "". */ - } + MA_ZERO_OBJECT(&caps); - if (shareMode == ma_share_mode_shared) { - if (deviceType == ma_device_type_playback) { - ma_strcpy_s(hwid, sizeof(hwid), "dmix"); - } else { - ma_strcpy_s(hwid, sizeof(hwid), "dsnoop"); - } + result = ((MA_PFN_waveInGetDevCapsA)pContext->winmm.waveInGetDevCapsA)(iCaptureDevice, (WAVEINCAPSA*)&caps, sizeof(caps)); + if (result == MMSYSERR_NOERROR) { + ma_device_info deviceInfo; - if (ma_strcat_s(hwid, sizeof(hwid), deviceID.alsa) == 0) { - if (((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, hwid, stream, openMode) == 0) { - isDeviceOpen = MA_TRUE; - } - } - } + MA_ZERO_OBJECT(&deviceInfo); + deviceInfo.id.winmm = iCaptureDevice; - /* If at this point we still don't have an open device it means we're either preferencing exclusive mode or opening with "dmix"/"dsnoop" failed. */ - if (!isDeviceOpen) { - ma_strcpy_s(hwid, sizeof(hwid), "hw"); - if (ma_strcat_s(hwid, sizeof(hwid), deviceID.alsa) == 0) { - if (((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, hwid, stream, openMode) == 0) { - isDeviceOpen = MA_TRUE; - } + if (ma_context_get_device_info_from_WAVEINCAPS2(pContext, &caps, &deviceInfo) == MA_SUCCESS) { + ma_bool32 cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); + if (cbResult == MA_FALSE) { + return MA_SUCCESS; /* Enumeration was stopped. */ } } } - - if (!isDeviceOpen) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_open() failed.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } } - *ppPCM = pPCM; return MA_SUCCESS; } - -ma_bool32 ma_context_is_device_id_equal__alsa(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) -{ - ma_assert(pContext != NULL); - ma_assert(pID0 != NULL); - ma_assert(pID1 != NULL); - (void)pContext; - - return ma_strcmp(pID0->alsa, pID1->alsa) == 0; -} - -ma_result ma_context_enumerate_devices__alsa(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +static ma_result ma_context_get_device_info__winmm(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) { - ma_bool32 cbResult = MA_TRUE; - char** ppDeviceHints; - ma_device_id* pUniqueIDs = NULL; - ma_uint32 uniqueIDCount = 0; - char** ppNextDeviceHint; + UINT winMMDeviceID; - ma_assert(pContext != NULL); - ma_assert(callback != NULL); + MA_ASSERT(pContext != NULL); - ma_mutex_lock(&pContext->alsa.internalDeviceEnumLock); + if (shareMode == ma_share_mode_exclusive) { + return MA_SHARE_MODE_NOT_SUPPORTED; + } - if (((ma_snd_device_name_hint_proc)pContext->alsa.snd_device_name_hint)(-1, "pcm", (void***)&ppDeviceHints) < 0) { - ma_mutex_unlock(&pContext->alsa.internalDeviceEnumLock); - return MA_NO_BACKEND; + winMMDeviceID = 0; + if (pDeviceID != NULL) { + winMMDeviceID = (UINT)pDeviceID->winmm; } - ppNextDeviceHint = ppDeviceHints; - while (*ppNextDeviceHint != NULL) { - char* NAME = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "NAME"); - char* DESC = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "DESC"); - char* IOID = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "IOID"); - ma_device_type deviceType = ma_device_type_playback; - ma_bool32 stopEnumeration = MA_FALSE; - char hwid[sizeof(pUniqueIDs->alsa)]; - ma_device_info deviceInfo; + pDeviceInfo->id.winmm = winMMDeviceID; - if ((IOID == NULL || ma_strcmp(IOID, "Output") == 0)) { - deviceType = ma_device_type_playback; + if (deviceType == ma_device_type_playback) { + MMRESULT result; + MA_WAVEOUTCAPS2A caps; + + MA_ZERO_OBJECT(&caps); + + result = ((MA_PFN_waveOutGetDevCapsA)pContext->winmm.waveOutGetDevCapsA)(winMMDeviceID, (WAVEOUTCAPSA*)&caps, sizeof(caps)); + if (result == MMSYSERR_NOERROR) { + return ma_context_get_device_info_from_WAVEOUTCAPS2(pContext, &caps, pDeviceInfo); } - if ((IOID != NULL && ma_strcmp(IOID, "Input" ) == 0)) { - deviceType = ma_device_type_capture; + } else { + MMRESULT result; + MA_WAVEINCAPS2A caps; + + MA_ZERO_OBJECT(&caps); + + result = ((MA_PFN_waveInGetDevCapsA)pContext->winmm.waveInGetDevCapsA)(winMMDeviceID, (WAVEINCAPSA*)&caps, sizeof(caps)); + if (result == MMSYSERR_NOERROR) { + return ma_context_get_device_info_from_WAVEINCAPS2(pContext, &caps, pDeviceInfo); } + } - if (NAME != NULL) { - if (pContext->alsa.useVerboseDeviceEnumeration) { - /* Verbose mode. Use the name exactly as-is. */ - ma_strncpy_s(hwid, sizeof(hwid), NAME, (size_t)-1); - } else { - /* Simplified mode. Use ":%d,%d" format. */ - if (ma_convert_device_name_to_hw_format__alsa(pContext, hwid, sizeof(hwid), NAME) == 0) { - /* - At this point, hwid looks like "hw:0,0". In simplified enumeration mode, we actually want to strip off the - plugin name so it looks like ":0,0". The reason for this is that this special format is detected at device - initialization time and is used as an indicator to try and use the most appropriate plugin depending on the - device type and sharing mode. - */ - char* dst = hwid; - char* src = hwid+2; - while ((*dst++ = *src++)); - } else { - /* Conversion to "hw:%d,%d" failed. Just use the name as-is. */ - ma_strncpy_s(hwid, sizeof(hwid), NAME, (size_t)-1); - } + return MA_NO_DEVICE; +} + + +static void ma_device_uninit__winmm(ma_device* pDevice) +{ + MA_ASSERT(pDevice != NULL); + + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + ((MA_PFN_waveInClose)pDevice->pContext->winmm.waveInClose)((HWAVEIN)pDevice->winmm.hDeviceCapture); + CloseHandle((HANDLE)pDevice->winmm.hEventCapture); + } - if (ma_does_id_exist_in_list__alsa(pUniqueIDs, uniqueIDCount, hwid)) { - goto next_device; /* The device has already been enumerated. Move on to the next one. */ - } else { - /* The device has not yet been enumerated. Make sure it's added to our list so that it's not enumerated again. */ - ma_device_id* pNewUniqueIDs = (ma_device_id*)ma_realloc(pUniqueIDs, sizeof(*pUniqueIDs) * (uniqueIDCount + 1)); - if (pNewUniqueIDs == NULL) { - goto next_device; /* Failed to allocate memory. */ - } + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + ((MA_PFN_waveOutReset)pDevice->pContext->winmm.waveOutReset)((HWAVEOUT)pDevice->winmm.hDevicePlayback); + ((MA_PFN_waveOutClose)pDevice->pContext->winmm.waveOutClose)((HWAVEOUT)pDevice->winmm.hDevicePlayback); + CloseHandle((HANDLE)pDevice->winmm.hEventPlayback); + } - pUniqueIDs = pNewUniqueIDs; - ma_copy_memory(pUniqueIDs[uniqueIDCount].alsa, hwid, sizeof(hwid)); - uniqueIDCount += 1; - } - } + ma__free_from_callbacks(pDevice->winmm._pHeapData, &pDevice->pContext->allocationCallbacks); + + MA_ZERO_OBJECT(&pDevice->winmm); /* Safety. */ +} + +static ma_result ma_device_init__winmm(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) +{ + const char* errorMsg = ""; + ma_result errorCode = MA_ERROR; + ma_result result = MA_SUCCESS; + ma_uint32 heapSize; + UINT winMMDeviceIDPlayback = 0; + UINT winMMDeviceIDCapture = 0; + ma_uint32 periodSizeInMilliseconds; + + MA_ASSERT(pDevice != NULL); + MA_ZERO_OBJECT(&pDevice->winmm); + + if (pConfig->deviceType == ma_device_type_loopback) { + return MA_DEVICE_TYPE_NOT_SUPPORTED; + } + + /* No exlusive mode with WinMM. */ + if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) || + ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) { + return MA_SHARE_MODE_NOT_SUPPORTED; + } + + periodSizeInMilliseconds = pConfig->periodSizeInMilliseconds; + if (periodSizeInMilliseconds == 0) { + periodSizeInMilliseconds = ma_calculate_buffer_size_in_milliseconds_from_frames(pConfig->periodSizeInFrames, pConfig->sampleRate); + } + + /* WinMM has horrible latency. */ + if (pDevice->usingDefaultBufferSize) { + if (pConfig->performanceProfile == ma_performance_profile_low_latency) { + periodSizeInMilliseconds = 40; } else { - ma_zero_memory(hwid, sizeof(hwid)); + periodSizeInMilliseconds = 400; } + } - ma_zero_object(&deviceInfo); - ma_strncpy_s(deviceInfo.id.alsa, sizeof(deviceInfo.id.alsa), hwid, (size_t)-1); - /* - DESC is the friendly name. We treat this slightly differently depending on whether or not we are using verbose - device enumeration. In verbose mode we want to take the entire description so that the end-user can distinguish - between the subdevices of each card/dev pair. In simplified mode, however, we only want the first part of the - description. - - The value in DESC seems to be split into two lines, with the first line being the name of the device and the - second line being a description of the device. I don't like having the description be across two lines because - it makes formatting ugly and annoying. I'm therefore deciding to put it all on a single line with the second line - being put into parentheses. In simplified mode I'm just stripping the second line entirely. - */ - if (DESC != NULL) { - int lfPos; - const char* line2 = ma_find_char(DESC, '\n', &lfPos); - if (line2 != NULL) { - line2 += 1; /* Skip past the new-line character. */ + if (pConfig->playback.pDeviceID != NULL) { + winMMDeviceIDPlayback = (UINT)pConfig->playback.pDeviceID->winmm; + } + if (pConfig->capture.pDeviceID != NULL) { + winMMDeviceIDCapture = (UINT)pConfig->capture.pDeviceID->winmm; + } - if (pContext->alsa.useVerboseDeviceEnumeration) { - /* Verbose mode. Put the second line in brackets. */ - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), DESC, lfPos); - ma_strcat_s (deviceInfo.name, sizeof(deviceInfo.name), " ("); - ma_strcat_s (deviceInfo.name, sizeof(deviceInfo.name), line2); - ma_strcat_s (deviceInfo.name, sizeof(deviceInfo.name), ")"); - } else { - /* Simplified mode. Strip the second line entirely. */ - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), DESC, lfPos); - } - } else { - /* There's no second line. Just copy the whole description. */ - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), DESC, (size_t)-1); - } - } + /* The capture device needs to be initialized first. */ + if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { + WAVEINCAPSA caps; + WAVEFORMATEX wf; + MMRESULT resultMM; - if (!ma_is_device_blacklisted__alsa(deviceType, NAME)) { - cbResult = callback(pContext, deviceType, &deviceInfo, pUserData); + /* We use an event to know when a new fragment needs to be enqueued. */ + pDevice->winmm.hEventCapture = (ma_handle)CreateEvent(NULL, TRUE, TRUE, NULL); + if (pDevice->winmm.hEventCapture == NULL) { + errorMsg = "[WinMM] Failed to create event for fragment enqueing for the capture device.", errorCode = MA_FAILED_TO_CREATE_EVENT; + goto on_error; } - /* - Some devices are both playback and capture, but they are only enumerated by ALSA once. We need to fire the callback - again for the other device type in this case. We do this for known devices. - */ - if (cbResult) { - if (ma_is_common_device_name__alsa(NAME)) { - if (deviceType == ma_device_type_playback) { - if (!ma_is_capture_device_blacklisted__alsa(NAME)) { - cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); - } - } else { - if (!ma_is_playback_device_blacklisted__alsa(NAME)) { - cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); - } - } - } + /* The format should be based on the device's actual format. */ + if (((MA_PFN_waveInGetDevCapsA)pContext->winmm.waveInGetDevCapsA)(winMMDeviceIDCapture, &caps, sizeof(caps)) != MMSYSERR_NOERROR) { + errorMsg = "[WinMM] Failed to retrieve internal device caps.", errorCode = MA_FORMAT_NOT_SUPPORTED; + goto on_error; } - if (cbResult == MA_FALSE) { - stopEnumeration = MA_TRUE; + result = ma_formats_flags_to_WAVEFORMATEX__winmm(caps.dwFormats, caps.wChannels, &wf); + if (result != MA_SUCCESS) { + errorMsg = "[WinMM] Could not find appropriate format for internal device.", errorCode = result; + goto on_error; } - next_device: - free(NAME); - free(DESC); - free(IOID); - ppNextDeviceHint += 1; - - /* We need to stop enumeration if the callback returned false. */ - if (stopEnumeration) { - break; + resultMM = ((MA_PFN_waveInOpen)pDevice->pContext->winmm.waveInOpen)((LPHWAVEIN)&pDevice->winmm.hDeviceCapture, winMMDeviceIDCapture, &wf, (DWORD_PTR)pDevice->winmm.hEventCapture, (DWORD_PTR)pDevice, CALLBACK_EVENT | WAVE_ALLOWSYNC); + if (resultMM != MMSYSERR_NOERROR) { + errorMsg = "[WinMM] Failed to open capture device.", errorCode = MA_FAILED_TO_OPEN_BACKEND_DEVICE; + goto on_error; } + + pDevice->capture.internalFormat = ma_format_from_WAVEFORMATEX(&wf); + pDevice->capture.internalChannels = wf.nChannels; + pDevice->capture.internalSampleRate = wf.nSamplesPerSec; + ma_get_standard_channel_map(ma_standard_channel_map_microsoft, pDevice->capture.internalChannels, pDevice->capture.internalChannelMap); + pDevice->capture.internalPeriods = pConfig->periods; + pDevice->capture.internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(periodSizeInMilliseconds, pDevice->capture.internalSampleRate); } - ma_free(pUniqueIDs); - ((ma_snd_device_name_free_hint_proc)pContext->alsa.snd_device_name_free_hint)((void**)ppDeviceHints); + if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { + WAVEOUTCAPSA caps; + WAVEFORMATEX wf; + MMRESULT resultMM; - ma_mutex_unlock(&pContext->alsa.internalDeviceEnumLock); + /* We use an event to know when a new fragment needs to be enqueued. */ + pDevice->winmm.hEventPlayback = (ma_handle)CreateEvent(NULL, TRUE, TRUE, NULL); + if (pDevice->winmm.hEventPlayback == NULL) { + errorMsg = "[WinMM] Failed to create event for fragment enqueing for the playback device.", errorCode = MA_FAILED_TO_CREATE_EVENT; + goto on_error; + } - return MA_SUCCESS; -} + /* The format should be based on the device's actual format. */ + if (((MA_PFN_waveOutGetDevCapsA)pContext->winmm.waveOutGetDevCapsA)(winMMDeviceIDPlayback, &caps, sizeof(caps)) != MMSYSERR_NOERROR) { + errorMsg = "[WinMM] Failed to retrieve internal device caps.", errorCode = MA_FORMAT_NOT_SUPPORTED; + goto on_error; + } + result = ma_formats_flags_to_WAVEFORMATEX__winmm(caps.dwFormats, caps.wChannels, &wf); + if (result != MA_SUCCESS) { + errorMsg = "[WinMM] Could not find appropriate format for internal device.", errorCode = result; + goto on_error; + } -typedef struct -{ - ma_device_type deviceType; - const ma_device_id* pDeviceID; - ma_share_mode shareMode; - ma_device_info* pDeviceInfo; - ma_bool32 foundDevice; -} ma_context_get_device_info_enum_callback_data__alsa; + resultMM = ((MA_PFN_waveOutOpen)pContext->winmm.waveOutOpen)((LPHWAVEOUT)&pDevice->winmm.hDevicePlayback, winMMDeviceIDPlayback, &wf, (DWORD_PTR)pDevice->winmm.hEventPlayback, (DWORD_PTR)pDevice, CALLBACK_EVENT | WAVE_ALLOWSYNC); + if (resultMM != MMSYSERR_NOERROR) { + errorMsg = "[WinMM] Failed to open playback device.", errorCode = MA_FAILED_TO_OPEN_BACKEND_DEVICE; + goto on_error; + } -ma_bool32 ma_context_get_device_info_enum_callback__alsa(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pDeviceInfo, void* pUserData) -{ - ma_context_get_device_info_enum_callback_data__alsa* pData = (ma_context_get_device_info_enum_callback_data__alsa*)pUserData; - ma_assert(pData != NULL); + pDevice->playback.internalFormat = ma_format_from_WAVEFORMATEX(&wf); + pDevice->playback.internalChannels = wf.nChannels; + pDevice->playback.internalSampleRate = wf.nSamplesPerSec; + ma_get_standard_channel_map(ma_standard_channel_map_microsoft, pDevice->playback.internalChannels, pDevice->playback.internalChannelMap); + pDevice->playback.internalPeriods = pConfig->periods; + pDevice->playback.internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(periodSizeInMilliseconds, pDevice->playback.internalSampleRate); + } - if (pData->pDeviceID == NULL && ma_strcmp(pDeviceInfo->id.alsa, "default") == 0) { - ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pDeviceInfo->name, (size_t)-1); - pData->foundDevice = MA_TRUE; - } else { - if (pData->deviceType == deviceType && ma_context_is_device_id_equal__alsa(pContext, pData->pDeviceID, &pDeviceInfo->id)) { - ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pDeviceInfo->name, (size_t)-1); - pData->foundDevice = MA_TRUE; - } + /* + The heap allocated data is allocated like so: + + [Capture WAVEHDRs][Playback WAVEHDRs][Capture Intermediary Buffer][Playback Intermediary Buffer] + */ + heapSize = 0; + if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { + heapSize += sizeof(WAVEHDR)*pDevice->capture.internalPeriods + (pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); + } + if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { + heapSize += sizeof(WAVEHDR)*pDevice->playback.internalPeriods + (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); } - /* Keep enumerating until we have found the device. */ - return !pData->foundDevice; -} + pDevice->winmm._pHeapData = (ma_uint8*)ma__calloc_from_callbacks(heapSize, &pContext->allocationCallbacks); + if (pDevice->winmm._pHeapData == NULL) { + errorMsg = "[WinMM] Failed to allocate memory for the intermediary buffer.", errorCode = MA_OUT_OF_MEMORY; + goto on_error; + } -ma_result ma_context_get_device_info__alsa(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) -{ - ma_context_get_device_info_enum_callback_data__alsa data; - ma_result result; - ma_snd_pcm_t* pPCM; - ma_snd_pcm_hw_params_t* pHWParams; - ma_snd_pcm_format_mask_t* pFormatMask; - int sampleRateDir = 0; + MA_ZERO_MEMORY(pDevice->winmm._pHeapData, heapSize); - ma_assert(pContext != NULL); + if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { + ma_uint32 iPeriod; - /* We just enumerate to find basic information about the device. */ - data.deviceType = deviceType; - data.pDeviceID = pDeviceID; - data.shareMode = shareMode; - data.pDeviceInfo = pDeviceInfo; - data.foundDevice = MA_FALSE; - result = ma_context_enumerate_devices__alsa(pContext, ma_context_get_device_info_enum_callback__alsa, &data); - if (result != MA_SUCCESS) { - return result; - } + if (pConfig->deviceType == ma_device_type_capture) { + pDevice->winmm.pWAVEHDRCapture = pDevice->winmm._pHeapData; + pDevice->winmm.pIntermediaryBufferCapture = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDevice->capture.internalPeriods)); + } else { + pDevice->winmm.pWAVEHDRCapture = pDevice->winmm._pHeapData; + pDevice->winmm.pIntermediaryBufferCapture = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDevice->capture.internalPeriods + pDevice->playback.internalPeriods)); + } - if (!data.foundDevice) { - return MA_NO_DEVICE; - } + /* Prepare headers. */ + for (iPeriod = 0; iPeriod < pDevice->capture.internalPeriods; ++iPeriod) { + ma_uint32 periodSizeInBytes = ma_get_period_size_in_bytes(pDevice->capture.internalPeriodSizeInFrames, pDevice->capture.internalFormat, pDevice->capture.internalChannels); - /* For detailed info we need to open the device. */ - result = ma_context_open_pcm__alsa(pContext, shareMode, deviceType, pDeviceID, &pPCM); - if (result != MA_SUCCESS) { - return result; + ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].lpData = (LPSTR)(pDevice->winmm.pIntermediaryBufferCapture + (periodSizeInBytes*iPeriod)); + ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwBufferLength = periodSizeInBytes; + ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwFlags = 0L; + ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwLoops = 0L; + ((MA_PFN_waveInPrepareHeader)pContext->winmm.waveInPrepareHeader)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod], sizeof(WAVEHDR)); + + /* + The user data of the WAVEHDR structure is a single flag the controls whether or not it is ready for writing. Consider it to be named "isLocked". A value of 0 means + it's unlocked and available for writing. A value of 1 means it's locked. + */ + ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwUser = 0; + } } + if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { + ma_uint32 iPeriod; + + if (pConfig->deviceType == ma_device_type_playback) { + pDevice->winmm.pWAVEHDRPlayback = pDevice->winmm._pHeapData; + pDevice->winmm.pIntermediaryBufferPlayback = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*pDevice->playback.internalPeriods); + } else { + pDevice->winmm.pWAVEHDRPlayback = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDevice->capture.internalPeriods)); + pDevice->winmm.pIntermediaryBufferPlayback = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDevice->capture.internalPeriods + pDevice->playback.internalPeriods)) + (pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); + } - /* We need to initialize a HW parameters object in order to know what formats are supported. */ - pHWParams = (ma_snd_pcm_hw_params_t*)calloc(1, ((ma_snd_pcm_hw_params_sizeof_proc)pContext->alsa.snd_pcm_hw_params_sizeof)()); - if (pHWParams == NULL) { - return MA_OUT_OF_MEMORY; - } + /* Prepare headers. */ + for (iPeriod = 0; iPeriod < pDevice->playback.internalPeriods; ++iPeriod) { + ma_uint32 periodSizeInBytes = ma_get_period_size_in_bytes(pDevice->playback.internalPeriodSizeInFrames, pDevice->playback.internalFormat, pDevice->playback.internalChannels); - if (((ma_snd_pcm_hw_params_any_proc)pContext->alsa.snd_pcm_hw_params_any)(pPCM, pHWParams) < 0) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to initialize hardware parameters. snd_pcm_hw_params_any() failed.", MA_FAILED_TO_CONFIGURE_BACKEND_DEVICE); + ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].lpData = (LPSTR)(pDevice->winmm.pIntermediaryBufferPlayback + (periodSizeInBytes*iPeriod)); + ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwBufferLength = periodSizeInBytes; + ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwFlags = 0L; + ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwLoops = 0L; + ((MA_PFN_waveOutPrepareHeader)pContext->winmm.waveOutPrepareHeader)((HWAVEOUT)pDevice->winmm.hDevicePlayback, &((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod], sizeof(WAVEHDR)); + + /* + The user data of the WAVEHDR structure is a single flag the controls whether or not it is ready for writing. Consider it to be named "isLocked". A value of 0 means + it's unlocked and available for writing. A value of 1 means it's locked. + */ + ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwUser = 0; + } } - ((ma_snd_pcm_hw_params_get_channels_min_proc)pContext->alsa.snd_pcm_hw_params_get_channels_min)(pHWParams, &pDeviceInfo->minChannels); - ((ma_snd_pcm_hw_params_get_channels_max_proc)pContext->alsa.snd_pcm_hw_params_get_channels_max)(pHWParams, &pDeviceInfo->maxChannels); - ((ma_snd_pcm_hw_params_get_rate_min_proc)pContext->alsa.snd_pcm_hw_params_get_rate_min)(pHWParams, &pDeviceInfo->minSampleRate, &sampleRateDir); - ((ma_snd_pcm_hw_params_get_rate_max_proc)pContext->alsa.snd_pcm_hw_params_get_rate_max)(pHWParams, &pDeviceInfo->maxSampleRate, &sampleRateDir); + return MA_SUCCESS; - /* Formats. */ - pFormatMask = (ma_snd_pcm_format_mask_t*)calloc(1, ((ma_snd_pcm_format_mask_sizeof_proc)pContext->alsa.snd_pcm_format_mask_sizeof)()); - if (pFormatMask == NULL) { - return MA_OUT_OF_MEMORY; +on_error: + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + if (pDevice->winmm.pWAVEHDRCapture != NULL) { + ma_uint32 iPeriod; + for (iPeriod = 0; iPeriod < pDevice->capture.internalPeriods; ++iPeriod) { + ((MA_PFN_waveInUnprepareHeader)pContext->winmm.waveInUnprepareHeader)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod], sizeof(WAVEHDR)); + } + } + + ((MA_PFN_waveInClose)pContext->winmm.waveInClose)((HWAVEIN)pDevice->winmm.hDeviceCapture); } - ((ma_snd_pcm_hw_params_get_format_mask_proc)pContext->alsa.snd_pcm_hw_params_get_format_mask)(pHWParams, pFormatMask); + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + if (pDevice->winmm.pWAVEHDRCapture != NULL) { + ma_uint32 iPeriod; + for (iPeriod = 0; iPeriod < pDevice->playback.internalPeriods; ++iPeriod) { + ((MA_PFN_waveOutUnprepareHeader)pContext->winmm.waveOutUnprepareHeader)((HWAVEOUT)pDevice->winmm.hDevicePlayback, &((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod], sizeof(WAVEHDR)); + } + } - pDeviceInfo->formatCount = 0; - if (((ma_snd_pcm_format_mask_test_proc)pContext->alsa.snd_pcm_format_mask_test)(pFormatMask, MA_SND_PCM_FORMAT_U8)) { - pDeviceInfo->formats[pDeviceInfo->formatCount++] = ma_format_u8; - } - if (((ma_snd_pcm_format_mask_test_proc)pContext->alsa.snd_pcm_format_mask_test)(pFormatMask, MA_SND_PCM_FORMAT_S16_LE)) { - pDeviceInfo->formats[pDeviceInfo->formatCount++] = ma_format_s16; - } - if (((ma_snd_pcm_format_mask_test_proc)pContext->alsa.snd_pcm_format_mask_test)(pFormatMask, MA_SND_PCM_FORMAT_S24_3LE)) { - pDeviceInfo->formats[pDeviceInfo->formatCount++] = ma_format_s24; - } - if (((ma_snd_pcm_format_mask_test_proc)pContext->alsa.snd_pcm_format_mask_test)(pFormatMask, MA_SND_PCM_FORMAT_S32_LE)) { - pDeviceInfo->formats[pDeviceInfo->formatCount++] = ma_format_s32; - } - if (((ma_snd_pcm_format_mask_test_proc)pContext->alsa.snd_pcm_format_mask_test)(pFormatMask, MA_SND_PCM_FORMAT_FLOAT_LE)) { - pDeviceInfo->formats[pDeviceInfo->formatCount++] = ma_format_f32; + ((MA_PFN_waveOutClose)pContext->winmm.waveOutClose)((HWAVEOUT)pDevice->winmm.hDevicePlayback); } - ma_free(pFormatMask); - ma_free(pHWParams); - - ((ma_snd_pcm_close_proc)pContext->alsa.snd_pcm_close)(pPCM); - return MA_SUCCESS; + ma__free_from_callbacks(pDevice->winmm._pHeapData, &pContext->allocationCallbacks); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, errorMsg, errorCode); } - -#if 0 -/* -Waits for a number of frames to become available for either capture or playback. The return -value is the number of frames available. - -This will return early if the main loop is broken with ma_device__break_main_loop(). -*/ -ma_uint32 ma_device__wait_for_frames__alsa(ma_device* pDevice, ma_bool32* pRequiresRestart) +static ma_result ma_device_stop__winmm(ma_device* pDevice) { - ma_assert(pDevice != NULL); - - if (pRequiresRestart) *pRequiresRestart = MA_FALSE; - - /* I want it so that this function returns the period size in frames. We just wait until that number of frames are available and then return. */ - ma_uint32 periodSizeInFrames = pDevice->bufferSizeInFrames / pDevice->periods; - while (!pDevice->alsa.breakFromMainLoop) { - ma_snd_pcm_sframes_t framesAvailable = ((ma_snd_pcm_avail_update_proc)pDevice->pContext->alsa.snd_pcm_avail_update)((ma_snd_pcm_t*)pDevice->alsa.pPCM); - if (framesAvailable < 0) { - if (framesAvailable == -EPIPE) { - if (((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCM, framesAvailable, MA_TRUE) < 0) { - return 0; - } + MMRESULT resultMM; - /* A device recovery means a restart for mmap mode. */ - if (pRequiresRestart) { - *pRequiresRestart = MA_TRUE; - } + MA_ASSERT(pDevice != NULL); - /* Try again, but if it fails this time just return an error. */ - framesAvailable = ((ma_snd_pcm_avail_update_proc)pDevice->pContext->alsa.snd_pcm_avail_update)((ma_snd_pcm_t*)pDevice->alsa.pPCM); - if (framesAvailable < 0) { - return 0; - } - } + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + if (pDevice->winmm.hDeviceCapture == NULL) { + return MA_INVALID_ARGS; } - if (framesAvailable >= periodSizeInFrames) { - return periodSizeInFrames; + resultMM = ((MA_PFN_waveInReset)pDevice->pContext->winmm.waveInReset)((HWAVEIN)pDevice->winmm.hDeviceCapture); + if (resultMM != MMSYSERR_NOERROR) { + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WinMM] WARNING: Failed to reset capture device.", ma_result_from_MMRESULT(resultMM)); } + } - if (framesAvailable < periodSizeInFrames) { - /* Less than a whole period is available so keep waiting. */ - int waitResult = ((ma_snd_pcm_wait_proc)pDevice->pContext->alsa.snd_pcm_wait)((ma_snd_pcm_t*)pDevice->alsa.pPCM, -1); - if (waitResult < 0) { - if (waitResult == -EPIPE) { - if (((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCM, waitResult, MA_TRUE) < 0) { - return 0; - } + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + ma_uint32 iPeriod; + WAVEHDR* pWAVEHDR; - /* A device recovery means a restart for mmap mode. */ - if (pRequiresRestart) { - *pRequiresRestart = MA_TRUE; - } + if (pDevice->winmm.hDevicePlayback == NULL) { + return MA_INVALID_ARGS; + } + + /* We need to drain the device. To do this we just loop over each header and if it's locked just wait for the event. */ + pWAVEHDR = (WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback; + for (iPeriod = 0; iPeriod < pDevice->playback.internalPeriods; iPeriod += 1) { + if (pWAVEHDR[iPeriod].dwUser == 1) { /* 1 = locked. */ + if (WaitForSingleObject((HANDLE)pDevice->winmm.hEventPlayback, INFINITE) != WAIT_OBJECT_0) { + break; /* An error occurred so just abandon ship and stop the device without draining. */ } + + pWAVEHDR[iPeriod].dwUser = 0; } } - } - /* We'll get here if the loop was terminated. Just return whatever's available. */ - ma_snd_pcm_sframes_t framesAvailable = ((ma_snd_pcm_avail_update_proc)pDevice->pContext->alsa.snd_pcm_avail_update)((ma_snd_pcm_t*)pDevice->alsa.pPCM); - if (framesAvailable < 0) { - return 0; + resultMM = ((MA_PFN_waveOutReset)pDevice->pContext->winmm.waveOutReset)((HWAVEOUT)pDevice->winmm.hDevicePlayback); + if (resultMM != MMSYSERR_NOERROR) { + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WinMM] WARNING: Failed to reset playback device.", ma_result_from_MMRESULT(resultMM)); + } } - return framesAvailable; + return MA_SUCCESS; } -ma_bool32 ma_device_read_from_client_and_write__alsa(ma_device* pDevice) +static ma_result ma_device_write__winmm(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) { - ma_assert(pDevice != NULL); - if (!ma_device_is_started(pDevice) && ma_device__get_state(pDevice) != MA_STATE_STARTING) { - return MA_FALSE; - } - if (pDevice->alsa.breakFromMainLoop) { - return MA_FALSE; + ma_result result = MA_SUCCESS; + MMRESULT resultMM; + ma_uint32 totalFramesWritten; + WAVEHDR* pWAVEHDR; + + MA_ASSERT(pDevice != NULL); + MA_ASSERT(pPCMFrames != NULL); + + if (pFramesWritten != NULL) { + *pFramesWritten = 0; } - if (pDevice->alsa.isUsingMMap) { - /* mmap. */ - ma_bool32 requiresRestart; - ma_uint32 framesAvailable = ma_device__wait_for_frames__alsa(pDevice, &requiresRestart); - if (framesAvailable == 0) { - return MA_FALSE; - } + pWAVEHDR = (WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback; - /* Don't bother asking the client for more audio data if we're just stopping the device anyway. */ - if (pDevice->alsa.breakFromMainLoop) { - return MA_FALSE; - } + /* Keep processing as much data as possible. */ + totalFramesWritten = 0; + while (totalFramesWritten < frameCount) { + /* If the current header has some space available we need to write part of it. */ + if (pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwUser == 0) { /* 0 = unlocked. */ + /* + This header has room in it. We copy as much of it as we can. If we end up fully consuming the buffer we need to + write it out and move on to the next iteration. + */ + ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + ma_uint32 framesRemainingInHeader = (pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwBufferLength/bpf) - pDevice->winmm.headerFramesConsumedPlayback; - const ma_snd_pcm_channel_area_t* pAreas; - ma_snd_pcm_uframes_t mappedOffset; - ma_snd_pcm_uframes_t mappedFrames = framesAvailable; - while (framesAvailable > 0) { - int result = ((ma_snd_pcm_mmap_begin_proc)pDevice->pContext->alsa.snd_pcm_mmap_begin)((ma_snd_pcm_t*)pDevice->alsa.pPCM, &pAreas, &mappedOffset, &mappedFrames); - if (result < 0) { - return MA_FALSE; - } + ma_uint32 framesToCopy = ma_min(framesRemainingInHeader, (frameCount - totalFramesWritten)); + const void* pSrc = ma_offset_ptr(pPCMFrames, totalFramesWritten*bpf); + void* pDst = ma_offset_ptr(pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].lpData, pDevice->winmm.headerFramesConsumedPlayback*bpf); + MA_COPY_MEMORY(pDst, pSrc, framesToCopy*bpf); - if (mappedFrames > 0) { - void* pBuffer = (ma_uint8*)pAreas[0].addr + ((pAreas[0].first + (mappedOffset * pAreas[0].step)) / 8); - ma_device__read_frames_from_client(pDevice, mappedFrames, pBuffer); - } + pDevice->winmm.headerFramesConsumedPlayback += framesToCopy; + totalFramesWritten += framesToCopy; - result = ((ma_snd_pcm_mmap_commit_proc)pDevice->pContext->alsa.snd_pcm_mmap_commit)((ma_snd_pcm_t*)pDevice->alsa.pPCM, mappedOffset, mappedFrames); - if (result < 0 || (ma_snd_pcm_uframes_t)result != mappedFrames) { - ((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCM, result, MA_TRUE); - return MA_FALSE; - } + /* If we've consumed the buffer entirely we need to write it out to the device. */ + if (pDevice->winmm.headerFramesConsumedPlayback == (pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwBufferLength/bpf)) { + pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwUser = 1; /* 1 = locked. */ + pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwFlags &= ~WHDR_DONE; /* <-- Need to make sure the WHDR_DONE flag is unset. */ - if (requiresRestart) { - if (((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCM) < 0) { - return MA_FALSE; + /* Make sure the event is reset to a non-signaled state to ensure we don't prematurely return from WaitForSingleObject(). */ + ResetEvent((HANDLE)pDevice->winmm.hEventPlayback); + + /* The device will be started here. */ + resultMM = ((MA_PFN_waveOutWrite)pDevice->pContext->winmm.waveOutWrite)((HWAVEOUT)pDevice->winmm.hDevicePlayback, &pWAVEHDR[pDevice->winmm.iNextHeaderPlayback], sizeof(WAVEHDR)); + if (resultMM != MMSYSERR_NOERROR) { + result = ma_result_from_MMRESULT(resultMM); + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WinMM] waveOutWrite() failed.", result); + break; } - } - if (framesAvailable >= mappedFrames) { - framesAvailable -= mappedFrames; - } else { - framesAvailable = 0; - } - } - } else { - /* readi/writei. */ - while (!pDevice->alsa.breakFromMainLoop) { - ma_uint32 framesAvailable = ma_device__wait_for_frames__alsa(pDevice, NULL); - if (framesAvailable == 0) { - continue; + /* Make sure we move to the next header. */ + pDevice->winmm.iNextHeaderPlayback = (pDevice->winmm.iNextHeaderPlayback + 1) % pDevice->playback.internalPeriods; + pDevice->winmm.headerFramesConsumedPlayback = 0; } - /* Don't bother asking the client for more audio data if we're just stopping the device anyway. */ - if (pDevice->alsa.breakFromMainLoop) { - return MA_FALSE; + /* If at this point we have consumed the entire input buffer we can return. */ + MA_ASSERT(totalFramesWritten <= frameCount); + if (totalFramesWritten == frameCount) { + break; } - ma_device__read_frames_from_client(pDevice, framesAvailable, pDevice->alsa.pIntermediaryBuffer); - - ma_snd_pcm_sframes_t framesWritten = ((ma_snd_pcm_writei_proc)pDevice->pContext->alsa.snd_pcm_writei)((ma_snd_pcm_t*)pDevice->alsa.pPCM, pDevice->alsa.pIntermediaryBuffer, framesAvailable); - if (framesWritten < 0) { - if (framesWritten == -EAGAIN) { - continue; /* Just keep trying... */ - } else if (framesWritten == -EPIPE) { - /* Underrun. Just recover and try writing again. */ - if (((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCM, framesWritten, MA_TRUE) < 0) { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to recover device after underrun.", MA_FAILED_TO_START_BACKEND_DEVICE); - return MA_FALSE; - } + /* Getting here means there's more to process. */ + continue; + } - framesWritten = ((ma_snd_pcm_writei_proc)pDevice->pContext->alsa.snd_pcm_writei)((ma_snd_pcm_t*)pDevice->alsa.pPCM, pDevice->alsa.pIntermediaryBuffer, framesAvailable); - if (framesWritten < 0) { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to write data to the internal device.", MA_FAILED_TO_SEND_DATA_TO_DEVICE); - return MA_FALSE; - } + /* Getting here means there isn't enough room in the buffer and we need to wait for one to become available. */ + if (WaitForSingleObject((HANDLE)pDevice->winmm.hEventPlayback, INFINITE) != WAIT_OBJECT_0) { + result = MA_ERROR; + break; + } - break; /* Success. */ - } else { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_writei() failed when writing initial data.", MA_FAILED_TO_SEND_DATA_TO_DEVICE); - return MA_FALSE; - } - } else { - break; /* Success. */ - } + /* Something happened. If the next buffer has been marked as done we need to reset a bit of state. */ + if ((pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwFlags & WHDR_DONE) != 0) { + pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwUser = 0; /* 0 = unlocked (make it available for writing). */ + pDevice->winmm.headerFramesConsumedPlayback = 0; + } + + /* If the device has been stopped we need to break. */ + if (ma_device__get_state(pDevice) != MA_STATE_STARTED) { + break; } } - return MA_TRUE; + if (pFramesWritten != NULL) { + *pFramesWritten = totalFramesWritten; + } + + return result; } -ma_bool32 ma_device_read_and_send_to_client__alsa(ma_device* pDevice) +static ma_result ma_device_read__winmm(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead) { - ma_assert(pDevice != NULL); - if (!ma_device_is_started(pDevice)) { - return MA_FALSE; - } - if (pDevice->alsa.breakFromMainLoop) { - return MA_FALSE; + ma_result result = MA_SUCCESS; + MMRESULT resultMM; + ma_uint32 totalFramesRead; + WAVEHDR* pWAVEHDR; + + MA_ASSERT(pDevice != NULL); + MA_ASSERT(pPCMFrames != NULL); + + if (pFramesRead != NULL) { + *pFramesRead = 0; } - ma_uint32 framesToSend = 0; - void* pBuffer = NULL; - if (pDevice->alsa.pIntermediaryBuffer == NULL) { - /* mmap. */ - ma_bool32 requiresRestart; - ma_uint32 framesAvailable = ma_device__wait_for_frames__alsa(pDevice, &requiresRestart); - if (framesAvailable == 0) { - return MA_FALSE; - } + pWAVEHDR = (WAVEHDR*)pDevice->winmm.pWAVEHDRCapture; - const ma_snd_pcm_channel_area_t* pAreas; - ma_snd_pcm_uframes_t mappedOffset; - ma_snd_pcm_uframes_t mappedFrames = framesAvailable; - while (framesAvailable > 0) { - int result = ((ma_snd_pcm_mmap_begin_proc)pDevice->pContext->alsa.snd_pcm_mmap_begin)((ma_snd_pcm_t*)pDevice->alsa.pPCM, &pAreas, &mappedOffset, &mappedFrames); - if (result < 0) { - return MA_FALSE; - } + /* Keep processing as much data as possible. */ + totalFramesRead = 0; + while (totalFramesRead < frameCount) { + /* If the current header has some space available we need to write part of it. */ + if (pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwUser == 0) { /* 0 = unlocked. */ + /* The buffer is available for reading. If we fully consume it we need to add it back to the buffer. */ + ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + ma_uint32 framesRemainingInHeader = (pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwBufferLength/bpf) - pDevice->winmm.headerFramesConsumedCapture; - if (mappedFrames > 0) { - void* pBuffer = (ma_uint8*)pAreas[0].addr + ((pAreas[0].first + (mappedOffset * pAreas[0].step)) / 8); - ma_device__send_frames_to_client(pDevice, mappedFrames, pBuffer); - } + ma_uint32 framesToCopy = ma_min(framesRemainingInHeader, (frameCount - totalFramesRead)); + const void* pSrc = ma_offset_ptr(pWAVEHDR[pDevice->winmm.iNextHeaderCapture].lpData, pDevice->winmm.headerFramesConsumedCapture*bpf); + void* pDst = ma_offset_ptr(pPCMFrames, totalFramesRead*bpf); + MA_COPY_MEMORY(pDst, pSrc, framesToCopy*bpf); - result = ((ma_snd_pcm_mmap_commit_proc)pDevice->pContext->alsa.snd_pcm_mmap_commit)((ma_snd_pcm_t*)pDevice->alsa.pPCM, mappedOffset, mappedFrames); - if (result < 0 || (ma_snd_pcm_uframes_t)result != mappedFrames) { - ((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCM, result, MA_TRUE); - return MA_FALSE; - } + pDevice->winmm.headerFramesConsumedCapture += framesToCopy; + totalFramesRead += framesToCopy; - if (requiresRestart) { - if (((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCM) < 0) { - return MA_FALSE; + /* If we've consumed the buffer entirely we need to add it back to the device. */ + if (pDevice->winmm.headerFramesConsumedCapture == (pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwBufferLength/bpf)) { + pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwUser = 1; /* 1 = locked. */ + pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwFlags &= ~WHDR_DONE; /* <-- Need to make sure the WHDR_DONE flag is unset. */ + + /* Make sure the event is reset to a non-signaled state to ensure we don't prematurely return from WaitForSingleObject(). */ + ResetEvent((HANDLE)pDevice->winmm.hEventCapture); + + /* The device will be started here. */ + resultMM = ((MA_PFN_waveInAddBuffer)pDevice->pContext->winmm.waveInAddBuffer)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((LPWAVEHDR)pDevice->winmm.pWAVEHDRCapture)[pDevice->winmm.iNextHeaderCapture], sizeof(WAVEHDR)); + if (resultMM != MMSYSERR_NOERROR) { + result = ma_result_from_MMRESULT(resultMM); + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WinMM] waveInAddBuffer() failed.", result); + break; } - } - if (framesAvailable >= mappedFrames) { - framesAvailable -= mappedFrames; - } else { - framesAvailable = 0; + /* Make sure we move to the next header. */ + pDevice->winmm.iNextHeaderCapture = (pDevice->winmm.iNextHeaderCapture + 1) % pDevice->capture.internalPeriods; + pDevice->winmm.headerFramesConsumedCapture = 0; } - } - } else { - /* readi/writei. */ - ma_snd_pcm_sframes_t framesRead = 0; - while (!pDevice->alsa.breakFromMainLoop) { - ma_uint32 framesAvailable = ma_device__wait_for_frames__alsa(pDevice, NULL); - if (framesAvailable == 0) { - continue; + + /* If at this point we have filled the entire input buffer we can return. */ + MA_ASSERT(totalFramesRead <= frameCount); + if (totalFramesRead == frameCount) { + break; } - framesRead = ((ma_snd_pcm_readi_proc)pDevice->pContext->alsa.snd_pcm_readi)((ma_snd_pcm_t*)pDevice->alsa.pPCM, pDevice->alsa.pIntermediaryBuffer, framesAvailable); - if (framesRead < 0) { - if (framesRead == -EAGAIN) { - continue; /* Just keep trying... */ - } else if (framesRead == -EPIPE) { - /* Overrun. Just recover and try reading again. */ - if (((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCM, framesRead, MA_TRUE) < 0) { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to recover device after overrun.", MA_FAILED_TO_START_BACKEND_DEVICE); - return MA_FALSE; - } + /* Getting here means there's more to process. */ + continue; + } - framesRead = ((ma_snd_pcm_readi_proc)pDevice->pContext->alsa.snd_pcm_readi)((ma_snd_pcm_t*)pDevice->alsa.pPCM, pDevice->alsa.pIntermediaryBuffer, framesAvailable); - if (framesRead < 0) { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to read data from the internal device.", MA_FAILED_TO_READ_DATA_FROM_DEVICE); - return MA_FALSE; - } + /* Getting here means there isn't enough any data left to send to the client which means we need to wait for more. */ + if (WaitForSingleObject((HANDLE)pDevice->winmm.hEventCapture, INFINITE) != WAIT_OBJECT_0) { + result = MA_ERROR; + break; + } - break; /* Success. */ - } else { - return MA_FALSE; - } - } else { - break; /* Success. */ - } + /* Something happened. If the next buffer has been marked as done we need to reset a bit of state. */ + if ((pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwFlags & WHDR_DONE) != 0) { + pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwUser = 0; /* 0 = unlocked (make it available for reading). */ + pDevice->winmm.headerFramesConsumedCapture = 0; } - framesToSend = framesRead; - pBuffer = pDevice->alsa.pIntermediaryBuffer; + /* If the device has been stopped we need to break. */ + if (ma_device__get_state(pDevice) != MA_STATE_STARTED) { + break; + } } - if (framesToSend > 0) { - ma_device__send_frames_to_client(pDevice, framesToSend, pBuffer); + if (pFramesRead != NULL) { + *pFramesRead = totalFramesRead; } - return MA_TRUE; + return result; } -#endif /* 0 */ -void ma_device_uninit__alsa(ma_device* pDevice) +static ma_result ma_device_main_loop__winmm(ma_device* pDevice) { - ma_assert(pDevice != NULL); - - if ((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture) { - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture); - } + ma_result result = MA_SUCCESS; + ma_bool32 exitLoop = MA_FALSE; + + MA_ASSERT(pDevice != NULL); - if ((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback) { - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback); - } -} + /* The capture device needs to be started immediately. */ + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + MMRESULT resultMM; + WAVEHDR* pWAVEHDR; + ma_uint32 iPeriod; -ma_result ma_device_init_by_type__alsa(ma_context* pContext, const ma_device_config* pConfig, ma_device_type deviceType, ma_device* pDevice) -{ - ma_result result; - ma_snd_pcm_t* pPCM; - ma_bool32 isUsingMMap; - ma_snd_pcm_format_t formatALSA; - ma_share_mode shareMode; - ma_device_id* pDeviceID; - ma_format internalFormat; - ma_uint32 internalChannels; - ma_uint32 internalSampleRate; - ma_channel internalChannelMap[MA_MAX_CHANNELS]; - ma_uint32 internalBufferSizeInFrames; - ma_uint32 internalPeriods; - ma_snd_pcm_hw_params_t* pHWParams; - ma_snd_pcm_sw_params_t* pSWParams; - ma_snd_pcm_uframes_t bufferBoundary; - float bufferSizeScaleFactor; + pWAVEHDR = (WAVEHDR*)pDevice->winmm.pWAVEHDRCapture; - ma_assert(pContext != NULL); - ma_assert(pConfig != NULL); - ma_assert(deviceType != ma_device_type_duplex); /* This function should only be called for playback _or_ capture, never duplex. */ - ma_assert(pDevice != NULL); + /* Make sure the event is reset to a non-signaled state to ensure we don't prematurely return from WaitForSingleObject(). */ + ResetEvent((HANDLE)pDevice->winmm.hEventCapture); - formatALSA = ma_convert_ma_format_to_alsa_format((deviceType == ma_device_type_capture) ? pConfig->capture.format : pConfig->playback.format); - shareMode = (deviceType == ma_device_type_capture) ? pConfig->capture.shareMode : pConfig->playback.shareMode; - pDeviceID = (deviceType == ma_device_type_capture) ? pConfig->capture.pDeviceID : pConfig->playback.pDeviceID; + /* To start the device we attach all of the buffers and then start it. As the buffers are filled with data we will get notifications. */ + for (iPeriod = 0; iPeriod < pDevice->capture.internalPeriods; ++iPeriod) { + resultMM = ((MA_PFN_waveInAddBuffer)pDevice->pContext->winmm.waveInAddBuffer)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((LPWAVEHDR)pDevice->winmm.pWAVEHDRCapture)[iPeriod], sizeof(WAVEHDR)); + if (resultMM != MMSYSERR_NOERROR) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WinMM] Failed to attach input buffers to capture device in preparation for capture.", ma_result_from_MMRESULT(resultMM)); + } - result = ma_context_open_pcm__alsa(pContext, shareMode, deviceType, pDeviceID, &pPCM); - if (result != MA_SUCCESS) { - return result; - } + /* Make sure all of the buffers start out locked. We don't want to access them until the backend tells us we can. */ + pWAVEHDR[iPeriod].dwUser = 1; /* 1 = locked. */ + } - /* If using the default buffer size we may want to apply some device-specific scaling for known devices that have peculiar latency characteristics */ - bufferSizeScaleFactor = 1; - if (pDevice->usingDefaultBufferSize) { - ma_snd_pcm_info_t* pInfo = (ma_snd_pcm_info_t*)calloc(1, ((ma_snd_pcm_info_sizeof_proc)pContext->alsa.snd_pcm_info_sizeof)()); - if (pInfo == NULL) { - return MA_OUT_OF_MEMORY; + /* Capture devices need to be explicitly started, unlike playback devices. */ + resultMM = ((MA_PFN_waveInStart)pDevice->pContext->winmm.waveInStart)((HWAVEIN)pDevice->winmm.hDeviceCapture); + if (resultMM != MMSYSERR_NOERROR) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WinMM] Failed to start backend device.", ma_result_from_MMRESULT(resultMM)); } + } - /* We may need to scale the size of the buffer depending on the device. */ - if (((ma_snd_pcm_info_proc)pContext->alsa.snd_pcm_info)(pPCM, pInfo) == 0) { - const char* deviceName = ((ma_snd_pcm_info_get_name_proc)pContext->alsa.snd_pcm_info_get_name)(pInfo); - if (deviceName != NULL) { - if (ma_strcmp(deviceName, "default") == 0) { - char** ppDeviceHints; - char** ppNextDeviceHint; - /* It's the default device. We need to use DESC from snd_device_name_hint(). */ - if (((ma_snd_device_name_hint_proc)pContext->alsa.snd_device_name_hint)(-1, "pcm", (void***)&ppDeviceHints) < 0) { - ma_free(pInfo); - return MA_NO_BACKEND; + while (ma_device__get_state(pDevice) == MA_STATE_STARTED && !exitLoop) { + switch (pDevice->type) + { + case ma_device_type_duplex: + { + /* The process is: device_read -> convert -> callback -> convert -> device_write */ + ma_uint32 totalCapturedDeviceFramesProcessed = 0; + ma_uint32 capturedDevicePeriodSizeInFrames = ma_min(pDevice->capture.internalPeriodSizeInFrames, pDevice->playback.internalPeriodSizeInFrames); + + while (totalCapturedDeviceFramesProcessed < capturedDevicePeriodSizeInFrames) { + ma_uint8 capturedDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint8 playbackDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 capturedDeviceDataCapInFrames = sizeof(capturedDeviceData) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + ma_uint32 playbackDeviceDataCapInFrames = sizeof(playbackDeviceData) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + ma_uint32 capturedDeviceFramesRemaining; + ma_uint32 capturedDeviceFramesProcessed; + ma_uint32 capturedDeviceFramesToProcess; + ma_uint32 capturedDeviceFramesToTryProcessing = capturedDevicePeriodSizeInFrames - totalCapturedDeviceFramesProcessed; + if (capturedDeviceFramesToTryProcessing > capturedDeviceDataCapInFrames) { + capturedDeviceFramesToTryProcessing = capturedDeviceDataCapInFrames; } - ppNextDeviceHint = ppDeviceHints; - while (*ppNextDeviceHint != NULL) { - char* NAME = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "NAME"); - char* DESC = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "DESC"); - char* IOID = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "IOID"); - - ma_bool32 foundDevice = MA_FALSE; - if ((deviceType == ma_device_type_playback && (IOID == NULL || ma_strcmp(IOID, "Output") == 0)) || - (deviceType == ma_device_type_capture && (IOID != NULL && ma_strcmp(IOID, "Input" ) == 0))) { - if (ma_strcmp(NAME, deviceName) == 0) { - bufferSizeScaleFactor = ma_find_default_buffer_size_scale__alsa(DESC); - foundDevice = MA_TRUE; - } - } + result = ma_device_read__winmm(pDevice, capturedDeviceData, capturedDeviceFramesToTryProcessing, &capturedDeviceFramesToProcess); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } - free(NAME); - free(DESC); - free(IOID); - ppNextDeviceHint += 1; + capturedDeviceFramesRemaining = capturedDeviceFramesToProcess; + capturedDeviceFramesProcessed = 0; - if (foundDevice) { + for (;;) { + ma_uint8 capturedClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint8 playbackClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 capturedClientDataCapInFrames = sizeof(capturedClientData) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); + ma_uint32 playbackClientDataCapInFrames = sizeof(playbackClientData) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); + ma_uint64 capturedClientFramesToProcessThisIteration = ma_min(capturedClientDataCapInFrames, playbackClientDataCapInFrames); + ma_uint64 capturedDeviceFramesToProcessThisIteration = capturedDeviceFramesRemaining; + ma_uint8* pRunningCapturedDeviceFrames = ma_offset_ptr(capturedDeviceData, capturedDeviceFramesProcessed * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); + + /* Convert capture data from device format to client format. */ + result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningCapturedDeviceFrames, &capturedDeviceFramesToProcessThisIteration, capturedClientData, &capturedClientFramesToProcessThisIteration); + if (result != MA_SUCCESS) { break; } - } - ((ma_snd_device_name_free_hint_proc)pContext->alsa.snd_device_name_free_hint)((void**)ppDeviceHints); - } else { - bufferSizeScaleFactor = ma_find_default_buffer_size_scale__alsa(deviceName); - } - } - } + /* + If we weren't able to generate any output frames it must mean we've exhaused all of our input. The only time this would not be the case is if capturedClientData was too small + which should never be the case when it's of the size MA_DATA_CONVERTER_STACK_BUFFER_SIZE. + */ + if (capturedClientFramesToProcessThisIteration == 0) { + break; + } - ma_free(pInfo); - } + ma_device__on_data(pDevice, playbackClientData, capturedClientData, (ma_uint32)capturedClientFramesToProcessThisIteration); /* Safe cast .*/ + capturedDeviceFramesProcessed += (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */ + capturedDeviceFramesRemaining -= (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */ - /* Hardware parameters. */ - pHWParams = (ma_snd_pcm_hw_params_t*)calloc(1, ((ma_snd_pcm_hw_params_sizeof_proc)pContext->alsa.snd_pcm_hw_params_sizeof)()); - if (pHWParams == NULL) { - return MA_OUT_OF_MEMORY; - } + /* At this point the playbackClientData buffer should be holding data that needs to be written to the device. */ + for (;;) { + ma_uint64 convertedClientFrameCount = capturedClientFramesToProcessThisIteration; + ma_uint64 convertedDeviceFrameCount = playbackDeviceDataCapInFrames; + result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, playbackClientData, &convertedClientFrameCount, playbackDeviceData, &convertedDeviceFrameCount); + if (result != MA_SUCCESS) { + break; + } - if (((ma_snd_pcm_hw_params_any_proc)pContext->alsa.snd_pcm_hw_params_any)(pPCM, pHWParams) < 0) { - ma_free(pHWParams); - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to initialize hardware parameters. snd_pcm_hw_params_any() failed.", MA_FAILED_TO_CONFIGURE_BACKEND_DEVICE); - } + result = ma_device_write__winmm(pDevice, playbackDeviceData, (ma_uint32)convertedDeviceFrameCount, NULL); /* Safe cast. */ + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } - /* MMAP Mode. Try using interleaved MMAP access. If this fails, fall back to standard readi/writei. */ - isUsingMMap = MA_FALSE; -#if 0 /* NOTE: MMAP mode temporarily disabled. */ - if (deviceType != ma_device_type_capture) { /* <-- Disabling MMAP mode for capture devices because I apparently do not have a device that supports it which means I can't test it... Contributions welcome. */ - if (!pConfig->alsa.noMMap && ma_device__is_async(pDevice)) { - if (((ma_snd_pcm_hw_params_set_access_proc)pContext->alsa.snd_pcm_hw_params_set_access)(pPCM, pHWParams, MA_SND_PCM_ACCESS_MMAP_INTERLEAVED) == 0) { - pDevice->alsa.isUsingMMap = MA_TRUE; - } - } - } -#endif + capturedClientFramesToProcessThisIteration -= (ma_uint32)convertedClientFrameCount; /* Safe cast. */ + if (capturedClientFramesToProcessThisIteration == 0) { + break; + } + } - if (!isUsingMMap) { - if (((ma_snd_pcm_hw_params_set_access_proc)pContext->alsa.snd_pcm_hw_params_set_access)(pPCM, pHWParams, MA_SND_PCM_ACCESS_RW_INTERLEAVED) < 0) { - ma_free(pHWParams); - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set access mode to neither SND_PCM_ACCESS_MMAP_INTERLEAVED nor SND_PCM_ACCESS_RW_INTERLEAVED. snd_pcm_hw_params_set_access() failed.", MA_FORMAT_NOT_SUPPORTED); - } - } + /* In case an error happened from ma_device_write__winmm()... */ + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } + } - /* - Most important properties first. The documentation for OSS (yes, I know this is ALSA!) recommends format, channels, then sample rate. I can't - find any documentation for ALSA specifically, so I'm going to copy the recommendation for OSS. - */ + totalCapturedDeviceFramesProcessed += capturedDeviceFramesProcessed; + } + } break; - /* Format. */ - { - ma_snd_pcm_format_mask_t* pFormatMask; + case ma_device_type_capture: + { + /* We read in chunks of the period size, but use a stack allocated buffer for the intermediary. */ + ma_uint8 intermediaryBuffer[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + ma_uint32 periodSizeInFrames = pDevice->capture.internalPeriodSizeInFrames; + ma_uint32 framesReadThisPeriod = 0; + while (framesReadThisPeriod < periodSizeInFrames) { + ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesReadThisPeriod; + ma_uint32 framesProcessed; + ma_uint32 framesToReadThisIteration = framesRemainingInPeriod; + if (framesToReadThisIteration > intermediaryBufferSizeInFrames) { + framesToReadThisIteration = intermediaryBufferSizeInFrames; + } - /* Try getting every supported format first. */ - pFormatMask = (ma_snd_pcm_format_mask_t*)calloc(1, ((ma_snd_pcm_format_mask_sizeof_proc)pContext->alsa.snd_pcm_format_mask_sizeof)()); - if (pFormatMask == NULL) { - ma_free(pHWParams); - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); - return MA_OUT_OF_MEMORY; - } + result = ma_device_read__winmm(pDevice, intermediaryBuffer, framesToReadThisIteration, &framesProcessed); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } - ((ma_snd_pcm_hw_params_get_format_mask_proc)pContext->alsa.snd_pcm_hw_params_get_format_mask)(pHWParams, pFormatMask); + ma_device__send_frames_to_client(pDevice, framesProcessed, intermediaryBuffer); - /* - At this point we should have a list of supported formats, so now we need to find the best one. We first check if the requested format is - supported, and if so, use that one. If it's not supported, we just run though a list of formats and try to find the best one. - */ - if (!((ma_snd_pcm_format_mask_test_proc)pContext->alsa.snd_pcm_format_mask_test)(pFormatMask, formatALSA)) { - size_t i; + framesReadThisPeriod += framesProcessed; + } + } break; - /* The requested format is not supported so now try running through the list of formats and return the best one. */ - ma_snd_pcm_format_t preferredFormatsALSA[] = { - MA_SND_PCM_FORMAT_S16_LE, /* ma_format_s16 */ - MA_SND_PCM_FORMAT_FLOAT_LE, /* ma_format_f32 */ - MA_SND_PCM_FORMAT_S32_LE, /* ma_format_s32 */ - MA_SND_PCM_FORMAT_S24_3LE, /* ma_format_s24 */ - MA_SND_PCM_FORMAT_U8 /* ma_format_u8 */ - }; + case ma_device_type_playback: + { + /* We write in chunks of the period size, but use a stack allocated buffer for the intermediary. */ + ma_uint8 intermediaryBuffer[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + ma_uint32 periodSizeInFrames = pDevice->playback.internalPeriodSizeInFrames; + ma_uint32 framesWrittenThisPeriod = 0; + while (framesWrittenThisPeriod < periodSizeInFrames) { + ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesWrittenThisPeriod; + ma_uint32 framesProcessed; + ma_uint32 framesToWriteThisIteration = framesRemainingInPeriod; + if (framesToWriteThisIteration > intermediaryBufferSizeInFrames) { + framesToWriteThisIteration = intermediaryBufferSizeInFrames; + } - if (ma_is_big_endian()) { - preferredFormatsALSA[0] = MA_SND_PCM_FORMAT_S16_BE; - preferredFormatsALSA[1] = MA_SND_PCM_FORMAT_FLOAT_BE; - preferredFormatsALSA[2] = MA_SND_PCM_FORMAT_S32_BE; - preferredFormatsALSA[3] = MA_SND_PCM_FORMAT_S24_3BE; - preferredFormatsALSA[4] = MA_SND_PCM_FORMAT_U8; - } + ma_device__read_frames_from_client(pDevice, framesToWriteThisIteration, intermediaryBuffer); - formatALSA = MA_SND_PCM_FORMAT_UNKNOWN; - for (i = 0; i < (sizeof(preferredFormatsALSA) / sizeof(preferredFormatsALSA[0])); ++i) { - if (((ma_snd_pcm_format_mask_test_proc)pContext->alsa.snd_pcm_format_mask_test)(pFormatMask, preferredFormatsALSA[i])) { - formatALSA = preferredFormatsALSA[i]; - break; + result = ma_device_write__winmm(pDevice, intermediaryBuffer, framesToWriteThisIteration, &framesProcessed); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } + + framesWrittenThisPeriod += framesProcessed; } - } + } break; - if (formatALSA == MA_SND_PCM_FORMAT_UNKNOWN) { - ma_free(pHWParams); - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Format not supported. The device does not support any miniaudio formats.", MA_FORMAT_NOT_SUPPORTED); - } + /* To silence a warning. Will never hit this. */ + case ma_device_type_loopback: + default: break; } + } - ma_free(pFormatMask); - pFormatMask = NULL; - if (((ma_snd_pcm_hw_params_set_format_proc)pContext->alsa.snd_pcm_hw_params_set_format)(pPCM, pHWParams, formatALSA) < 0) { - ma_free(pHWParams); - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Format not supported. snd_pcm_hw_params_set_format() failed.", MA_FORMAT_NOT_SUPPORTED); - } - - internalFormat = ma_format_from_alsa(formatALSA); - if (internalFormat == ma_format_unknown) { - ma_free(pHWParams); - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] The chosen format is not supported by miniaudio.", MA_FORMAT_NOT_SUPPORTED); - } - } + /* Here is where the device is started. */ + ma_device_stop__winmm(pDevice); - /* Channels. */ - { - unsigned int channels = (deviceType == ma_device_type_capture) ? pConfig->capture.channels : pConfig->playback.channels; - if (((ma_snd_pcm_hw_params_set_channels_near_proc)pContext->alsa.snd_pcm_hw_params_set_channels_near)(pPCM, pHWParams, &channels) < 0) { - ma_free(pHWParams); - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set channel count. snd_pcm_hw_params_set_channels_near() failed.", MA_FORMAT_NOT_SUPPORTED); - } - internalChannels = (ma_uint32)channels; - } + return result; +} - /* Sample Rate */ - { - unsigned int sampleRate; +static ma_result ma_context_uninit__winmm(ma_context* pContext) +{ + MA_ASSERT(pContext != NULL); + MA_ASSERT(pContext->backend == ma_backend_winmm); - /* - It appears there's either a bug in ALSA, a bug in some drivers, or I'm doing something silly; but having resampling enabled causes - problems with some device configurations when used in conjunction with MMAP access mode. To fix this problem we need to disable - resampling. - - To reproduce this problem, open the "plug:dmix" device, and set the sample rate to 44100. Internally, it looks like dmix uses a - sample rate of 48000. The hardware parameters will get set correctly with no errors, but it looks like the 44100 -> 48000 resampling - doesn't work properly - but only with MMAP access mode. You will notice skipping/crackling in the audio, and it'll run at a slightly - faster rate. - - miniaudio has built-in support for sample rate conversion (albeit low quality at the moment), so disabling resampling should be fine - for us. The only problem is that it won't be taking advantage of any kind of hardware-accelerated resampling and it won't be very - good quality until I get a chance to improve the quality of miniaudio's software sample rate conversion. - - I don't currently know if the dmix plugin is the only one with this error. Indeed, this is the only one I've been able to reproduce - this error with. In the future, we may want to restrict the disabling of resampling to only known bad plugins. - */ - ((ma_snd_pcm_hw_params_set_rate_resample_proc)pContext->alsa.snd_pcm_hw_params_set_rate_resample)(pPCM, pHWParams, 0); + ma_dlclose(pContext, pContext->winmm.hWinMM); + return MA_SUCCESS; +} - sampleRate = pConfig->sampleRate; - if (((ma_snd_pcm_hw_params_set_rate_near_proc)pContext->alsa.snd_pcm_hw_params_set_rate_near)(pPCM, pHWParams, &sampleRate, 0) < 0) { - ma_free(pHWParams); - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Sample rate not supported. snd_pcm_hw_params_set_rate_near() failed.", MA_FORMAT_NOT_SUPPORTED); - } - internalSampleRate = (ma_uint32)sampleRate; - } +static ma_result ma_context_init__winmm(const ma_context_config* pConfig, ma_context* pContext) +{ + MA_ASSERT(pContext != NULL); - /* Buffer Size */ - { - ma_snd_pcm_uframes_t actualBufferSizeInFrames = pConfig->bufferSizeInFrames; - if (actualBufferSizeInFrames == 0) { - actualBufferSizeInFrames = ma_scale_buffer_size(ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->bufferSizeInMilliseconds, internalSampleRate), bufferSizeScaleFactor); - } + (void)pConfig; - if (((ma_snd_pcm_hw_params_set_buffer_size_near_proc)pContext->alsa.snd_pcm_hw_params_set_buffer_size_near)(pPCM, pHWParams, &actualBufferSizeInFrames) < 0) { - ma_free(pHWParams); - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set buffer size for device. snd_pcm_hw_params_set_buffer_size() failed.", MA_FORMAT_NOT_SUPPORTED); - } - internalBufferSizeInFrames = actualBufferSizeInFrames; + pContext->winmm.hWinMM = ma_dlopen(pContext, "winmm.dll"); + if (pContext->winmm.hWinMM == NULL) { + return MA_NO_BACKEND; } - /* Periods. */ - { - ma_uint32 periods = pConfig->periods; - if (((ma_snd_pcm_hw_params_set_periods_near_proc)pContext->alsa.snd_pcm_hw_params_set_periods_near)(pPCM, pHWParams, &periods, NULL) < 0) { - ma_free(pHWParams); - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set period count. snd_pcm_hw_params_set_periods_near() failed.", MA_FORMAT_NOT_SUPPORTED); - } - internalPeriods = periods; - } + pContext->winmm.waveOutGetNumDevs = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutGetNumDevs"); + pContext->winmm.waveOutGetDevCapsA = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutGetDevCapsA"); + pContext->winmm.waveOutOpen = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutOpen"); + pContext->winmm.waveOutClose = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutClose"); + pContext->winmm.waveOutPrepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutPrepareHeader"); + pContext->winmm.waveOutUnprepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutUnprepareHeader"); + pContext->winmm.waveOutWrite = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutWrite"); + pContext->winmm.waveOutReset = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutReset"); + pContext->winmm.waveInGetNumDevs = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInGetNumDevs"); + pContext->winmm.waveInGetDevCapsA = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInGetDevCapsA"); + pContext->winmm.waveInOpen = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInOpen"); + pContext->winmm.waveInClose = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInClose"); + pContext->winmm.waveInPrepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInPrepareHeader"); + pContext->winmm.waveInUnprepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInUnprepareHeader"); + pContext->winmm.waveInAddBuffer = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInAddBuffer"); + pContext->winmm.waveInStart = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInStart"); + pContext->winmm.waveInReset = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInReset"); - /* Apply hardware parameters. */ - if (((ma_snd_pcm_hw_params_proc)pContext->alsa.snd_pcm_hw_params)(pPCM, pHWParams) < 0) { - ma_free(pHWParams); - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set hardware parameters. snd_pcm_hw_params() failed.", MA_FAILED_TO_CONFIGURE_BACKEND_DEVICE); - } + pContext->onUninit = ma_context_uninit__winmm; + pContext->onDeviceIDEqual = ma_context_is_device_id_equal__winmm; + pContext->onEnumDevices = ma_context_enumerate_devices__winmm; + pContext->onGetDeviceInfo = ma_context_get_device_info__winmm; + pContext->onDeviceInit = ma_device_init__winmm; + pContext->onDeviceUninit = ma_device_uninit__winmm; + pContext->onDeviceStart = NULL; /* Not used with synchronous backends. */ + pContext->onDeviceStop = NULL; /* Not used with synchronous backends. */ + pContext->onDeviceMainLoop = ma_device_main_loop__winmm; - ma_free(pHWParams); - pHWParams = NULL; + return MA_SUCCESS; +} +#endif - /* Software parameters. */ - pSWParams = (ma_snd_pcm_sw_params_t*)calloc(1, ((ma_snd_pcm_sw_params_sizeof_proc)pContext->alsa.snd_pcm_sw_params_sizeof)()); - if (pSWParams == NULL) { - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); - return MA_OUT_OF_MEMORY; - } - if (((ma_snd_pcm_sw_params_current_proc)pContext->alsa.snd_pcm_sw_params_current)(pPCM, pSWParams) != 0) { - ma_free(pSWParams); - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to initialize software parameters. snd_pcm_sw_params_current() failed.", MA_FAILED_TO_CONFIGURE_BACKEND_DEVICE); - } - if (((ma_snd_pcm_sw_params_set_avail_min_proc)pContext->alsa.snd_pcm_sw_params_set_avail_min)(pPCM, pSWParams, ma_prev_power_of_2(internalBufferSizeInFrames/internalPeriods)) != 0) { - ma_free(pSWParams); - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_sw_params_set_avail_min() failed.", MA_FORMAT_NOT_SUPPORTED); - } +/****************************************************************************** - if (((ma_snd_pcm_sw_params_get_boundary_proc)pContext->alsa.snd_pcm_sw_params_get_boundary)(pSWParams, &bufferBoundary) < 0) { - bufferBoundary = internalBufferSizeInFrames; - } +ALSA Backend - /*printf("TRACE: bufferBoundary=%ld\n", bufferBoundary);*/ +******************************************************************************/ +#ifdef MA_HAS_ALSA - if (deviceType == ma_device_type_playback && !isUsingMMap) { /* Only playback devices in writei/readi mode need a start threshold. */ - /* - Subtle detail here with the start threshold. When in playback-only mode (no full-duplex) we can set the start threshold to - the size of a period. But for full-duplex we need to set it such that it is at least two periods. - */ - if (((ma_snd_pcm_sw_params_set_start_threshold_proc)pContext->alsa.snd_pcm_sw_params_set_start_threshold)(pPCM, pSWParams, (internalBufferSizeInFrames/internalPeriods)*2) != 0) { - ma_free(pSWParams); - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set start threshold for playback device. snd_pcm_sw_params_set_start_threshold() failed.", MA_FAILED_TO_CONFIGURE_BACKEND_DEVICE); - } - if (((ma_snd_pcm_sw_params_set_stop_threshold_proc)pContext->alsa.snd_pcm_sw_params_set_stop_threshold)(pPCM, pSWParams, bufferBoundary) != 0) { /* Set to boundary to loop instead of stop in the event of an xrun. */ - ma_free(pSWParams); - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set stop threshold for playback device. snd_pcm_sw_params_set_stop_threshold() failed.", MA_FAILED_TO_CONFIGURE_BACKEND_DEVICE); - } - } +#ifdef MA_NO_RUNTIME_LINKING +#include +typedef snd_pcm_uframes_t ma_snd_pcm_uframes_t; +typedef snd_pcm_sframes_t ma_snd_pcm_sframes_t; +typedef snd_pcm_stream_t ma_snd_pcm_stream_t; +typedef snd_pcm_format_t ma_snd_pcm_format_t; +typedef snd_pcm_access_t ma_snd_pcm_access_t; +typedef snd_pcm_t ma_snd_pcm_t; +typedef snd_pcm_hw_params_t ma_snd_pcm_hw_params_t; +typedef snd_pcm_sw_params_t ma_snd_pcm_sw_params_t; +typedef snd_pcm_format_mask_t ma_snd_pcm_format_mask_t; +typedef snd_pcm_info_t ma_snd_pcm_info_t; +typedef snd_pcm_channel_area_t ma_snd_pcm_channel_area_t; +typedef snd_pcm_chmap_t ma_snd_pcm_chmap_t; - if (((ma_snd_pcm_sw_params_proc)pContext->alsa.snd_pcm_sw_params)(pPCM, pSWParams) != 0) { - ma_free(pSWParams); - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set software parameters. snd_pcm_sw_params() failed.", MA_FAILED_TO_CONFIGURE_BACKEND_DEVICE); - } +/* snd_pcm_stream_t */ +#define MA_SND_PCM_STREAM_PLAYBACK SND_PCM_STREAM_PLAYBACK +#define MA_SND_PCM_STREAM_CAPTURE SND_PCM_STREAM_CAPTURE - ma_free(pSWParams); - pSWParams = NULL; +/* snd_pcm_format_t */ +#define MA_SND_PCM_FORMAT_UNKNOWN SND_PCM_FORMAT_UNKNOWN +#define MA_SND_PCM_FORMAT_U8 SND_PCM_FORMAT_U8 +#define MA_SND_PCM_FORMAT_S16_LE SND_PCM_FORMAT_S16_LE +#define MA_SND_PCM_FORMAT_S16_BE SND_PCM_FORMAT_S16_BE +#define MA_SND_PCM_FORMAT_S24_LE SND_PCM_FORMAT_S24_LE +#define MA_SND_PCM_FORMAT_S24_BE SND_PCM_FORMAT_S24_BE +#define MA_SND_PCM_FORMAT_S32_LE SND_PCM_FORMAT_S32_LE +#define MA_SND_PCM_FORMAT_S32_BE SND_PCM_FORMAT_S32_BE +#define MA_SND_PCM_FORMAT_FLOAT_LE SND_PCM_FORMAT_FLOAT_LE +#define MA_SND_PCM_FORMAT_FLOAT_BE SND_PCM_FORMAT_FLOAT_BE +#define MA_SND_PCM_FORMAT_FLOAT64_LE SND_PCM_FORMAT_FLOAT64_LE +#define MA_SND_PCM_FORMAT_FLOAT64_BE SND_PCM_FORMAT_FLOAT64_BE +#define MA_SND_PCM_FORMAT_MU_LAW SND_PCM_FORMAT_MU_LAW +#define MA_SND_PCM_FORMAT_A_LAW SND_PCM_FORMAT_A_LAW +#define MA_SND_PCM_FORMAT_S24_3LE SND_PCM_FORMAT_S24_3LE +#define MA_SND_PCM_FORMAT_S24_3BE SND_PCM_FORMAT_S24_3BE +/* ma_snd_pcm_access_t */ +#define MA_SND_PCM_ACCESS_MMAP_INTERLEAVED SND_PCM_ACCESS_MMAP_INTERLEAVED +#define MA_SND_PCM_ACCESS_MMAP_NONINTERLEAVED SND_PCM_ACCESS_MMAP_NONINTERLEAVED +#define MA_SND_PCM_ACCESS_MMAP_COMPLEX SND_PCM_ACCESS_MMAP_COMPLEX +#define MA_SND_PCM_ACCESS_RW_INTERLEAVED SND_PCM_ACCESS_RW_INTERLEAVED +#define MA_SND_PCM_ACCESS_RW_NONINTERLEAVED SND_PCM_ACCESS_RW_NONINTERLEAVED - /* Grab the internal channel map. For now we're not going to bother trying to change the channel map and instead just do it ourselves. */ - { - ma_snd_pcm_chmap_t* pChmap = ((ma_snd_pcm_get_chmap_proc)pContext->alsa.snd_pcm_get_chmap)(pPCM); - if (pChmap != NULL) { - ma_uint32 iChannel; +/* Channel positions. */ +#define MA_SND_CHMAP_UNKNOWN SND_CHMAP_UNKNOWN +#define MA_SND_CHMAP_NA SND_CHMAP_NA +#define MA_SND_CHMAP_MONO SND_CHMAP_MONO +#define MA_SND_CHMAP_FL SND_CHMAP_FL +#define MA_SND_CHMAP_FR SND_CHMAP_FR +#define MA_SND_CHMAP_RL SND_CHMAP_RL +#define MA_SND_CHMAP_RR SND_CHMAP_RR +#define MA_SND_CHMAP_FC SND_CHMAP_FC +#define MA_SND_CHMAP_LFE SND_CHMAP_LFE +#define MA_SND_CHMAP_SL SND_CHMAP_SL +#define MA_SND_CHMAP_SR SND_CHMAP_SR +#define MA_SND_CHMAP_RC SND_CHMAP_RC +#define MA_SND_CHMAP_FLC SND_CHMAP_FLC +#define MA_SND_CHMAP_FRC SND_CHMAP_FRC +#define MA_SND_CHMAP_RLC SND_CHMAP_RLC +#define MA_SND_CHMAP_RRC SND_CHMAP_RRC +#define MA_SND_CHMAP_FLW SND_CHMAP_FLW +#define MA_SND_CHMAP_FRW SND_CHMAP_FRW +#define MA_SND_CHMAP_FLH SND_CHMAP_FLH +#define MA_SND_CHMAP_FCH SND_CHMAP_FCH +#define MA_SND_CHMAP_FRH SND_CHMAP_FRH +#define MA_SND_CHMAP_TC SND_CHMAP_TC +#define MA_SND_CHMAP_TFL SND_CHMAP_TFL +#define MA_SND_CHMAP_TFR SND_CHMAP_TFR +#define MA_SND_CHMAP_TFC SND_CHMAP_TFC +#define MA_SND_CHMAP_TRL SND_CHMAP_TRL +#define MA_SND_CHMAP_TRR SND_CHMAP_TRR +#define MA_SND_CHMAP_TRC SND_CHMAP_TRC +#define MA_SND_CHMAP_TFLC SND_CHMAP_TFLC +#define MA_SND_CHMAP_TFRC SND_CHMAP_TFRC +#define MA_SND_CHMAP_TSL SND_CHMAP_TSL +#define MA_SND_CHMAP_TSR SND_CHMAP_TSR +#define MA_SND_CHMAP_LLFE SND_CHMAP_LLFE +#define MA_SND_CHMAP_RLFE SND_CHMAP_RLFE +#define MA_SND_CHMAP_BC SND_CHMAP_BC +#define MA_SND_CHMAP_BLC SND_CHMAP_BLC +#define MA_SND_CHMAP_BRC SND_CHMAP_BRC - /* There are cases where the returned channel map can have a different channel count than was returned by snd_pcm_hw_params_set_channels_near(). */ - if (pChmap->channels >= internalChannels) { - /* Drop excess channels. */ - for (iChannel = 0; iChannel < internalChannels; ++iChannel) { - internalChannelMap[iChannel] = ma_convert_alsa_channel_position_to_ma_channel(pChmap->pos[iChannel]); - } - } else { - ma_uint32 i; +/* Open mode flags. */ +#define MA_SND_PCM_NO_AUTO_RESAMPLE SND_PCM_NO_AUTO_RESAMPLE +#define MA_SND_PCM_NO_AUTO_CHANNELS SND_PCM_NO_AUTO_CHANNELS +#define MA_SND_PCM_NO_AUTO_FORMAT SND_PCM_NO_AUTO_FORMAT +#else +#include /* For EPIPE, etc. */ +typedef unsigned long ma_snd_pcm_uframes_t; +typedef long ma_snd_pcm_sframes_t; +typedef int ma_snd_pcm_stream_t; +typedef int ma_snd_pcm_format_t; +typedef int ma_snd_pcm_access_t; +typedef struct ma_snd_pcm_t ma_snd_pcm_t; +typedef struct ma_snd_pcm_hw_params_t ma_snd_pcm_hw_params_t; +typedef struct ma_snd_pcm_sw_params_t ma_snd_pcm_sw_params_t; +typedef struct ma_snd_pcm_format_mask_t ma_snd_pcm_format_mask_t; +typedef struct ma_snd_pcm_info_t ma_snd_pcm_info_t; +typedef struct +{ + void* addr; + unsigned int first; + unsigned int step; +} ma_snd_pcm_channel_area_t; +typedef struct +{ + unsigned int channels; + unsigned int pos[1]; +} ma_snd_pcm_chmap_t; - /* - Excess channels use defaults. Do an initial fill with defaults, overwrite the first pChmap->channels, validate to ensure there are no duplicate - channels. If validation fails, fall back to defaults. - */ - ma_bool32 isValid = MA_TRUE; +/* snd_pcm_state_t */ +#define MA_SND_PCM_STATE_OPEN 0 +#define MA_SND_PCM_STATE_SETUP 1 +#define MA_SND_PCM_STATE_PREPARED 2 +#define MA_SND_PCM_STATE_RUNNING 3 +#define MA_SND_PCM_STATE_XRUN 4 +#define MA_SND_PCM_STATE_DRAINING 5 +#define MA_SND_PCM_STATE_PAUSED 6 +#define MA_SND_PCM_STATE_SUSPENDED 7 +#define MA_SND_PCM_STATE_DISCONNECTED 8 - /* Fill with defaults. */ - ma_get_standard_channel_map(ma_standard_channel_map_alsa, internalChannels, internalChannelMap); +/* snd_pcm_stream_t */ +#define MA_SND_PCM_STREAM_PLAYBACK 0 +#define MA_SND_PCM_STREAM_CAPTURE 1 - /* Overwrite first pChmap->channels channels. */ - for (iChannel = 0; iChannel < pChmap->channels; ++iChannel) { - internalChannelMap[iChannel] = ma_convert_alsa_channel_position_to_ma_channel(pChmap->pos[iChannel]); - } +/* snd_pcm_format_t */ +#define MA_SND_PCM_FORMAT_UNKNOWN -1 +#define MA_SND_PCM_FORMAT_U8 1 +#define MA_SND_PCM_FORMAT_S16_LE 2 +#define MA_SND_PCM_FORMAT_S16_BE 3 +#define MA_SND_PCM_FORMAT_S24_LE 6 +#define MA_SND_PCM_FORMAT_S24_BE 7 +#define MA_SND_PCM_FORMAT_S32_LE 10 +#define MA_SND_PCM_FORMAT_S32_BE 11 +#define MA_SND_PCM_FORMAT_FLOAT_LE 14 +#define MA_SND_PCM_FORMAT_FLOAT_BE 15 +#define MA_SND_PCM_FORMAT_FLOAT64_LE 16 +#define MA_SND_PCM_FORMAT_FLOAT64_BE 17 +#define MA_SND_PCM_FORMAT_MU_LAW 20 +#define MA_SND_PCM_FORMAT_A_LAW 21 +#define MA_SND_PCM_FORMAT_S24_3LE 32 +#define MA_SND_PCM_FORMAT_S24_3BE 33 - /* Validate. */ - for (i = 0; i < internalChannels && isValid; ++i) { - ma_uint32 j; - for (j = i+1; j < internalChannels; ++j) { - if (internalChannelMap[i] == internalChannelMap[j]) { - isValid = MA_FALSE; - break; - } - } - } +/* snd_pcm_access_t */ +#define MA_SND_PCM_ACCESS_MMAP_INTERLEAVED 0 +#define MA_SND_PCM_ACCESS_MMAP_NONINTERLEAVED 1 +#define MA_SND_PCM_ACCESS_MMAP_COMPLEX 2 +#define MA_SND_PCM_ACCESS_RW_INTERLEAVED 3 +#define MA_SND_PCM_ACCESS_RW_NONINTERLEAVED 4 - /* If our channel map is invalid, fall back to defaults. */ - if (!isValid) { - ma_get_standard_channel_map(ma_standard_channel_map_alsa, internalChannels, internalChannelMap); - } - } +/* Channel positions. */ +#define MA_SND_CHMAP_UNKNOWN 0 +#define MA_SND_CHMAP_NA 1 +#define MA_SND_CHMAP_MONO 2 +#define MA_SND_CHMAP_FL 3 +#define MA_SND_CHMAP_FR 4 +#define MA_SND_CHMAP_RL 5 +#define MA_SND_CHMAP_RR 6 +#define MA_SND_CHMAP_FC 7 +#define MA_SND_CHMAP_LFE 8 +#define MA_SND_CHMAP_SL 9 +#define MA_SND_CHMAP_SR 10 +#define MA_SND_CHMAP_RC 11 +#define MA_SND_CHMAP_FLC 12 +#define MA_SND_CHMAP_FRC 13 +#define MA_SND_CHMAP_RLC 14 +#define MA_SND_CHMAP_RRC 15 +#define MA_SND_CHMAP_FLW 16 +#define MA_SND_CHMAP_FRW 17 +#define MA_SND_CHMAP_FLH 18 +#define MA_SND_CHMAP_FCH 19 +#define MA_SND_CHMAP_FRH 20 +#define MA_SND_CHMAP_TC 21 +#define MA_SND_CHMAP_TFL 22 +#define MA_SND_CHMAP_TFR 23 +#define MA_SND_CHMAP_TFC 24 +#define MA_SND_CHMAP_TRL 25 +#define MA_SND_CHMAP_TRR 26 +#define MA_SND_CHMAP_TRC 27 +#define MA_SND_CHMAP_TFLC 28 +#define MA_SND_CHMAP_TFRC 29 +#define MA_SND_CHMAP_TSL 30 +#define MA_SND_CHMAP_TSR 31 +#define MA_SND_CHMAP_LLFE 32 +#define MA_SND_CHMAP_RLFE 33 +#define MA_SND_CHMAP_BC 34 +#define MA_SND_CHMAP_BLC 35 +#define MA_SND_CHMAP_BRC 36 - free(pChmap); - pChmap = NULL; - } else { - /* Could not retrieve the channel map. Fall back to a hard-coded assumption. */ - ma_get_standard_channel_map(ma_standard_channel_map_alsa, internalChannels, internalChannelMap); - } - } +/* Open mode flags. */ +#define MA_SND_PCM_NO_AUTO_RESAMPLE 0x00010000 +#define MA_SND_PCM_NO_AUTO_CHANNELS 0x00020000 +#define MA_SND_PCM_NO_AUTO_FORMAT 0x00040000 +#endif +typedef int (* ma_snd_pcm_open_proc) (ma_snd_pcm_t **pcm, const char *name, ma_snd_pcm_stream_t stream, int mode); +typedef int (* ma_snd_pcm_close_proc) (ma_snd_pcm_t *pcm); +typedef size_t (* ma_snd_pcm_hw_params_sizeof_proc) (void); +typedef int (* ma_snd_pcm_hw_params_any_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params); +typedef int (* ma_snd_pcm_hw_params_set_format_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t val); +typedef int (* ma_snd_pcm_hw_params_set_format_first_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t *format); +typedef void (* ma_snd_pcm_hw_params_get_format_mask_proc) (ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_mask_t *mask); +typedef int (* ma_snd_pcm_hw_params_set_channels_near_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *val); +typedef int (* ma_snd_pcm_hw_params_set_rate_resample_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val); +typedef int (* ma_snd_pcm_hw_params_set_rate_near_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *val, int *dir); +typedef int (* ma_snd_pcm_hw_params_set_buffer_size_near_proc)(ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_uframes_t *val); +typedef int (* ma_snd_pcm_hw_params_set_periods_near_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *val, int *dir); +typedef int (* ma_snd_pcm_hw_params_set_access_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_access_t _access); +typedef int (* ma_snd_pcm_hw_params_get_format_proc) (const ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t *format); +typedef int (* ma_snd_pcm_hw_params_get_channels_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val); +typedef int (* ma_snd_pcm_hw_params_get_channels_min_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val); +typedef int (* ma_snd_pcm_hw_params_get_channels_max_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val); +typedef int (* ma_snd_pcm_hw_params_get_rate_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *rate, int *dir); +typedef int (* ma_snd_pcm_hw_params_get_rate_min_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *rate, int *dir); +typedef int (* ma_snd_pcm_hw_params_get_rate_max_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *rate, int *dir); +typedef int (* ma_snd_pcm_hw_params_get_buffer_size_proc) (const ma_snd_pcm_hw_params_t *params, ma_snd_pcm_uframes_t *val); +typedef int (* ma_snd_pcm_hw_params_get_periods_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val, int *dir); +typedef int (* ma_snd_pcm_hw_params_get_access_proc) (const ma_snd_pcm_hw_params_t *params, ma_snd_pcm_access_t *_access); +typedef int (* ma_snd_pcm_hw_params_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params); +typedef size_t (* ma_snd_pcm_sw_params_sizeof_proc) (void); +typedef int (* ma_snd_pcm_sw_params_current_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params); +typedef int (* ma_snd_pcm_sw_params_get_boundary_proc) (ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t* val); +typedef int (* ma_snd_pcm_sw_params_set_avail_min_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t val); +typedef int (* ma_snd_pcm_sw_params_set_start_threshold_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t val); +typedef int (* ma_snd_pcm_sw_params_set_stop_threshold_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t val); +typedef int (* ma_snd_pcm_sw_params_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params); +typedef size_t (* ma_snd_pcm_format_mask_sizeof_proc) (void); +typedef int (* ma_snd_pcm_format_mask_test_proc) (const ma_snd_pcm_format_mask_t *mask, ma_snd_pcm_format_t val); +typedef ma_snd_pcm_chmap_t * (* ma_snd_pcm_get_chmap_proc) (ma_snd_pcm_t *pcm); +typedef int (* ma_snd_pcm_state_proc) (ma_snd_pcm_t *pcm); +typedef int (* ma_snd_pcm_prepare_proc) (ma_snd_pcm_t *pcm); +typedef int (* ma_snd_pcm_start_proc) (ma_snd_pcm_t *pcm); +typedef int (* ma_snd_pcm_drop_proc) (ma_snd_pcm_t *pcm); +typedef int (* ma_snd_pcm_drain_proc) (ma_snd_pcm_t *pcm); +typedef int (* ma_snd_device_name_hint_proc) (int card, const char *iface, void ***hints); +typedef char * (* ma_snd_device_name_get_hint_proc) (const void *hint, const char *id); +typedef int (* ma_snd_card_get_index_proc) (const char *name); +typedef int (* ma_snd_device_name_free_hint_proc) (void **hints); +typedef int (* ma_snd_pcm_mmap_begin_proc) (ma_snd_pcm_t *pcm, const ma_snd_pcm_channel_area_t **areas, ma_snd_pcm_uframes_t *offset, ma_snd_pcm_uframes_t *frames); +typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_mmap_commit_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_uframes_t offset, ma_snd_pcm_uframes_t frames); +typedef int (* ma_snd_pcm_recover_proc) (ma_snd_pcm_t *pcm, int err, int silent); +typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_readi_proc) (ma_snd_pcm_t *pcm, void *buffer, ma_snd_pcm_uframes_t size); +typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_writei_proc) (ma_snd_pcm_t *pcm, const void *buffer, ma_snd_pcm_uframes_t size); +typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_avail_proc) (ma_snd_pcm_t *pcm); +typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_avail_update_proc) (ma_snd_pcm_t *pcm); +typedef int (* ma_snd_pcm_wait_proc) (ma_snd_pcm_t *pcm, int timeout); +typedef int (* ma_snd_pcm_info_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_info_t* info); +typedef size_t (* ma_snd_pcm_info_sizeof_proc) (); +typedef const char* (* ma_snd_pcm_info_get_name_proc) (const ma_snd_pcm_info_t* info); +typedef int (* ma_snd_config_update_free_global_proc) (); - /* We're done. Prepare the device. */ - if (((ma_snd_pcm_prepare_proc)pDevice->pContext->alsa.snd_pcm_prepare)(pPCM) < 0) { - ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to prepare device.", MA_FAILED_TO_START_BACKEND_DEVICE); - } +/* This array specifies each of the common devices that can be used for both playback and capture. */ +static const char* g_maCommonDeviceNamesALSA[] = { + "default", + "null", + "pulse", + "jack" +}; +/* This array allows us to blacklist specific playback devices. */ +static const char* g_maBlacklistedPlaybackDeviceNamesALSA[] = { + "" +}; - if (deviceType == ma_device_type_capture) { - pDevice->alsa.pPCMCapture = (ma_ptr)pPCM; - pDevice->alsa.isUsingMMapCapture = isUsingMMap; - pDevice->capture.internalFormat = internalFormat; - pDevice->capture.internalChannels = internalChannels; - pDevice->capture.internalSampleRate = internalSampleRate; - ma_channel_map_copy(pDevice->capture.internalChannelMap, internalChannelMap, internalChannels); - pDevice->capture.internalBufferSizeInFrames = internalBufferSizeInFrames; - pDevice->capture.internalPeriods = internalPeriods; - } else { - pDevice->alsa.pPCMPlayback = (ma_ptr)pPCM; - pDevice->alsa.isUsingMMapPlayback = isUsingMMap; - pDevice->playback.internalFormat = internalFormat; - pDevice->playback.internalChannels = internalChannels; - pDevice->playback.internalSampleRate = internalSampleRate; - ma_channel_map_copy(pDevice->playback.internalChannelMap, internalChannelMap, internalChannels); - pDevice->playback.internalBufferSizeInFrames = internalBufferSizeInFrames; - pDevice->playback.internalPeriods = internalPeriods; - } +/* This array allows us to blacklist specific capture devices. */ +static const char* g_maBlacklistedCaptureDeviceNamesALSA[] = { + "" +}; - return MA_SUCCESS; -} -ma_result ma_device_init__alsa(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) +/* +This array allows miniaudio to control device-specific default buffer sizes. This uses a scaling factor. Order is important. If +any part of the string is present in the device's name, the associated scale will be used. +*/ +static struct { - ma_assert(pDevice != NULL); - - ma_zero_object(&pDevice->alsa); + const char* name; + float scale; +} g_maDefaultBufferSizeScalesALSA[] = { + {"bcm2835 IEC958/HDMI", 2.0f}, + {"bcm2835 ALSA", 2.0f} +}; - if (pConfig->deviceType == ma_device_type_loopback) { - return MA_DEVICE_TYPE_NOT_SUPPORTED; - } +static float ma_find_default_buffer_size_scale__alsa(const char* deviceName) +{ + size_t i; - if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { - ma_result result = ma_device_init_by_type__alsa(pContext, pConfig, ma_device_type_capture, pDevice); - if (result != MA_SUCCESS) { - return result; - } + if (deviceName == NULL) { + return 1; } - if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { - ma_result result = ma_device_init_by_type__alsa(pContext, pConfig, ma_device_type_playback, pDevice); - if (result != MA_SUCCESS) { - return result; + for (i = 0; i < ma_countof(g_maDefaultBufferSizeScalesALSA); ++i) { + if (strstr(g_maDefaultBufferSizeScalesALSA[i].name, deviceName) != NULL) { + return g_maDefaultBufferSizeScalesALSA[i].scale; } } - return MA_SUCCESS; + return 1; } -#if 0 -ma_result ma_device_start__alsa(ma_device* pDevice) +static ma_snd_pcm_format_t ma_convert_ma_format_to_alsa_format(ma_format format) { - ma_assert(pDevice != NULL); - - /* Prepare the device first... */ - if (((ma_snd_pcm_prepare_proc)pDevice->pContext->alsa.snd_pcm_prepare)((ma_snd_pcm_t*)pDevice->alsa.pPCM) < 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to prepare device.", MA_FAILED_TO_START_BACKEND_DEVICE); - } - - /* - ... and then grab an initial chunk from the client. After this is done, the device should - automatically start playing, since that's how we configured the software parameters. - */ - if (pDevice->type == ma_device_type_playback) { - if (!ma_device_read_from_client_and_write__alsa(pDevice)) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to write initial chunk of data to the playback device.", MA_FAILED_TO_SEND_DATA_TO_DEVICE); - } + ma_snd_pcm_format_t ALSAFormats[] = { + MA_SND_PCM_FORMAT_UNKNOWN, /* ma_format_unknown */ + MA_SND_PCM_FORMAT_U8, /* ma_format_u8 */ + MA_SND_PCM_FORMAT_S16_LE, /* ma_format_s16 */ + MA_SND_PCM_FORMAT_S24_3LE, /* ma_format_s24 */ + MA_SND_PCM_FORMAT_S32_LE, /* ma_format_s32 */ + MA_SND_PCM_FORMAT_FLOAT_LE /* ma_format_f32 */ + }; - /* mmap mode requires an explicit start. */ - if (pDevice->alsa.isUsingMMap) { - if (((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCM) < 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start capture device.", MA_FAILED_TO_START_BACKEND_DEVICE); - } - } - } else { - if (((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCM) < 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start capture device.", MA_FAILED_TO_START_BACKEND_DEVICE); - } + if (ma_is_big_endian()) { + ALSAFormats[0] = MA_SND_PCM_FORMAT_UNKNOWN; + ALSAFormats[1] = MA_SND_PCM_FORMAT_U8; + ALSAFormats[2] = MA_SND_PCM_FORMAT_S16_BE; + ALSAFormats[3] = MA_SND_PCM_FORMAT_S24_3BE; + ALSAFormats[4] = MA_SND_PCM_FORMAT_S32_BE; + ALSAFormats[5] = MA_SND_PCM_FORMAT_FLOAT_BE; } - return MA_SUCCESS; -} - -ma_result ma_device_break_main_loop__alsa(ma_device* pDevice) -{ - ma_assert(pDevice != NULL); - - pDevice->alsa.breakFromMainLoop = MA_TRUE; - return MA_SUCCESS; + return ALSAFormats[format]; } -ma_result ma_device_main_loop__alsa(ma_device* pDevice) +static ma_format ma_format_from_alsa(ma_snd_pcm_format_t formatALSA) { - ma_assert(pDevice != NULL); - - pDevice->alsa.breakFromMainLoop = MA_FALSE; - if (pDevice->type == ma_device_type_playback) { - /* Playback. Read from client, write to device. */ - while (!pDevice->alsa.breakFromMainLoop && ma_device_read_from_client_and_write__alsa(pDevice)) { + if (ma_is_little_endian()) { + switch (formatALSA) { + case MA_SND_PCM_FORMAT_S16_LE: return ma_format_s16; + case MA_SND_PCM_FORMAT_S24_3LE: return ma_format_s24; + case MA_SND_PCM_FORMAT_S32_LE: return ma_format_s32; + case MA_SND_PCM_FORMAT_FLOAT_LE: return ma_format_f32; + default: break; } } else { - /* Capture. Read from device, write to client. */ - while (!pDevice->alsa.breakFromMainLoop && ma_device_read_and_send_to_client__alsa(pDevice)) { + switch (formatALSA) { + case MA_SND_PCM_FORMAT_S16_BE: return ma_format_s16; + case MA_SND_PCM_FORMAT_S24_3BE: return ma_format_s24; + case MA_SND_PCM_FORMAT_S32_BE: return ma_format_s32; + case MA_SND_PCM_FORMAT_FLOAT_BE: return ma_format_f32; + default: break; } } - return MA_SUCCESS; + /* Endian agnostic. */ + switch (formatALSA) { + case MA_SND_PCM_FORMAT_U8: return ma_format_u8; + default: return ma_format_unknown; + } } -#endif /* 0 */ -ma_result ma_device_read__alsa(ma_device* pDevice, void* pFramesOut, ma_uint32 frameCount, ma_uint32* pFramesRead) +static ma_channel ma_convert_alsa_channel_position_to_ma_channel(unsigned int alsaChannelPos) { - ma_snd_pcm_sframes_t resultALSA; - - ma_assert(pDevice != NULL); - ma_assert(pFramesOut != NULL); - - if (pFramesRead != NULL) { - *pFramesRead = 0; - } - - for (;;) { - resultALSA = ((ma_snd_pcm_readi_proc)pDevice->pContext->alsa.snd_pcm_readi)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture, pFramesOut, frameCount); - if (resultALSA >= 0) { - break; /* Success. */ - } else { - if (resultALSA == -EAGAIN) { - /*printf("TRACE: EGAIN (read)\n");*/ - continue; /* Try again. */ - } else if (resultALSA == -EPIPE) { - /*printf("TRACE: EPIPE (read)\n");*/ - - /* Overrun. Recover and try again. If this fails we need to return an error. */ - if (((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture, resultALSA, MA_TRUE) < 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to recover device after overrun.", MA_FAILED_TO_START_BACKEND_DEVICE); - } - - if (((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture) < 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start device after underrun.", MA_FAILED_TO_START_BACKEND_DEVICE); - } - - resultALSA = ((ma_snd_pcm_readi_proc)pDevice->pContext->alsa.snd_pcm_readi)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture, pFramesOut, frameCount); - if (resultALSA < 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to read data from the internal device.", MA_FAILED_TO_READ_DATA_FROM_DEVICE); - } - } - } - } - - if (pFramesRead != NULL) { - *pFramesRead = resultALSA; + switch (alsaChannelPos) + { + case MA_SND_CHMAP_MONO: return MA_CHANNEL_MONO; + case MA_SND_CHMAP_FL: return MA_CHANNEL_FRONT_LEFT; + case MA_SND_CHMAP_FR: return MA_CHANNEL_FRONT_RIGHT; + case MA_SND_CHMAP_RL: return MA_CHANNEL_BACK_LEFT; + case MA_SND_CHMAP_RR: return MA_CHANNEL_BACK_RIGHT; + case MA_SND_CHMAP_FC: return MA_CHANNEL_FRONT_CENTER; + case MA_SND_CHMAP_LFE: return MA_CHANNEL_LFE; + case MA_SND_CHMAP_SL: return MA_CHANNEL_SIDE_LEFT; + case MA_SND_CHMAP_SR: return MA_CHANNEL_SIDE_RIGHT; + case MA_SND_CHMAP_RC: return MA_CHANNEL_BACK_CENTER; + case MA_SND_CHMAP_FLC: return MA_CHANNEL_FRONT_LEFT_CENTER; + case MA_SND_CHMAP_FRC: return MA_CHANNEL_FRONT_RIGHT_CENTER; + case MA_SND_CHMAP_RLC: return 0; + case MA_SND_CHMAP_RRC: return 0; + case MA_SND_CHMAP_FLW: return 0; + case MA_SND_CHMAP_FRW: return 0; + case MA_SND_CHMAP_FLH: return 0; + case MA_SND_CHMAP_FCH: return 0; + case MA_SND_CHMAP_FRH: return 0; + case MA_SND_CHMAP_TC: return MA_CHANNEL_TOP_CENTER; + case MA_SND_CHMAP_TFL: return MA_CHANNEL_TOP_FRONT_LEFT; + case MA_SND_CHMAP_TFR: return MA_CHANNEL_TOP_FRONT_RIGHT; + case MA_SND_CHMAP_TFC: return MA_CHANNEL_TOP_FRONT_CENTER; + case MA_SND_CHMAP_TRL: return MA_CHANNEL_TOP_BACK_LEFT; + case MA_SND_CHMAP_TRR: return MA_CHANNEL_TOP_BACK_RIGHT; + case MA_SND_CHMAP_TRC: return MA_CHANNEL_TOP_BACK_CENTER; + default: break; } - return MA_SUCCESS; + return 0; } -ma_result ma_device_write__alsa(ma_device* pDevice, const void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) +static ma_bool32 ma_is_common_device_name__alsa(const char* name) { - ma_snd_pcm_sframes_t resultALSA; - - ma_assert(pDevice != NULL); - ma_assert(pFrames != NULL); - - if (pFramesWritten != NULL) { - *pFramesWritten = 0; + size_t iName; + for (iName = 0; iName < ma_countof(g_maCommonDeviceNamesALSA); ++iName) { + if (ma_strcmp(name, g_maCommonDeviceNamesALSA[iName]) == 0) { + return MA_TRUE; + } } - for (;;) { - resultALSA = ((ma_snd_pcm_writei_proc)pDevice->pContext->alsa.snd_pcm_writei)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback, pFrames, frameCount); - if (resultALSA >= 0) { - break; /* Success. */ - } else { - if (resultALSA == -EAGAIN) { - /*printf("TRACE: EGAIN (write)\n");*/ - continue; /* Try again. */ - } else if (resultALSA == -EPIPE) { - /*printf("TRACE: EPIPE (write)\n");*/ - - /* Underrun. Recover and try again. If this fails we need to return an error. */ - if (((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback, resultALSA, MA_TRUE) < 0) { /* MA_TRUE=silent (don't print anything on error). */ - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to recover device after underrun.", MA_FAILED_TO_START_BACKEND_DEVICE); - } + return MA_FALSE; +} - /* - In my testing I have had a situation where writei() does not automatically restart the device even though I've set it - up as such in the software parameters. What will happen is writei() will block indefinitely even though the number of - frames is well beyond the auto-start threshold. To work around this I've needed to add an explicit start here. Not sure - if this is me just being stupid and not recovering the device properly, but this definitely feels like something isn't - quite right here. - */ - if (((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback) < 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start device after underrun.", MA_FAILED_TO_START_BACKEND_DEVICE); - } - resultALSA = ((ma_snd_pcm_writei_proc)pDevice->pContext->alsa.snd_pcm_writei)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback, pFrames, frameCount); - if (resultALSA < 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to write data to device after underrun.", MA_FAILED_TO_START_BACKEND_DEVICE); - } - } +static ma_bool32 ma_is_playback_device_blacklisted__alsa(const char* name) +{ + size_t iName; + for (iName = 0; iName < ma_countof(g_maBlacklistedPlaybackDeviceNamesALSA); ++iName) { + if (ma_strcmp(name, g_maBlacklistedPlaybackDeviceNamesALSA[iName]) == 0) { + return MA_TRUE; } } - if (pFramesWritten != NULL) { - *pFramesWritten = resultALSA; + return MA_FALSE; +} + +static ma_bool32 ma_is_capture_device_blacklisted__alsa(const char* name) +{ + size_t iName; + for (iName = 0; iName < ma_countof(g_maBlacklistedCaptureDeviceNamesALSA); ++iName) { + if (ma_strcmp(name, g_maBlacklistedCaptureDeviceNamesALSA[iName]) == 0) { + return MA_TRUE; + } } - return MA_SUCCESS; + return MA_FALSE; } -ma_result ma_device_main_loop__alsa(ma_device* pDevice) +static ma_bool32 ma_is_device_blacklisted__alsa(ma_device_type deviceType, const char* name) { - ma_result result = MA_SUCCESS; - ma_bool32 exitLoop = MA_FALSE; + if (deviceType == ma_device_type_playback) { + return ma_is_playback_device_blacklisted__alsa(name); + } else { + return ma_is_capture_device_blacklisted__alsa(name); + } +} - ma_assert(pDevice != NULL); - /* Capture devices need to be started immediately. */ - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - if (((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture) < 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start device in preparation for reading.", MA_FAILED_TO_START_BACKEND_DEVICE); +static const char* ma_find_char(const char* str, char c, int* index) +{ + int i = 0; + for (;;) { + if (str[i] == '\0') { + if (index) *index = -1; + return NULL; } - } - - while (ma_device__get_state(pDevice) == MA_STATE_STARTED && !exitLoop) { - switch (pDevice->type) - { - case ma_device_type_duplex: - { - if (pDevice->alsa.isUsingMMapCapture || pDevice->alsa.isUsingMMapPlayback) { - /* MMAP */ - return MA_INVALID_OPERATION; /* Not yet implemented. */ - } else { - /* readi() and writei() */ - /* The process is: device_read -> convert -> callback -> convert -> device_write */ - ma_uint8 capturedDeviceData[8192]; - ma_uint8 playbackDeviceData[8192]; - ma_uint32 capturedDeviceDataCapInFrames = sizeof(capturedDeviceData) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 playbackDeviceDataCapInFrames = sizeof(playbackDeviceData) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + if (str[i] == c) { + if (index) *index = i; + return str + i; + } - ma_uint32 totalFramesProcessed = 0; - ma_uint32 periodSizeInFrames = ma_min(pDevice->capture.internalBufferSizeInFrames/pDevice->capture.internalPeriods, pDevice->playback.internalBufferSizeInFrames/pDevice->playback.internalPeriods); - - while (totalFramesProcessed < periodSizeInFrames) { - ma_uint32 framesRemaining = periodSizeInFrames - totalFramesProcessed; - ma_uint32 framesProcessed; - ma_uint32 framesToProcess = framesRemaining; - if (framesToProcess > capturedDeviceDataCapInFrames) { - framesToProcess = capturedDeviceDataCapInFrames; - } + i += 1; + } - result = ma_device_read__alsa(pDevice, capturedDeviceData, framesToProcess, &framesProcessed); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } + /* Should never get here, but treat it as though the character was not found to make me feel better inside. */ + if (index) *index = -1; + return NULL; +} - pDevice->capture._dspFrameCount = framesToProcess; - pDevice->capture._dspFrames = capturedDeviceData; +static ma_bool32 ma_is_device_name_in_hw_format__alsa(const char* hwid) +{ + /* This function is just checking whether or not hwid is in "hw:%d,%d" format. */ - for (;;) { - ma_uint8 capturedData[8192]; - ma_uint8 playbackData[8192]; - ma_uint32 capturedDataCapInFrames = sizeof(capturedData) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); - ma_uint32 playbackDataCapInFrames = sizeof(playbackData) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); - - ma_uint32 capturedFramesToTryProcessing = ma_min(capturedDataCapInFrames, playbackDataCapInFrames); - ma_uint32 capturedFramesToProcess = (ma_uint32)ma_pcm_converter_read(&pDevice->capture.converter, capturedData, capturedFramesToTryProcessing); - if (capturedFramesToProcess == 0) { - break; /* Don't fire the data callback with zero frames. */ - } + int commaPos; + const char* dev; + int i; - ma_device__on_data(pDevice, playbackData, capturedData, capturedFramesToProcess); + if (hwid == NULL) { + return MA_FALSE; + } - /* At this point the playbackData buffer should be holding data that needs to be written to the device. */ - pDevice->playback._dspFrameCount = capturedFramesToProcess; - pDevice->playback._dspFrames = playbackData; - for (;;) { - ma_uint32 playbackDeviceFramesCount = (ma_uint32)ma_pcm_converter_read(&pDevice->playback.converter, playbackDeviceData, playbackDeviceDataCapInFrames); - if (playbackDeviceFramesCount == 0) { - break; - } + if (hwid[0] != 'h' || hwid[1] != 'w' || hwid[2] != ':') { + return MA_FALSE; + } - result = ma_device_write__alsa(pDevice, playbackDeviceData, playbackDeviceFramesCount, NULL); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } + hwid += 3; - if (playbackDeviceFramesCount < playbackDeviceDataCapInFrames) { - break; - } - } + dev = ma_find_char(hwid, ',', &commaPos); + if (dev == NULL) { + return MA_FALSE; + } else { + dev += 1; /* Skip past the ",". */ + } - if (capturedFramesToProcess < capturedFramesToTryProcessing) { - break; - } + /* Check if the part between the ":" and the "," contains only numbers. If not, return false. */ + for (i = 0; i < commaPos; ++i) { + if (hwid[i] < '0' || hwid[i] > '9') { + return MA_FALSE; + } + } - /* In case an error happened from ma_device_write2__alsa()... */ - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } - } + /* Check if everything after the "," is numeric. If not, return false. */ + i = 0; + while (dev[i] != '\0') { + if (dev[i] < '0' || dev[i] > '9') { + return MA_FALSE; + } + i += 1; + } - totalFramesProcessed += framesProcessed; - } - } - } break; + return MA_TRUE; +} - case ma_device_type_capture: - { - if (pDevice->alsa.isUsingMMapCapture) { - /* MMAP */ - return MA_INVALID_OPERATION; /* Not yet implemented. */ - } else { - /* readi() */ +static int ma_convert_device_name_to_hw_format__alsa(ma_context* pContext, char* dst, size_t dstSize, const char* src) /* Returns 0 on success, non-0 on error. */ +{ + /* src should look something like this: "hw:CARD=I82801AAICH,DEV=0" */ - /* We read in chunks of the period size, but use a stack allocated buffer for the intermediary. */ - ma_uint8 intermediaryBuffer[8192]; - ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 periodSizeInFrames = pDevice->capture.internalBufferSizeInFrames / pDevice->capture.internalPeriods; - ma_uint32 framesReadThisPeriod = 0; - while (framesReadThisPeriod < periodSizeInFrames) { - ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesReadThisPeriod; - ma_uint32 framesProcessed; - ma_uint32 framesToReadThisIteration = framesRemainingInPeriod; - if (framesToReadThisIteration > intermediaryBufferSizeInFrames) { - framesToReadThisIteration = intermediaryBufferSizeInFrames; - } + int colonPos; + int commaPos; + char card[256]; + const char* dev; + int cardIndex; - result = ma_device_read__alsa(pDevice, intermediaryBuffer, framesToReadThisIteration, &framesProcessed); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } + if (dst == NULL) { + return -1; + } + if (dstSize < 7) { + return -1; /* Absolute minimum size of the output buffer is 7 bytes. */ + } - ma_device__send_frames_to_client(pDevice, framesProcessed, intermediaryBuffer); + *dst = '\0'; /* Safety. */ + if (src == NULL) { + return -1; + } - framesReadThisPeriod += framesProcessed; - } - } - } break; + /* If the input name is already in "hw:%d,%d" format, just return that verbatim. */ + if (ma_is_device_name_in_hw_format__alsa(src)) { + return ma_strcpy_s(dst, dstSize, src); + } - case ma_device_type_playback: - { - if (pDevice->alsa.isUsingMMapPlayback) { - /* MMAP */ - return MA_INVALID_OPERATION; /* Not yet implemented. */ - } else { - /* writei() */ + src = ma_find_char(src, ':', &colonPos); + if (src == NULL) { + return -1; /* Couldn't find a colon */ + } - /* We write in chunks of the period size, but use a stack allocated buffer for the intermediary. */ - ma_uint8 intermediaryBuffer[8192]; - ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - ma_uint32 periodSizeInFrames = pDevice->playback.internalBufferSizeInFrames / pDevice->playback.internalPeriods; - ma_uint32 framesWrittenThisPeriod = 0; - while (framesWrittenThisPeriod < periodSizeInFrames) { - ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesWrittenThisPeriod; - ma_uint32 framesProcessed; - ma_uint32 framesToWriteThisIteration = framesRemainingInPeriod; - if (framesToWriteThisIteration > intermediaryBufferSizeInFrames) { - framesToWriteThisIteration = intermediaryBufferSizeInFrames; - } + dev = ma_find_char(src, ',', &commaPos); + if (dev == NULL) { + dev = "0"; + ma_strncpy_s(card, sizeof(card), src+6, (size_t)-1); /* +6 = ":CARD=" */ + } else { + dev = dev + 5; /* +5 = ",DEV=" */ + ma_strncpy_s(card, sizeof(card), src+6, commaPos-6); /* +6 = ":CARD=" */ + } - ma_device__read_frames_from_client(pDevice, framesToWriteThisIteration, intermediaryBuffer); + cardIndex = ((ma_snd_card_get_index_proc)pContext->alsa.snd_card_get_index)(card); + if (cardIndex < 0) { + return -2; /* Failed to retrieve the card index. */ + } - result = ma_device_write__alsa(pDevice, intermediaryBuffer, framesToWriteThisIteration, &framesProcessed); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } + /*printf("TESTING: CARD=%s,DEV=%s\n", card, dev); */ - framesWrittenThisPeriod += framesProcessed; - } - } - } break; - /* To silence a warning. Will never hit this. */ - case ma_device_type_loopback: - default: break; - } + /* Construction. */ + dst[0] = 'h'; dst[1] = 'w'; dst[2] = ':'; + if (ma_itoa_s(cardIndex, dst+3, dstSize-3, 10) != 0) { + return -3; + } + if (ma_strcat_s(dst, dstSize, ",") != 0) { + return -3; + } + if (ma_strcat_s(dst, dstSize, dev) != 0) { + return -3; } - /* Here is where the device needs to be stopped. */ - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - ((ma_snd_pcm_drain_proc)pDevice->pContext->alsa.snd_pcm_drain)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture); + return 0; +} - /* We need to prepare the device again, otherwise we won't be able to restart the device. */ - if (((ma_snd_pcm_prepare_proc)pDevice->pContext->alsa.snd_pcm_prepare)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture) < 0) { - #ifdef MA_DEBUG_OUTPUT - printf("[ALSA] Failed to prepare capture device after stopping.\n"); - #endif - } - } +static ma_bool32 ma_does_id_exist_in_list__alsa(ma_device_id* pUniqueIDs, ma_uint32 count, const char* pHWID) +{ + ma_uint32 i; - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - ((ma_snd_pcm_drain_proc)pDevice->pContext->alsa.snd_pcm_drain)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback); + MA_ASSERT(pHWID != NULL); - /* We need to prepare the device again, otherwise we won't be able to restart the device. */ - if (((ma_snd_pcm_prepare_proc)pDevice->pContext->alsa.snd_pcm_prepare)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback) < 0) { - #ifdef MA_DEBUG_OUTPUT - printf("[ALSA] Failed to prepare playback device after stopping.\n"); - #endif + for (i = 0; i < count; ++i) { + if (ma_strcmp(pUniqueIDs[i].alsa, pHWID) == 0) { + return MA_TRUE; } } - return result; + return MA_FALSE; } -ma_result ma_context_uninit__alsa(ma_context* pContext) + +static ma_result ma_context_open_pcm__alsa(ma_context* pContext, ma_share_mode shareMode, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_snd_pcm_t** ppPCM) { - ma_assert(pContext != NULL); - ma_assert(pContext->backend == ma_backend_alsa); + ma_snd_pcm_t* pPCM; + ma_snd_pcm_stream_t stream; + int openMode; - /* Clean up memory for memory leak checkers. */ - ((ma_snd_config_update_free_global_proc)pContext->alsa.snd_config_update_free_global)(); + MA_ASSERT(pContext != NULL); + MA_ASSERT(ppPCM != NULL); -#ifndef MA_NO_RUNTIME_LINKING - ma_dlclose(pContext, pContext->alsa.asoundSO); -#endif + *ppPCM = NULL; + pPCM = NULL; - ma_mutex_uninit(&pContext->alsa.internalDeviceEnumLock); + stream = (deviceType == ma_device_type_playback) ? MA_SND_PCM_STREAM_PLAYBACK : MA_SND_PCM_STREAM_CAPTURE; + openMode = MA_SND_PCM_NO_AUTO_RESAMPLE | MA_SND_PCM_NO_AUTO_CHANNELS | MA_SND_PCM_NO_AUTO_FORMAT; - return MA_SUCCESS; -} + if (pDeviceID == NULL) { + ma_bool32 isDeviceOpen; + size_t i; -ma_result ma_context_init__alsa(const ma_context_config* pConfig, ma_context* pContext) -{ -#ifndef MA_NO_RUNTIME_LINKING - const char* libasoundNames[] = { - "libasound.so.2", - "libasound.so" - }; - size_t i; + /* + We're opening the default device. I don't know if trying anything other than "default" is necessary, but it makes + me feel better to try as hard as we can get to get _something_ working. + */ + const char* defaultDeviceNames[] = { + "default", + NULL, + NULL, + NULL, + NULL, + NULL, + NULL + }; - for (i = 0; i < ma_countof(libasoundNames); ++i) { - pContext->alsa.asoundSO = ma_dlopen(pContext, libasoundNames[i]); - if (pContext->alsa.asoundSO != NULL) { - break; + if (shareMode == ma_share_mode_exclusive) { + defaultDeviceNames[1] = "hw"; + defaultDeviceNames[2] = "hw:0"; + defaultDeviceNames[3] = "hw:0,0"; + } else { + if (deviceType == ma_device_type_playback) { + defaultDeviceNames[1] = "dmix"; + defaultDeviceNames[2] = "dmix:0"; + defaultDeviceNames[3] = "dmix:0,0"; + } else { + defaultDeviceNames[1] = "dsnoop"; + defaultDeviceNames[2] = "dsnoop:0"; + defaultDeviceNames[3] = "dsnoop:0,0"; + } + defaultDeviceNames[4] = "hw"; + defaultDeviceNames[5] = "hw:0"; + defaultDeviceNames[6] = "hw:0,0"; } - } - if (pContext->alsa.asoundSO == NULL) { -#ifdef MA_DEBUG_OUTPUT - printf("[ALSA] Failed to open shared object.\n"); -#endif - return MA_NO_BACKEND; - } + isDeviceOpen = MA_FALSE; + for (i = 0; i < ma_countof(defaultDeviceNames); ++i) { + if (defaultDeviceNames[i] != NULL && defaultDeviceNames[i][0] != '\0') { + if (((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, defaultDeviceNames[i], stream, openMode) == 0) { + isDeviceOpen = MA_TRUE; + break; + } + } + } - pContext->alsa.snd_pcm_open = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_open"); - pContext->alsa.snd_pcm_close = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_close"); - pContext->alsa.snd_pcm_hw_params_sizeof = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_sizeof"); - pContext->alsa.snd_pcm_hw_params_any = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_any"); - pContext->alsa.snd_pcm_hw_params_set_format = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_format"); - pContext->alsa.snd_pcm_hw_params_set_format_first = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_format_first"); - pContext->alsa.snd_pcm_hw_params_get_format_mask = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_format_mask"); - pContext->alsa.snd_pcm_hw_params_set_channels_near = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_channels_near"); - pContext->alsa.snd_pcm_hw_params_set_rate_resample = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_rate_resample"); - pContext->alsa.snd_pcm_hw_params_set_rate_near = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_rate_near"); - pContext->alsa.snd_pcm_hw_params_set_buffer_size_near = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_buffer_size_near"); - pContext->alsa.snd_pcm_hw_params_set_periods_near = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_periods_near"); - pContext->alsa.snd_pcm_hw_params_set_access = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_access"); - pContext->alsa.snd_pcm_hw_params_get_format = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_format"); - pContext->alsa.snd_pcm_hw_params_get_channels = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_channels"); - pContext->alsa.snd_pcm_hw_params_get_channels_min = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_channels_min"); - pContext->alsa.snd_pcm_hw_params_get_channels_max = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_channels_max"); - pContext->alsa.snd_pcm_hw_params_get_rate = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_rate"); - pContext->alsa.snd_pcm_hw_params_get_rate_min = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_rate_min"); - pContext->alsa.snd_pcm_hw_params_get_rate_max = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_rate_max"); - pContext->alsa.snd_pcm_hw_params_get_buffer_size = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_buffer_size"); - pContext->alsa.snd_pcm_hw_params_get_periods = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_periods"); - pContext->alsa.snd_pcm_hw_params_get_access = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_access"); - pContext->alsa.snd_pcm_hw_params = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params"); - pContext->alsa.snd_pcm_sw_params_sizeof = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_sizeof"); - pContext->alsa.snd_pcm_sw_params_current = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_current"); - pContext->alsa.snd_pcm_sw_params_get_boundary = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_get_boundary"); - pContext->alsa.snd_pcm_sw_params_set_avail_min = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_set_avail_min"); - pContext->alsa.snd_pcm_sw_params_set_start_threshold = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_set_start_threshold"); - pContext->alsa.snd_pcm_sw_params_set_stop_threshold = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_set_stop_threshold"); - pContext->alsa.snd_pcm_sw_params = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params"); - pContext->alsa.snd_pcm_format_mask_sizeof = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_format_mask_sizeof"); - pContext->alsa.snd_pcm_format_mask_test = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_format_mask_test"); - pContext->alsa.snd_pcm_get_chmap = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_get_chmap"); - pContext->alsa.snd_pcm_state = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_state"); - pContext->alsa.snd_pcm_prepare = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_prepare"); - pContext->alsa.snd_pcm_start = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_start"); - pContext->alsa.snd_pcm_drop = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_drop"); - pContext->alsa.snd_pcm_drain = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_drain"); - pContext->alsa.snd_device_name_hint = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_device_name_hint"); - pContext->alsa.snd_device_name_get_hint = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_device_name_get_hint"); - pContext->alsa.snd_card_get_index = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_card_get_index"); - pContext->alsa.snd_device_name_free_hint = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_device_name_free_hint"); - pContext->alsa.snd_pcm_mmap_begin = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_mmap_begin"); - pContext->alsa.snd_pcm_mmap_commit = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_mmap_commit"); - pContext->alsa.snd_pcm_recover = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_recover"); - pContext->alsa.snd_pcm_readi = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_readi"); - pContext->alsa.snd_pcm_writei = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_writei"); - pContext->alsa.snd_pcm_avail = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_avail"); - pContext->alsa.snd_pcm_avail_update = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_avail_update"); - pContext->alsa.snd_pcm_wait = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_wait"); - pContext->alsa.snd_pcm_info = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_info"); - pContext->alsa.snd_pcm_info_sizeof = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_info_sizeof"); - pContext->alsa.snd_pcm_info_get_name = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_info_get_name"); - pContext->alsa.snd_config_update_free_global = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_config_update_free_global"); -#else - /* The system below is just for type safety. */ - ma_snd_pcm_open_proc _snd_pcm_open = snd_pcm_open; - ma_snd_pcm_close_proc _snd_pcm_close = snd_pcm_close; - ma_snd_pcm_hw_params_sizeof_proc _snd_pcm_hw_params_sizeof = snd_pcm_hw_params_sizeof; - ma_snd_pcm_hw_params_any_proc _snd_pcm_hw_params_any = snd_pcm_hw_params_any; - ma_snd_pcm_hw_params_set_format_proc _snd_pcm_hw_params_set_format = snd_pcm_hw_params_set_format; - ma_snd_pcm_hw_params_set_format_first_proc _snd_pcm_hw_params_set_format_first = snd_pcm_hw_params_set_format_first; - ma_snd_pcm_hw_params_get_format_mask_proc _snd_pcm_hw_params_get_format_mask = snd_pcm_hw_params_get_format_mask; - ma_snd_pcm_hw_params_set_channels_near_proc _snd_pcm_hw_params_set_channels_near = snd_pcm_hw_params_set_channels_near; - ma_snd_pcm_hw_params_set_rate_resample_proc _snd_pcm_hw_params_set_rate_resample = snd_pcm_hw_params_set_rate_resample; - ma_snd_pcm_hw_params_set_rate_near_proc _snd_pcm_hw_params_set_rate_near = snd_pcm_hw_params_set_rate_near; - ma_snd_pcm_hw_params_set_buffer_size_near_proc _snd_pcm_hw_params_set_buffer_size_near = snd_pcm_hw_params_set_buffer_size_near; - ma_snd_pcm_hw_params_set_periods_near_proc _snd_pcm_hw_params_set_periods_near = snd_pcm_hw_params_set_periods_near; - ma_snd_pcm_hw_params_set_access_proc _snd_pcm_hw_params_set_access = snd_pcm_hw_params_set_access; - ma_snd_pcm_hw_params_get_format_proc _snd_pcm_hw_params_get_format = snd_pcm_hw_params_get_format; - ma_snd_pcm_hw_params_get_channels_proc _snd_pcm_hw_params_get_channels = snd_pcm_hw_params_get_channels; - ma_snd_pcm_hw_params_get_channels_min_proc _snd_pcm_hw_params_get_channels_min = snd_pcm_hw_params_get_channels_min; - ma_snd_pcm_hw_params_get_channels_max_proc _snd_pcm_hw_params_get_channels_max = snd_pcm_hw_params_get_channels_max; - ma_snd_pcm_hw_params_get_rate_proc _snd_pcm_hw_params_get_rate = snd_pcm_hw_params_get_rate; - ma_snd_pcm_hw_params_get_rate_min_proc _snd_pcm_hw_params_get_rate_min = snd_pcm_hw_params_get_rate_min; - ma_snd_pcm_hw_params_get_rate_max_proc _snd_pcm_hw_params_get_rate_max = snd_pcm_hw_params_get_rate_max; - ma_snd_pcm_hw_params_get_buffer_size_proc _snd_pcm_hw_params_get_buffer_size = snd_pcm_hw_params_get_buffer_size; - ma_snd_pcm_hw_params_get_periods_proc _snd_pcm_hw_params_get_periods = snd_pcm_hw_params_get_periods; - ma_snd_pcm_hw_params_get_access_proc _snd_pcm_hw_params_get_access = snd_pcm_hw_params_get_access; - ma_snd_pcm_hw_params_proc _snd_pcm_hw_params = snd_pcm_hw_params; - ma_snd_pcm_sw_params_sizeof_proc _snd_pcm_sw_params_sizeof = snd_pcm_sw_params_sizeof; - ma_snd_pcm_sw_params_current_proc _snd_pcm_sw_params_current = snd_pcm_sw_params_current; - ma_snd_pcm_sw_params_get_boundary_proc _snd_pcm_sw_params_get_boundary = snd_pcm_sw_params_get_boundary; - ma_snd_pcm_sw_params_set_avail_min_proc _snd_pcm_sw_params_set_avail_min = snd_pcm_sw_params_set_avail_min; - ma_snd_pcm_sw_params_set_start_threshold_proc _snd_pcm_sw_params_set_start_threshold = snd_pcm_sw_params_set_start_threshold; - ma_snd_pcm_sw_params_set_stop_threshold_proc _snd_pcm_sw_params_set_stop_threshold = snd_pcm_sw_params_set_stop_threshold; - ma_snd_pcm_sw_params_proc _snd_pcm_sw_params = snd_pcm_sw_params; - ma_snd_pcm_format_mask_sizeof_proc _snd_pcm_format_mask_sizeof = snd_pcm_format_mask_sizeof; - ma_snd_pcm_format_mask_test_proc _snd_pcm_format_mask_test = snd_pcm_format_mask_test; - ma_snd_pcm_get_chmap_proc _snd_pcm_get_chmap = snd_pcm_get_chmap; - ma_snd_pcm_state_proc _snd_pcm_state = snd_pcm_state; - ma_snd_pcm_prepare_proc _snd_pcm_prepare = snd_pcm_prepare; - ma_snd_pcm_start_proc _snd_pcm_start = snd_pcm_start; - ma_snd_pcm_drop_proc _snd_pcm_drop = snd_pcm_drop; - ma_snd_pcm_drain_proc _snd_pcm_drain = snd_pcm_drain; - ma_snd_device_name_hint_proc _snd_device_name_hint = snd_device_name_hint; - ma_snd_device_name_get_hint_proc _snd_device_name_get_hint = snd_device_name_get_hint; - ma_snd_card_get_index_proc _snd_card_get_index = snd_card_get_index; - ma_snd_device_name_free_hint_proc _snd_device_name_free_hint = snd_device_name_free_hint; - ma_snd_pcm_mmap_begin_proc _snd_pcm_mmap_begin = snd_pcm_mmap_begin; - ma_snd_pcm_mmap_commit_proc _snd_pcm_mmap_commit = snd_pcm_mmap_commit; - ma_snd_pcm_recover_proc _snd_pcm_recover = snd_pcm_recover; - ma_snd_pcm_readi_proc _snd_pcm_readi = snd_pcm_readi; - ma_snd_pcm_writei_proc _snd_pcm_writei = snd_pcm_writei; - ma_snd_pcm_avail_proc _snd_pcm_avail = snd_pcm_avail; - ma_snd_pcm_avail_update_proc _snd_pcm_avail_update = snd_pcm_avail_update; - ma_snd_pcm_wait_proc _snd_pcm_wait = snd_pcm_wait; - ma_snd_pcm_info_proc _snd_pcm_info = snd_pcm_info; - ma_snd_pcm_info_sizeof_proc _snd_pcm_info_sizeof = snd_pcm_info_sizeof; - ma_snd_pcm_info_get_name_proc _snd_pcm_info_get_name = snd_pcm_info_get_name; - ma_snd_config_update_free_global_proc _snd_config_update_free_global = snd_config_update_free_global; + if (!isDeviceOpen) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_open() failed when trying to open an appropriate default device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } + } else { + /* + We're trying to open a specific device. There's a few things to consider here: + + miniaudio recongnizes a special format of device id that excludes the "hw", "dmix", etc. prefix. It looks like this: ":0,0", ":0,1", etc. When + an ID of this format is specified, it indicates to miniaudio that it can try different combinations of plugins ("hw", "dmix", etc.) until it + finds an appropriate one that works. This comes in very handy when trying to open a device in shared mode ("dmix"), vs exclusive mode ("hw"). + */ + + /* May end up needing to make small adjustments to the ID, so make a copy. */ + ma_device_id deviceID = *pDeviceID; + ma_bool32 isDeviceOpen = MA_FALSE; - pContext->alsa.snd_pcm_open = (ma_proc)_snd_pcm_open; - pContext->alsa.snd_pcm_close = (ma_proc)_snd_pcm_close; - pContext->alsa.snd_pcm_hw_params_sizeof = (ma_proc)_snd_pcm_hw_params_sizeof; - pContext->alsa.snd_pcm_hw_params_any = (ma_proc)_snd_pcm_hw_params_any; - pContext->alsa.snd_pcm_hw_params_set_format = (ma_proc)_snd_pcm_hw_params_set_format; - pContext->alsa.snd_pcm_hw_params_set_format_first = (ma_proc)_snd_pcm_hw_params_set_format_first; - pContext->alsa.snd_pcm_hw_params_get_format_mask = (ma_proc)_snd_pcm_hw_params_get_format_mask; - pContext->alsa.snd_pcm_hw_params_set_channels_near = (ma_proc)_snd_pcm_hw_params_set_channels_near; - pContext->alsa.snd_pcm_hw_params_set_rate_resample = (ma_proc)_snd_pcm_hw_params_set_rate_resample; - pContext->alsa.snd_pcm_hw_params_set_rate_near = (ma_proc)_snd_pcm_hw_params_set_rate_near; - pContext->alsa.snd_pcm_hw_params_set_buffer_size_near = (ma_proc)_snd_pcm_hw_params_set_buffer_size_near; - pContext->alsa.snd_pcm_hw_params_set_periods_near = (ma_proc)_snd_pcm_hw_params_set_periods_near; - pContext->alsa.snd_pcm_hw_params_set_access = (ma_proc)_snd_pcm_hw_params_set_access; - pContext->alsa.snd_pcm_hw_params_get_format = (ma_proc)_snd_pcm_hw_params_get_format; - pContext->alsa.snd_pcm_hw_params_get_channels = (ma_proc)_snd_pcm_hw_params_get_channels; - pContext->alsa.snd_pcm_hw_params_get_channels_min = (ma_proc)_snd_pcm_hw_params_get_channels_min; - pContext->alsa.snd_pcm_hw_params_get_channels_max = (ma_proc)_snd_pcm_hw_params_get_channels_max; - pContext->alsa.snd_pcm_hw_params_get_rate = (ma_proc)_snd_pcm_hw_params_get_rate; - pContext->alsa.snd_pcm_hw_params_get_buffer_size = (ma_proc)_snd_pcm_hw_params_get_buffer_size; - pContext->alsa.snd_pcm_hw_params_get_periods = (ma_proc)_snd_pcm_hw_params_get_periods; - pContext->alsa.snd_pcm_hw_params_get_access = (ma_proc)_snd_pcm_hw_params_get_access; - pContext->alsa.snd_pcm_hw_params = (ma_proc)_snd_pcm_hw_params; - pContext->alsa.snd_pcm_sw_params_sizeof = (ma_proc)_snd_pcm_sw_params_sizeof; - pContext->alsa.snd_pcm_sw_params_current = (ma_proc)_snd_pcm_sw_params_current; - pContext->alsa.snd_pcm_sw_params_get_boundary = (ma_proc)_snd_pcm_sw_params_get_boundary; - pContext->alsa.snd_pcm_sw_params_set_avail_min = (ma_proc)_snd_pcm_sw_params_set_avail_min; - pContext->alsa.snd_pcm_sw_params_set_start_threshold = (ma_proc)_snd_pcm_sw_params_set_start_threshold; - pContext->alsa.snd_pcm_sw_params_set_stop_threshold = (ma_proc)_snd_pcm_sw_params_set_stop_threshold; - pContext->alsa.snd_pcm_sw_params = (ma_proc)_snd_pcm_sw_params; - pContext->alsa.snd_pcm_format_mask_sizeof = (ma_proc)_snd_pcm_format_mask_sizeof; - pContext->alsa.snd_pcm_format_mask_test = (ma_proc)_snd_pcm_format_mask_test; - pContext->alsa.snd_pcm_get_chmap = (ma_proc)_snd_pcm_get_chmap; - pContext->alsa.snd_pcm_state = (ma_proc)_snd_pcm_state; - pContext->alsa.snd_pcm_prepare = (ma_proc)_snd_pcm_prepare; - pContext->alsa.snd_pcm_start = (ma_proc)_snd_pcm_start; - pContext->alsa.snd_pcm_drop = (ma_proc)_snd_pcm_drop; - pContext->alsa.snd_pcm_drain = (ma_proc)_snd_pcm_drain; - pContext->alsa.snd_device_name_hint = (ma_proc)_snd_device_name_hint; - pContext->alsa.snd_device_name_get_hint = (ma_proc)_snd_device_name_get_hint; - pContext->alsa.snd_card_get_index = (ma_proc)_snd_card_get_index; - pContext->alsa.snd_device_name_free_hint = (ma_proc)_snd_device_name_free_hint; - pContext->alsa.snd_pcm_mmap_begin = (ma_proc)_snd_pcm_mmap_begin; - pContext->alsa.snd_pcm_mmap_commit = (ma_proc)_snd_pcm_mmap_commit; - pContext->alsa.snd_pcm_recover = (ma_proc)_snd_pcm_recover; - pContext->alsa.snd_pcm_readi = (ma_proc)_snd_pcm_readi; - pContext->alsa.snd_pcm_writei = (ma_proc)_snd_pcm_writei; - pContext->alsa.snd_pcm_avail = (ma_proc)_snd_pcm_avail; - pContext->alsa.snd_pcm_avail_update = (ma_proc)_snd_pcm_avail_update; - pContext->alsa.snd_pcm_wait = (ma_proc)_snd_pcm_wait; - pContext->alsa.snd_pcm_info = (ma_proc)_snd_pcm_info; - pContext->alsa.snd_pcm_info_sizeof = (ma_proc)_snd_pcm_info_sizeof; - pContext->alsa.snd_pcm_info_get_name = (ma_proc)_snd_pcm_info_get_name; - pContext->alsa.snd_config_update_free_global = (ma_proc)_snd_config_update_free_global; -#endif + if (deviceID.alsa[0] != ':') { + /* The ID is not in ":0,0" format. Use the ID exactly as-is. */ + if (((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, deviceID.alsa, stream, openMode) == 0) { + isDeviceOpen = MA_TRUE; + } + } else { + char hwid[256]; - pContext->alsa.useVerboseDeviceEnumeration = pConfig->alsa.useVerboseDeviceEnumeration; + /* The ID is in ":0,0" format. Try different plugins depending on the shared mode. */ + if (deviceID.alsa[1] == '\0') { + deviceID.alsa[0] = '\0'; /* An ID of ":" should be converted to "". */ + } - if (ma_mutex_init(pContext, &pContext->alsa.internalDeviceEnumLock) != MA_SUCCESS) { - ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[ALSA] WARNING: Failed to initialize mutex for internal device enumeration.", MA_ERROR); - } + if (shareMode == ma_share_mode_shared) { + if (deviceType == ma_device_type_playback) { + ma_strcpy_s(hwid, sizeof(hwid), "dmix"); + } else { + ma_strcpy_s(hwid, sizeof(hwid), "dsnoop"); + } - pContext->onUninit = ma_context_uninit__alsa; - pContext->onDeviceIDEqual = ma_context_is_device_id_equal__alsa; - pContext->onEnumDevices = ma_context_enumerate_devices__alsa; - pContext->onGetDeviceInfo = ma_context_get_device_info__alsa; - pContext->onDeviceInit = ma_device_init__alsa; - pContext->onDeviceUninit = ma_device_uninit__alsa; - pContext->onDeviceStart = NULL; /* Not used. Started in the main loop. */ - pContext->onDeviceStop = NULL; /* Not used. Started in the main loop. */ - pContext->onDeviceMainLoop = ma_device_main_loop__alsa; + if (ma_strcat_s(hwid, sizeof(hwid), deviceID.alsa) == 0) { + if (((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, hwid, stream, openMode) == 0) { + isDeviceOpen = MA_TRUE; + } + } + } + + /* If at this point we still don't have an open device it means we're either preferencing exclusive mode or opening with "dmix"/"dsnoop" failed. */ + if (!isDeviceOpen) { + ma_strcpy_s(hwid, sizeof(hwid), "hw"); + if (ma_strcat_s(hwid, sizeof(hwid), deviceID.alsa) == 0) { + if (((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, hwid, stream, openMode) == 0) { + isDeviceOpen = MA_TRUE; + } + } + } + } + + if (!isDeviceOpen) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_open() failed.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } + } + *ppPCM = pPCM; return MA_SUCCESS; } -#endif /* ALSA */ +static ma_bool32 ma_context_is_device_id_equal__alsa(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) +{ + MA_ASSERT(pContext != NULL); + MA_ASSERT(pID0 != NULL); + MA_ASSERT(pID1 != NULL); + (void)pContext; -/****************************************************************************** + return ma_strcmp(pID0->alsa, pID1->alsa) == 0; +} + +static ma_result ma_context_enumerate_devices__alsa(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +{ + ma_bool32 cbResult = MA_TRUE; + char** ppDeviceHints; + ma_device_id* pUniqueIDs = NULL; + ma_uint32 uniqueIDCount = 0; + char** ppNextDeviceHint; + + MA_ASSERT(pContext != NULL); + MA_ASSERT(callback != NULL); + + ma_mutex_lock(&pContext->alsa.internalDeviceEnumLock); + + if (((ma_snd_device_name_hint_proc)pContext->alsa.snd_device_name_hint)(-1, "pcm", (void***)&ppDeviceHints) < 0) { + ma_mutex_unlock(&pContext->alsa.internalDeviceEnumLock); + return MA_NO_BACKEND; + } + + ppNextDeviceHint = ppDeviceHints; + while (*ppNextDeviceHint != NULL) { + char* NAME = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "NAME"); + char* DESC = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "DESC"); + char* IOID = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "IOID"); + ma_device_type deviceType = ma_device_type_playback; + ma_bool32 stopEnumeration = MA_FALSE; + char hwid[sizeof(pUniqueIDs->alsa)]; + ma_device_info deviceInfo; + + if ((IOID == NULL || ma_strcmp(IOID, "Output") == 0)) { + deviceType = ma_device_type_playback; + } + if ((IOID != NULL && ma_strcmp(IOID, "Input" ) == 0)) { + deviceType = ma_device_type_capture; + } + + if (NAME != NULL) { + if (pContext->alsa.useVerboseDeviceEnumeration) { + /* Verbose mode. Use the name exactly as-is. */ + ma_strncpy_s(hwid, sizeof(hwid), NAME, (size_t)-1); + } else { + /* Simplified mode. Use ":%d,%d" format. */ + if (ma_convert_device_name_to_hw_format__alsa(pContext, hwid, sizeof(hwid), NAME) == 0) { + /* + At this point, hwid looks like "hw:0,0". In simplified enumeration mode, we actually want to strip off the + plugin name so it looks like ":0,0". The reason for this is that this special format is detected at device + initialization time and is used as an indicator to try and use the most appropriate plugin depending on the + device type and sharing mode. + */ + char* dst = hwid; + char* src = hwid+2; + while ((*dst++ = *src++)); + } else { + /* Conversion to "hw:%d,%d" failed. Just use the name as-is. */ + ma_strncpy_s(hwid, sizeof(hwid), NAME, (size_t)-1); + } + + if (ma_does_id_exist_in_list__alsa(pUniqueIDs, uniqueIDCount, hwid)) { + goto next_device; /* The device has already been enumerated. Move on to the next one. */ + } else { + /* The device has not yet been enumerated. Make sure it's added to our list so that it's not enumerated again. */ + size_t oldCapacity = sizeof(*pUniqueIDs) * uniqueIDCount; + size_t newCapacity = sizeof(*pUniqueIDs) * (uniqueIDCount + 1); + ma_device_id* pNewUniqueIDs = (ma_device_id*)ma__realloc_from_callbacks(pUniqueIDs, newCapacity, oldCapacity, &pContext->allocationCallbacks); + if (pNewUniqueIDs == NULL) { + goto next_device; /* Failed to allocate memory. */ + } + + pUniqueIDs = pNewUniqueIDs; + MA_COPY_MEMORY(pUniqueIDs[uniqueIDCount].alsa, hwid, sizeof(hwid)); + uniqueIDCount += 1; + } + } + } else { + MA_ZERO_MEMORY(hwid, sizeof(hwid)); + } + + MA_ZERO_OBJECT(&deviceInfo); + ma_strncpy_s(deviceInfo.id.alsa, sizeof(deviceInfo.id.alsa), hwid, (size_t)-1); + + /* + DESC is the friendly name. We treat this slightly differently depending on whether or not we are using verbose + device enumeration. In verbose mode we want to take the entire description so that the end-user can distinguish + between the subdevices of each card/dev pair. In simplified mode, however, we only want the first part of the + description. + + The value in DESC seems to be split into two lines, with the first line being the name of the device and the + second line being a description of the device. I don't like having the description be across two lines because + it makes formatting ugly and annoying. I'm therefore deciding to put it all on a single line with the second line + being put into parentheses. In simplified mode I'm just stripping the second line entirely. + */ + if (DESC != NULL) { + int lfPos; + const char* line2 = ma_find_char(DESC, '\n', &lfPos); + if (line2 != NULL) { + line2 += 1; /* Skip past the new-line character. */ + + if (pContext->alsa.useVerboseDeviceEnumeration) { + /* Verbose mode. Put the second line in brackets. */ + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), DESC, lfPos); + ma_strcat_s (deviceInfo.name, sizeof(deviceInfo.name), " ("); + ma_strcat_s (deviceInfo.name, sizeof(deviceInfo.name), line2); + ma_strcat_s (deviceInfo.name, sizeof(deviceInfo.name), ")"); + } else { + /* Simplified mode. Strip the second line entirely. */ + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), DESC, lfPos); + } + } else { + /* There's no second line. Just copy the whole description. */ + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), DESC, (size_t)-1); + } + } + + if (!ma_is_device_blacklisted__alsa(deviceType, NAME)) { + cbResult = callback(pContext, deviceType, &deviceInfo, pUserData); + } + + /* + Some devices are both playback and capture, but they are only enumerated by ALSA once. We need to fire the callback + again for the other device type in this case. We do this for known devices. + */ + if (cbResult) { + if (ma_is_common_device_name__alsa(NAME)) { + if (deviceType == ma_device_type_playback) { + if (!ma_is_capture_device_blacklisted__alsa(NAME)) { + cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); + } + } else { + if (!ma_is_playback_device_blacklisted__alsa(NAME)) { + cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); + } + } + } + } + + if (cbResult == MA_FALSE) { + stopEnumeration = MA_TRUE; + } + + next_device: + free(NAME); + free(DESC); + free(IOID); + ppNextDeviceHint += 1; + + /* We need to stop enumeration if the callback returned false. */ + if (stopEnumeration) { + break; + } + } + + ma__free_from_callbacks(pUniqueIDs, &pContext->allocationCallbacks); + ((ma_snd_device_name_free_hint_proc)pContext->alsa.snd_device_name_free_hint)((void**)ppDeviceHints); -PulseAudio Backend + ma_mutex_unlock(&pContext->alsa.internalDeviceEnumLock); -******************************************************************************/ -#ifdef MA_HAS_PULSEAUDIO -/* -It is assumed pulseaudio.h is available when compile-time linking is being used. We use this for type safety when using -compile time linking (we don't have this luxury when using runtime linking without headers). + return MA_SUCCESS; +} -When using compile time linking, each of our ma_* equivalents should use the sames types as defined by the header. The -reason for this is that it allow us to take advantage of proper type safety. -*/ -#ifdef MA_NO_RUNTIME_LINKING -#include -#define MA_PA_OK PA_OK -#define MA_PA_ERR_ACCESS PA_ERR_ACCESS -#define MA_PA_ERR_INVALID PA_ERR_INVALID -#define MA_PA_ERR_NOENTITY PA_ERR_NOENTITY +typedef struct +{ + ma_device_type deviceType; + const ma_device_id* pDeviceID; + ma_share_mode shareMode; + ma_device_info* pDeviceInfo; + ma_bool32 foundDevice; +} ma_context_get_device_info_enum_callback_data__alsa; -#define MA_PA_CHANNELS_MAX PA_CHANNELS_MAX -#define MA_PA_RATE_MAX PA_RATE_MAX +static ma_bool32 ma_context_get_device_info_enum_callback__alsa(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pDeviceInfo, void* pUserData) +{ + ma_context_get_device_info_enum_callback_data__alsa* pData = (ma_context_get_device_info_enum_callback_data__alsa*)pUserData; + MA_ASSERT(pData != NULL); -typedef pa_context_flags_t ma_pa_context_flags_t; -#define MA_PA_CONTEXT_NOFLAGS PA_CONTEXT_NOFLAGS -#define MA_PA_CONTEXT_NOAUTOSPAWN PA_CONTEXT_NOAUTOSPAWN -#define MA_PA_CONTEXT_NOFAIL PA_CONTEXT_NOFAIL + if (pData->pDeviceID == NULL && ma_strcmp(pDeviceInfo->id.alsa, "default") == 0) { + ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pDeviceInfo->name, (size_t)-1); + pData->foundDevice = MA_TRUE; + } else { + if (pData->deviceType == deviceType && ma_context_is_device_id_equal__alsa(pContext, pData->pDeviceID, &pDeviceInfo->id)) { + ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pDeviceInfo->name, (size_t)-1); + pData->foundDevice = MA_TRUE; + } + } -typedef pa_stream_flags_t ma_pa_stream_flags_t; -#define MA_PA_STREAM_NOFLAGS PA_STREAM_NOFLAGS -#define MA_PA_STREAM_START_CORKED PA_STREAM_START_CORKED -#define MA_PA_STREAM_INTERPOLATE_TIMING PA_STREAM_INTERPOLATE_TIMING -#define MA_PA_STREAM_NOT_MONOTONIC PA_STREAM_NOT_MONOTONIC -#define MA_PA_STREAM_AUTO_TIMING_UPDATE PA_STREAM_AUTO_TIMING_UPDATE -#define MA_PA_STREAM_NO_REMAP_CHANNELS PA_STREAM_NO_REMAP_CHANNELS -#define MA_PA_STREAM_NO_REMIX_CHANNELS PA_STREAM_NO_REMIX_CHANNELS -#define MA_PA_STREAM_FIX_FORMAT PA_STREAM_FIX_FORMAT -#define MA_PA_STREAM_FIX_RATE PA_STREAM_FIX_RATE -#define MA_PA_STREAM_FIX_CHANNELS PA_STREAM_FIX_CHANNELS -#define MA_PA_STREAM_DONT_MOVE PA_STREAM_DONT_MOVE -#define MA_PA_STREAM_VARIABLE_RATE PA_STREAM_VARIABLE_RATE -#define MA_PA_STREAM_PEAK_DETECT PA_STREAM_PEAK_DETECT -#define MA_PA_STREAM_START_MUTED PA_STREAM_START_MUTED -#define MA_PA_STREAM_ADJUST_LATENCY PA_STREAM_ADJUST_LATENCY -#define MA_PA_STREAM_EARLY_REQUESTS PA_STREAM_EARLY_REQUESTS -#define MA_PA_STREAM_DONT_INHIBIT_AUTO_SUSPEND PA_STREAM_DONT_INHIBIT_AUTO_SUSPEND -#define MA_PA_STREAM_START_UNMUTED PA_STREAM_START_UNMUTED -#define MA_PA_STREAM_FAIL_ON_SUSPEND PA_STREAM_FAIL_ON_SUSPEND -#define MA_PA_STREAM_RELATIVE_VOLUME PA_STREAM_RELATIVE_VOLUME -#define MA_PA_STREAM_PASSTHROUGH PA_STREAM_PASSTHROUGH + /* Keep enumerating until we have found the device. */ + return !pData->foundDevice; +} -typedef pa_sink_flags_t ma_pa_sink_flags_t; -#define MA_PA_SINK_NOFLAGS PA_SINK_NOFLAGS -#define MA_PA_SINK_HW_VOLUME_CTRL PA_SINK_HW_VOLUME_CTRL -#define MA_PA_SINK_LATENCY PA_SINK_LATENCY -#define MA_PA_SINK_HARDWARE PA_SINK_HARDWARE -#define MA_PA_SINK_NETWORK PA_SINK_NETWORK -#define MA_PA_SINK_HW_MUTE_CTRL PA_SINK_HW_MUTE_CTRL -#define MA_PA_SINK_DECIBEL_VOLUME PA_SINK_DECIBEL_VOLUME -#define MA_PA_SINK_FLAT_VOLUME PA_SINK_FLAT_VOLUME -#define MA_PA_SINK_DYNAMIC_LATENCY PA_SINK_DYNAMIC_LATENCY -#define MA_PA_SINK_SET_FORMATS PA_SINK_SET_FORMATS +static ma_result ma_context_get_device_info__alsa(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) +{ + ma_context_get_device_info_enum_callback_data__alsa data; + ma_result result; + ma_snd_pcm_t* pPCM; + ma_snd_pcm_hw_params_t* pHWParams; + ma_snd_pcm_format_mask_t* pFormatMask; + int sampleRateDir = 0; -typedef pa_source_flags_t ma_pa_source_flags_t; -#define MA_PA_SOURCE_NOFLAGS PA_SOURCE_NOFLAGS -#define MA_PA_SOURCE_HW_VOLUME_CTRL PA_SOURCE_HW_VOLUME_CTRL -#define MA_PA_SOURCE_LATENCY PA_SOURCE_LATENCY -#define MA_PA_SOURCE_HARDWARE PA_SOURCE_HARDWARE -#define MA_PA_SOURCE_NETWORK PA_SOURCE_NETWORK -#define MA_PA_SOURCE_HW_MUTE_CTRL PA_SOURCE_HW_MUTE_CTRL -#define MA_PA_SOURCE_DECIBEL_VOLUME PA_SOURCE_DECIBEL_VOLUME -#define MA_PA_SOURCE_DYNAMIC_LATENCY PA_SOURCE_DYNAMIC_LATENCY -#define MA_PA_SOURCE_FLAT_VOLUME PA_SOURCE_FLAT_VOLUME + MA_ASSERT(pContext != NULL); -typedef pa_context_state_t ma_pa_context_state_t; -#define MA_PA_CONTEXT_UNCONNECTED PA_CONTEXT_UNCONNECTED -#define MA_PA_CONTEXT_CONNECTING PA_CONTEXT_CONNECTING -#define MA_PA_CONTEXT_AUTHORIZING PA_CONTEXT_AUTHORIZING -#define MA_PA_CONTEXT_SETTING_NAME PA_CONTEXT_SETTING_NAME -#define MA_PA_CONTEXT_READY PA_CONTEXT_READY -#define MA_PA_CONTEXT_FAILED PA_CONTEXT_FAILED -#define MA_PA_CONTEXT_TERMINATED PA_CONTEXT_TERMINATED + /* We just enumerate to find basic information about the device. */ + data.deviceType = deviceType; + data.pDeviceID = pDeviceID; + data.shareMode = shareMode; + data.pDeviceInfo = pDeviceInfo; + data.foundDevice = MA_FALSE; + result = ma_context_enumerate_devices__alsa(pContext, ma_context_get_device_info_enum_callback__alsa, &data); + if (result != MA_SUCCESS) { + return result; + } -typedef pa_stream_state_t ma_pa_stream_state_t; -#define MA_PA_STREAM_UNCONNECTED PA_STREAM_UNCONNECTED -#define MA_PA_STREAM_CREATING PA_STREAM_CREATING -#define MA_PA_STREAM_READY PA_STREAM_READY -#define MA_PA_STREAM_FAILED PA_STREAM_FAILED -#define MA_PA_STREAM_TERMINATED PA_STREAM_TERMINATED + if (!data.foundDevice) { + return MA_NO_DEVICE; + } -typedef pa_operation_state_t ma_pa_operation_state_t; -#define MA_PA_OPERATION_RUNNING PA_OPERATION_RUNNING -#define MA_PA_OPERATION_DONE PA_OPERATION_DONE -#define MA_PA_OPERATION_CANCELLED PA_OPERATION_CANCELLED + /* For detailed info we need to open the device. */ + result = ma_context_open_pcm__alsa(pContext, shareMode, deviceType, pDeviceID, &pPCM); + if (result != MA_SUCCESS) { + return result; + } -typedef pa_sink_state_t ma_pa_sink_state_t; -#define MA_PA_SINK_INVALID_STATE PA_SINK_INVALID_STATE -#define MA_PA_SINK_RUNNING PA_SINK_RUNNING -#define MA_PA_SINK_IDLE PA_SINK_IDLE -#define MA_PA_SINK_SUSPENDED PA_SINK_SUSPENDED + /* We need to initialize a HW parameters object in order to know what formats are supported. */ + pHWParams = (ma_snd_pcm_hw_params_t*)ma__calloc_from_callbacks(((ma_snd_pcm_hw_params_sizeof_proc)pContext->alsa.snd_pcm_hw_params_sizeof)(), &pContext->allocationCallbacks); + if (pHWParams == NULL) { + return MA_OUT_OF_MEMORY; + } -typedef pa_source_state_t ma_pa_source_state_t; -#define MA_PA_SOURCE_INVALID_STATE PA_SOURCE_INVALID_STATE -#define MA_PA_SOURCE_RUNNING PA_SOURCE_RUNNING -#define MA_PA_SOURCE_IDLE PA_SOURCE_IDLE -#define MA_PA_SOURCE_SUSPENDED PA_SOURCE_SUSPENDED + if (((ma_snd_pcm_hw_params_any_proc)pContext->alsa.snd_pcm_hw_params_any)(pPCM, pHWParams) < 0) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to initialize hardware parameters. snd_pcm_hw_params_any() failed.", MA_FAILED_TO_CONFIGURE_BACKEND_DEVICE); + } -typedef pa_seek_mode_t ma_pa_seek_mode_t; -#define MA_PA_SEEK_RELATIVE PA_SEEK_RELATIVE -#define MA_PA_SEEK_ABSOLUTE PA_SEEK_ABSOLUTE -#define MA_PA_SEEK_RELATIVE_ON_READ PA_SEEK_RELATIVE_ON_READ -#define MA_PA_SEEK_RELATIVE_END PA_SEEK_RELATIVE_END + ((ma_snd_pcm_hw_params_get_channels_min_proc)pContext->alsa.snd_pcm_hw_params_get_channels_min)(pHWParams, &pDeviceInfo->minChannels); + ((ma_snd_pcm_hw_params_get_channels_max_proc)pContext->alsa.snd_pcm_hw_params_get_channels_max)(pHWParams, &pDeviceInfo->maxChannels); + ((ma_snd_pcm_hw_params_get_rate_min_proc)pContext->alsa.snd_pcm_hw_params_get_rate_min)(pHWParams, &pDeviceInfo->minSampleRate, &sampleRateDir); + ((ma_snd_pcm_hw_params_get_rate_max_proc)pContext->alsa.snd_pcm_hw_params_get_rate_max)(pHWParams, &pDeviceInfo->maxSampleRate, &sampleRateDir); -typedef pa_channel_position_t ma_pa_channel_position_t; -#define MA_PA_CHANNEL_POSITION_INVALID PA_CHANNEL_POSITION_INVALID -#define MA_PA_CHANNEL_POSITION_MONO PA_CHANNEL_POSITION_MONO -#define MA_PA_CHANNEL_POSITION_FRONT_LEFT PA_CHANNEL_POSITION_FRONT_LEFT -#define MA_PA_CHANNEL_POSITION_FRONT_RIGHT PA_CHANNEL_POSITION_FRONT_RIGHT -#define MA_PA_CHANNEL_POSITION_FRONT_CENTER PA_CHANNEL_POSITION_FRONT_CENTER -#define MA_PA_CHANNEL_POSITION_REAR_CENTER PA_CHANNEL_POSITION_REAR_CENTER -#define MA_PA_CHANNEL_POSITION_REAR_LEFT PA_CHANNEL_POSITION_REAR_LEFT -#define MA_PA_CHANNEL_POSITION_REAR_RIGHT PA_CHANNEL_POSITION_REAR_RIGHT -#define MA_PA_CHANNEL_POSITION_LFE PA_CHANNEL_POSITION_LFE -#define MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER -#define MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER -#define MA_PA_CHANNEL_POSITION_SIDE_LEFT PA_CHANNEL_POSITION_SIDE_LEFT -#define MA_PA_CHANNEL_POSITION_SIDE_RIGHT PA_CHANNEL_POSITION_SIDE_RIGHT -#define MA_PA_CHANNEL_POSITION_AUX0 PA_CHANNEL_POSITION_AUX0 -#define MA_PA_CHANNEL_POSITION_AUX1 PA_CHANNEL_POSITION_AUX1 -#define MA_PA_CHANNEL_POSITION_AUX2 PA_CHANNEL_POSITION_AUX2 -#define MA_PA_CHANNEL_POSITION_AUX3 PA_CHANNEL_POSITION_AUX3 -#define MA_PA_CHANNEL_POSITION_AUX4 PA_CHANNEL_POSITION_AUX4 -#define MA_PA_CHANNEL_POSITION_AUX5 PA_CHANNEL_POSITION_AUX5 -#define MA_PA_CHANNEL_POSITION_AUX6 PA_CHANNEL_POSITION_AUX6 -#define MA_PA_CHANNEL_POSITION_AUX7 PA_CHANNEL_POSITION_AUX7 -#define MA_PA_CHANNEL_POSITION_AUX8 PA_CHANNEL_POSITION_AUX8 -#define MA_PA_CHANNEL_POSITION_AUX9 PA_CHANNEL_POSITION_AUX9 -#define MA_PA_CHANNEL_POSITION_AUX10 PA_CHANNEL_POSITION_AUX10 -#define MA_PA_CHANNEL_POSITION_AUX11 PA_CHANNEL_POSITION_AUX11 -#define MA_PA_CHANNEL_POSITION_AUX12 PA_CHANNEL_POSITION_AUX12 -#define MA_PA_CHANNEL_POSITION_AUX13 PA_CHANNEL_POSITION_AUX13 -#define MA_PA_CHANNEL_POSITION_AUX14 PA_CHANNEL_POSITION_AUX14 -#define MA_PA_CHANNEL_POSITION_AUX15 PA_CHANNEL_POSITION_AUX15 -#define MA_PA_CHANNEL_POSITION_AUX16 PA_CHANNEL_POSITION_AUX16 -#define MA_PA_CHANNEL_POSITION_AUX17 PA_CHANNEL_POSITION_AUX17 -#define MA_PA_CHANNEL_POSITION_AUX18 PA_CHANNEL_POSITION_AUX18 -#define MA_PA_CHANNEL_POSITION_AUX19 PA_CHANNEL_POSITION_AUX19 -#define MA_PA_CHANNEL_POSITION_AUX20 PA_CHANNEL_POSITION_AUX20 -#define MA_PA_CHANNEL_POSITION_AUX21 PA_CHANNEL_POSITION_AUX21 -#define MA_PA_CHANNEL_POSITION_AUX22 PA_CHANNEL_POSITION_AUX22 -#define MA_PA_CHANNEL_POSITION_AUX23 PA_CHANNEL_POSITION_AUX23 -#define MA_PA_CHANNEL_POSITION_AUX24 PA_CHANNEL_POSITION_AUX24 -#define MA_PA_CHANNEL_POSITION_AUX25 PA_CHANNEL_POSITION_AUX25 -#define MA_PA_CHANNEL_POSITION_AUX26 PA_CHANNEL_POSITION_AUX26 -#define MA_PA_CHANNEL_POSITION_AUX27 PA_CHANNEL_POSITION_AUX27 -#define MA_PA_CHANNEL_POSITION_AUX28 PA_CHANNEL_POSITION_AUX28 -#define MA_PA_CHANNEL_POSITION_AUX29 PA_CHANNEL_POSITION_AUX29 -#define MA_PA_CHANNEL_POSITION_AUX30 PA_CHANNEL_POSITION_AUX30 -#define MA_PA_CHANNEL_POSITION_AUX31 PA_CHANNEL_POSITION_AUX31 -#define MA_PA_CHANNEL_POSITION_TOP_CENTER PA_CHANNEL_POSITION_TOP_CENTER -#define MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT PA_CHANNEL_POSITION_TOP_FRONT_LEFT -#define MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT PA_CHANNEL_POSITION_TOP_FRONT_RIGHT -#define MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER PA_CHANNEL_POSITION_TOP_FRONT_CENTER -#define MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT PA_CHANNEL_POSITION_TOP_REAR_LEFT -#define MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT PA_CHANNEL_POSITION_TOP_REAR_RIGHT -#define MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER PA_CHANNEL_POSITION_TOP_REAR_CENTER -#define MA_PA_CHANNEL_POSITION_LEFT PA_CHANNEL_POSITION_LEFT -#define MA_PA_CHANNEL_POSITION_RIGHT PA_CHANNEL_POSITION_RIGHT -#define MA_PA_CHANNEL_POSITION_CENTER PA_CHANNEL_POSITION_CENTER -#define MA_PA_CHANNEL_POSITION_SUBWOOFER PA_CHANNEL_POSITION_SUBWOOFER + /* Formats. */ + pFormatMask = (ma_snd_pcm_format_mask_t*)ma__calloc_from_callbacks(((ma_snd_pcm_format_mask_sizeof_proc)pContext->alsa.snd_pcm_format_mask_sizeof)(), &pContext->allocationCallbacks); + if (pFormatMask == NULL) { + return MA_OUT_OF_MEMORY; + } -typedef pa_channel_map_def_t ma_pa_channel_map_def_t; -#define MA_PA_CHANNEL_MAP_AIFF PA_CHANNEL_MAP_AIFF -#define MA_PA_CHANNEL_MAP_ALSA PA_CHANNEL_MAP_ALSA -#define MA_PA_CHANNEL_MAP_AUX PA_CHANNEL_MAP_AUX -#define MA_PA_CHANNEL_MAP_WAVEEX PA_CHANNEL_MAP_WAVEEX -#define MA_PA_CHANNEL_MAP_OSS PA_CHANNEL_MAP_OSS -#define MA_PA_CHANNEL_MAP_DEFAULT PA_CHANNEL_MAP_DEFAULT + ((ma_snd_pcm_hw_params_get_format_mask_proc)pContext->alsa.snd_pcm_hw_params_get_format_mask)(pHWParams, pFormatMask); -typedef pa_sample_format_t ma_pa_sample_format_t; -#define MA_PA_SAMPLE_INVALID PA_SAMPLE_INVALID -#define MA_PA_SAMPLE_U8 PA_SAMPLE_U8 -#define MA_PA_SAMPLE_ALAW PA_SAMPLE_ALAW -#define MA_PA_SAMPLE_ULAW PA_SAMPLE_ULAW -#define MA_PA_SAMPLE_S16LE PA_SAMPLE_S16LE -#define MA_PA_SAMPLE_S16BE PA_SAMPLE_S16BE -#define MA_PA_SAMPLE_FLOAT32LE PA_SAMPLE_FLOAT32LE -#define MA_PA_SAMPLE_FLOAT32BE PA_SAMPLE_FLOAT32BE -#define MA_PA_SAMPLE_S32LE PA_SAMPLE_S32LE -#define MA_PA_SAMPLE_S32BE PA_SAMPLE_S32BE -#define MA_PA_SAMPLE_S24LE PA_SAMPLE_S24LE -#define MA_PA_SAMPLE_S24BE PA_SAMPLE_S24BE -#define MA_PA_SAMPLE_S24_32LE PA_SAMPLE_S24_32LE -#define MA_PA_SAMPLE_S24_32BE PA_SAMPLE_S24_32BE + pDeviceInfo->formatCount = 0; + if (((ma_snd_pcm_format_mask_test_proc)pContext->alsa.snd_pcm_format_mask_test)(pFormatMask, MA_SND_PCM_FORMAT_U8)) { + pDeviceInfo->formats[pDeviceInfo->formatCount++] = ma_format_u8; + } + if (((ma_snd_pcm_format_mask_test_proc)pContext->alsa.snd_pcm_format_mask_test)(pFormatMask, MA_SND_PCM_FORMAT_S16_LE)) { + pDeviceInfo->formats[pDeviceInfo->formatCount++] = ma_format_s16; + } + if (((ma_snd_pcm_format_mask_test_proc)pContext->alsa.snd_pcm_format_mask_test)(pFormatMask, MA_SND_PCM_FORMAT_S24_3LE)) { + pDeviceInfo->formats[pDeviceInfo->formatCount++] = ma_format_s24; + } + if (((ma_snd_pcm_format_mask_test_proc)pContext->alsa.snd_pcm_format_mask_test)(pFormatMask, MA_SND_PCM_FORMAT_S32_LE)) { + pDeviceInfo->formats[pDeviceInfo->formatCount++] = ma_format_s32; + } + if (((ma_snd_pcm_format_mask_test_proc)pContext->alsa.snd_pcm_format_mask_test)(pFormatMask, MA_SND_PCM_FORMAT_FLOAT_LE)) { + pDeviceInfo->formats[pDeviceInfo->formatCount++] = ma_format_f32; + } -typedef pa_mainloop ma_pa_mainloop; -typedef pa_mainloop_api ma_pa_mainloop_api; -typedef pa_context ma_pa_context; -typedef pa_operation ma_pa_operation; -typedef pa_stream ma_pa_stream; -typedef pa_spawn_api ma_pa_spawn_api; -typedef pa_buffer_attr ma_pa_buffer_attr; -typedef pa_channel_map ma_pa_channel_map; -typedef pa_cvolume ma_pa_cvolume; -typedef pa_sample_spec ma_pa_sample_spec; -typedef pa_sink_info ma_pa_sink_info; -typedef pa_source_info ma_pa_source_info; + ma__free_from_callbacks(pFormatMask, &pContext->allocationCallbacks); + ma__free_from_callbacks(pHWParams, &pContext->allocationCallbacks); -typedef pa_context_notify_cb_t ma_pa_context_notify_cb_t; -typedef pa_sink_info_cb_t ma_pa_sink_info_cb_t; -typedef pa_source_info_cb_t ma_pa_source_info_cb_t; -typedef pa_stream_success_cb_t ma_pa_stream_success_cb_t; -typedef pa_stream_request_cb_t ma_pa_stream_request_cb_t; -typedef pa_free_cb_t ma_pa_free_cb_t; -#else -#define MA_PA_OK 0 -#define MA_PA_ERR_ACCESS 1 -#define MA_PA_ERR_INVALID 2 -#define MA_PA_ERR_NOENTITY 5 + ((ma_snd_pcm_close_proc)pContext->alsa.snd_pcm_close)(pPCM); + return MA_SUCCESS; +} -#define MA_PA_CHANNELS_MAX 32 -#define MA_PA_RATE_MAX 384000 -typedef int ma_pa_context_flags_t; -#define MA_PA_CONTEXT_NOFLAGS 0x00000000 -#define MA_PA_CONTEXT_NOAUTOSPAWN 0x00000001 -#define MA_PA_CONTEXT_NOFAIL 0x00000002 +#if 0 +/* +Waits for a number of frames to become available for either capture or playback. The return +value is the number of frames available. -typedef int ma_pa_stream_flags_t; -#define MA_PA_STREAM_NOFLAGS 0x00000000 -#define MA_PA_STREAM_START_CORKED 0x00000001 -#define MA_PA_STREAM_INTERPOLATE_TIMING 0x00000002 -#define MA_PA_STREAM_NOT_MONOTONIC 0x00000004 -#define MA_PA_STREAM_AUTO_TIMING_UPDATE 0x00000008 -#define MA_PA_STREAM_NO_REMAP_CHANNELS 0x00000010 -#define MA_PA_STREAM_NO_REMIX_CHANNELS 0x00000020 -#define MA_PA_STREAM_FIX_FORMAT 0x00000040 -#define MA_PA_STREAM_FIX_RATE 0x00000080 -#define MA_PA_STREAM_FIX_CHANNELS 0x00000100 -#define MA_PA_STREAM_DONT_MOVE 0x00000200 -#define MA_PA_STREAM_VARIABLE_RATE 0x00000400 -#define MA_PA_STREAM_PEAK_DETECT 0x00000800 -#define MA_PA_STREAM_START_MUTED 0x00001000 -#define MA_PA_STREAM_ADJUST_LATENCY 0x00002000 -#define MA_PA_STREAM_EARLY_REQUESTS 0x00004000 -#define MA_PA_STREAM_DONT_INHIBIT_AUTO_SUSPEND 0x00008000 -#define MA_PA_STREAM_START_UNMUTED 0x00010000 -#define MA_PA_STREAM_FAIL_ON_SUSPEND 0x00020000 -#define MA_PA_STREAM_RELATIVE_VOLUME 0x00040000 -#define MA_PA_STREAM_PASSTHROUGH 0x00080000 +This will return early if the main loop is broken with ma_device__break_main_loop(). +*/ +static ma_uint32 ma_device__wait_for_frames__alsa(ma_device* pDevice, ma_bool32* pRequiresRestart) +{ + MA_ASSERT(pDevice != NULL); -typedef int ma_pa_sink_flags_t; -#define MA_PA_SINK_NOFLAGS 0x00000000 -#define MA_PA_SINK_HW_VOLUME_CTRL 0x00000001 -#define MA_PA_SINK_LATENCY 0x00000002 -#define MA_PA_SINK_HARDWARE 0x00000004 -#define MA_PA_SINK_NETWORK 0x00000008 -#define MA_PA_SINK_HW_MUTE_CTRL 0x00000010 -#define MA_PA_SINK_DECIBEL_VOLUME 0x00000020 -#define MA_PA_SINK_FLAT_VOLUME 0x00000040 -#define MA_PA_SINK_DYNAMIC_LATENCY 0x00000080 -#define MA_PA_SINK_SET_FORMATS 0x00000100 + if (pRequiresRestart) *pRequiresRestart = MA_FALSE; -typedef int ma_pa_source_flags_t; -#define MA_PA_SOURCE_NOFLAGS 0x00000000 -#define MA_PA_SOURCE_HW_VOLUME_CTRL 0x00000001 -#define MA_PA_SOURCE_LATENCY 0x00000002 -#define MA_PA_SOURCE_HARDWARE 0x00000004 -#define MA_PA_SOURCE_NETWORK 0x00000008 -#define MA_PA_SOURCE_HW_MUTE_CTRL 0x00000010 -#define MA_PA_SOURCE_DECIBEL_VOLUME 0x00000020 -#define MA_PA_SOURCE_DYNAMIC_LATENCY 0x00000040 -#define MA_PA_SOURCE_FLAT_VOLUME 0x00000080 + /* I want it so that this function returns the period size in frames. We just wait until that number of frames are available and then return. */ + ma_uint32 periodSizeInFrames = pDevice->bufferSizeInFrames / pDevice->periods; + while (!pDevice->alsa.breakFromMainLoop) { + ma_snd_pcm_sframes_t framesAvailable = ((ma_snd_pcm_avail_update_proc)pDevice->pContext->alsa.snd_pcm_avail_update)((ma_snd_pcm_t*)pDevice->alsa.pPCM); + if (framesAvailable < 0) { + if (framesAvailable == -EPIPE) { + if (((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCM, framesAvailable, MA_TRUE) < 0) { + return 0; + } -typedef int ma_pa_context_state_t; -#define MA_PA_CONTEXT_UNCONNECTED 0 -#define MA_PA_CONTEXT_CONNECTING 1 -#define MA_PA_CONTEXT_AUTHORIZING 2 -#define MA_PA_CONTEXT_SETTING_NAME 3 -#define MA_PA_CONTEXT_READY 4 -#define MA_PA_CONTEXT_FAILED 5 -#define MA_PA_CONTEXT_TERMINATED 6 + /* A device recovery means a restart for mmap mode. */ + if (pRequiresRestart) { + *pRequiresRestart = MA_TRUE; + } -typedef int ma_pa_stream_state_t; -#define MA_PA_STREAM_UNCONNECTED 0 -#define MA_PA_STREAM_CREATING 1 -#define MA_PA_STREAM_READY 2 -#define MA_PA_STREAM_FAILED 3 -#define MA_PA_STREAM_TERMINATED 4 + /* Try again, but if it fails this time just return an error. */ + framesAvailable = ((ma_snd_pcm_avail_update_proc)pDevice->pContext->alsa.snd_pcm_avail_update)((ma_snd_pcm_t*)pDevice->alsa.pPCM); + if (framesAvailable < 0) { + return 0; + } + } + } -typedef int ma_pa_operation_state_t; -#define MA_PA_OPERATION_RUNNING 0 -#define MA_PA_OPERATION_DONE 1 -#define MA_PA_OPERATION_CANCELLED 2 + if (framesAvailable >= periodSizeInFrames) { + return periodSizeInFrames; + } -typedef int ma_pa_sink_state_t; -#define MA_PA_SINK_INVALID_STATE -1 -#define MA_PA_SINK_RUNNING 0 -#define MA_PA_SINK_IDLE 1 -#define MA_PA_SINK_SUSPENDED 2 + if (framesAvailable < periodSizeInFrames) { + /* Less than a whole period is available so keep waiting. */ + int waitResult = ((ma_snd_pcm_wait_proc)pDevice->pContext->alsa.snd_pcm_wait)((ma_snd_pcm_t*)pDevice->alsa.pPCM, -1); + if (waitResult < 0) { + if (waitResult == -EPIPE) { + if (((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCM, waitResult, MA_TRUE) < 0) { + return 0; + } -typedef int ma_pa_source_state_t; -#define MA_PA_SOURCE_INVALID_STATE -1 -#define MA_PA_SOURCE_RUNNING 0 -#define MA_PA_SOURCE_IDLE 1 -#define MA_PA_SOURCE_SUSPENDED 2 + /* A device recovery means a restart for mmap mode. */ + if (pRequiresRestart) { + *pRequiresRestart = MA_TRUE; + } + } + } + } + } -typedef int ma_pa_seek_mode_t; -#define MA_PA_SEEK_RELATIVE 0 -#define MA_PA_SEEK_ABSOLUTE 1 -#define MA_PA_SEEK_RELATIVE_ON_READ 2 -#define MA_PA_SEEK_RELATIVE_END 3 + /* We'll get here if the loop was terminated. Just return whatever's available. */ + ma_snd_pcm_sframes_t framesAvailable = ((ma_snd_pcm_avail_update_proc)pDevice->pContext->alsa.snd_pcm_avail_update)((ma_snd_pcm_t*)pDevice->alsa.pPCM); + if (framesAvailable < 0) { + return 0; + } -typedef int ma_pa_channel_position_t; -#define MA_PA_CHANNEL_POSITION_INVALID -1 -#define MA_PA_CHANNEL_POSITION_MONO 0 -#define MA_PA_CHANNEL_POSITION_FRONT_LEFT 1 -#define MA_PA_CHANNEL_POSITION_FRONT_RIGHT 2 -#define MA_PA_CHANNEL_POSITION_FRONT_CENTER 3 -#define MA_PA_CHANNEL_POSITION_REAR_CENTER 4 -#define MA_PA_CHANNEL_POSITION_REAR_LEFT 5 -#define MA_PA_CHANNEL_POSITION_REAR_RIGHT 6 -#define MA_PA_CHANNEL_POSITION_LFE 7 -#define MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER 8 -#define MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER 9 -#define MA_PA_CHANNEL_POSITION_SIDE_LEFT 10 -#define MA_PA_CHANNEL_POSITION_SIDE_RIGHT 11 -#define MA_PA_CHANNEL_POSITION_AUX0 12 -#define MA_PA_CHANNEL_POSITION_AUX1 13 -#define MA_PA_CHANNEL_POSITION_AUX2 14 -#define MA_PA_CHANNEL_POSITION_AUX3 15 -#define MA_PA_CHANNEL_POSITION_AUX4 16 -#define MA_PA_CHANNEL_POSITION_AUX5 17 -#define MA_PA_CHANNEL_POSITION_AUX6 18 -#define MA_PA_CHANNEL_POSITION_AUX7 19 -#define MA_PA_CHANNEL_POSITION_AUX8 20 -#define MA_PA_CHANNEL_POSITION_AUX9 21 -#define MA_PA_CHANNEL_POSITION_AUX10 22 -#define MA_PA_CHANNEL_POSITION_AUX11 23 -#define MA_PA_CHANNEL_POSITION_AUX12 24 -#define MA_PA_CHANNEL_POSITION_AUX13 25 -#define MA_PA_CHANNEL_POSITION_AUX14 26 -#define MA_PA_CHANNEL_POSITION_AUX15 27 -#define MA_PA_CHANNEL_POSITION_AUX16 28 -#define MA_PA_CHANNEL_POSITION_AUX17 29 -#define MA_PA_CHANNEL_POSITION_AUX18 30 -#define MA_PA_CHANNEL_POSITION_AUX19 31 -#define MA_PA_CHANNEL_POSITION_AUX20 32 -#define MA_PA_CHANNEL_POSITION_AUX21 33 -#define MA_PA_CHANNEL_POSITION_AUX22 34 -#define MA_PA_CHANNEL_POSITION_AUX23 35 -#define MA_PA_CHANNEL_POSITION_AUX24 36 -#define MA_PA_CHANNEL_POSITION_AUX25 37 -#define MA_PA_CHANNEL_POSITION_AUX26 38 -#define MA_PA_CHANNEL_POSITION_AUX27 39 -#define MA_PA_CHANNEL_POSITION_AUX28 40 -#define MA_PA_CHANNEL_POSITION_AUX29 41 -#define MA_PA_CHANNEL_POSITION_AUX30 42 -#define MA_PA_CHANNEL_POSITION_AUX31 43 -#define MA_PA_CHANNEL_POSITION_TOP_CENTER 44 -#define MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT 45 -#define MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT 46 -#define MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER 47 -#define MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT 48 -#define MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT 49 -#define MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER 50 -#define MA_PA_CHANNEL_POSITION_LEFT MA_PA_CHANNEL_POSITION_FRONT_LEFT -#define MA_PA_CHANNEL_POSITION_RIGHT MA_PA_CHANNEL_POSITION_FRONT_RIGHT -#define MA_PA_CHANNEL_POSITION_CENTER MA_PA_CHANNEL_POSITION_FRONT_CENTER -#define MA_PA_CHANNEL_POSITION_SUBWOOFER MA_PA_CHANNEL_POSITION_LFE + return framesAvailable; +} -typedef int ma_pa_channel_map_def_t; -#define MA_PA_CHANNEL_MAP_AIFF 0 -#define MA_PA_CHANNEL_MAP_ALSA 1 -#define MA_PA_CHANNEL_MAP_AUX 2 -#define MA_PA_CHANNEL_MAP_WAVEEX 3 -#define MA_PA_CHANNEL_MAP_OSS 4 -#define MA_PA_CHANNEL_MAP_DEFAULT MA_PA_CHANNEL_MAP_AIFF +static ma_bool32 ma_device_read_from_client_and_write__alsa(ma_device* pDevice) +{ + MA_ASSERT(pDevice != NULL); + if (!ma_device_is_started(pDevice) && ma_device__get_state(pDevice) != MA_STATE_STARTING) { + return MA_FALSE; + } + if (pDevice->alsa.breakFromMainLoop) { + return MA_FALSE; + } -typedef int ma_pa_sample_format_t; -#define MA_PA_SAMPLE_INVALID -1 -#define MA_PA_SAMPLE_U8 0 -#define MA_PA_SAMPLE_ALAW 1 -#define MA_PA_SAMPLE_ULAW 2 -#define MA_PA_SAMPLE_S16LE 3 -#define MA_PA_SAMPLE_S16BE 4 -#define MA_PA_SAMPLE_FLOAT32LE 5 -#define MA_PA_SAMPLE_FLOAT32BE 6 -#define MA_PA_SAMPLE_S32LE 7 -#define MA_PA_SAMPLE_S32BE 8 -#define MA_PA_SAMPLE_S24LE 9 -#define MA_PA_SAMPLE_S24BE 10 -#define MA_PA_SAMPLE_S24_32LE 11 -#define MA_PA_SAMPLE_S24_32BE 12 + if (pDevice->alsa.isUsingMMap) { + /* mmap. */ + ma_bool32 requiresRestart; + ma_uint32 framesAvailable = ma_device__wait_for_frames__alsa(pDevice, &requiresRestart); + if (framesAvailable == 0) { + return MA_FALSE; + } -typedef struct ma_pa_mainloop ma_pa_mainloop; -typedef struct ma_pa_mainloop_api ma_pa_mainloop_api; -typedef struct ma_pa_context ma_pa_context; -typedef struct ma_pa_operation ma_pa_operation; -typedef struct ma_pa_stream ma_pa_stream; -typedef struct ma_pa_spawn_api ma_pa_spawn_api; + /* Don't bother asking the client for more audio data if we're just stopping the device anyway. */ + if (pDevice->alsa.breakFromMainLoop) { + return MA_FALSE; + } -typedef struct -{ - ma_uint32 maxlength; - ma_uint32 tlength; - ma_uint32 prebuf; - ma_uint32 minreq; - ma_uint32 fragsize; -} ma_pa_buffer_attr; + const ma_snd_pcm_channel_area_t* pAreas; + ma_snd_pcm_uframes_t mappedOffset; + ma_snd_pcm_uframes_t mappedFrames = framesAvailable; + while (framesAvailable > 0) { + int result = ((ma_snd_pcm_mmap_begin_proc)pDevice->pContext->alsa.snd_pcm_mmap_begin)((ma_snd_pcm_t*)pDevice->alsa.pPCM, &pAreas, &mappedOffset, &mappedFrames); + if (result < 0) { + return MA_FALSE; + } -typedef struct -{ - ma_uint8 channels; - ma_pa_channel_position_t map[MA_PA_CHANNELS_MAX]; -} ma_pa_channel_map; + if (mappedFrames > 0) { + void* pBuffer = (ma_uint8*)pAreas[0].addr + ((pAreas[0].first + (mappedOffset * pAreas[0].step)) / 8); + ma_device__read_frames_from_client(pDevice, mappedFrames, pBuffer); + } -typedef struct -{ - ma_uint8 channels; - ma_uint32 values[MA_PA_CHANNELS_MAX]; -} ma_pa_cvolume; + result = ((ma_snd_pcm_mmap_commit_proc)pDevice->pContext->alsa.snd_pcm_mmap_commit)((ma_snd_pcm_t*)pDevice->alsa.pPCM, mappedOffset, mappedFrames); + if (result < 0 || (ma_snd_pcm_uframes_t)result != mappedFrames) { + ((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCM, result, MA_TRUE); + return MA_FALSE; + } -typedef struct -{ - ma_pa_sample_format_t format; - ma_uint32 rate; - ma_uint8 channels; -} ma_pa_sample_spec; + if (requiresRestart) { + if (((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCM) < 0) { + return MA_FALSE; + } + } -typedef struct -{ - const char* name; - ma_uint32 index; - const char* description; - ma_pa_sample_spec sample_spec; - ma_pa_channel_map channel_map; - ma_uint32 owner_module; - ma_pa_cvolume volume; - int mute; - ma_uint32 monitor_source; - const char* monitor_source_name; - ma_uint64 latency; - const char* driver; - ma_pa_sink_flags_t flags; - void* proplist; - ma_uint64 configured_latency; - ma_uint32 base_volume; - ma_pa_sink_state_t state; - ma_uint32 n_volume_steps; - ma_uint32 card; - ma_uint32 n_ports; - void** ports; - void* active_port; - ma_uint8 n_formats; - void** formats; -} ma_pa_sink_info; + if (framesAvailable >= mappedFrames) { + framesAvailable -= mappedFrames; + } else { + framesAvailable = 0; + } + } + } else { + /* readi/writei. */ + while (!pDevice->alsa.breakFromMainLoop) { + ma_uint32 framesAvailable = ma_device__wait_for_frames__alsa(pDevice, NULL); + if (framesAvailable == 0) { + continue; + } -typedef struct -{ - const char *name; - ma_uint32 index; - const char *description; - ma_pa_sample_spec sample_spec; - ma_pa_channel_map channel_map; - ma_uint32 owner_module; - ma_pa_cvolume volume; - int mute; - ma_uint32 monitor_of_sink; - const char *monitor_of_sink_name; - ma_uint64 latency; - const char *driver; - ma_pa_source_flags_t flags; - void* proplist; - ma_uint64 configured_latency; - ma_uint32 base_volume; - ma_pa_source_state_t state; - ma_uint32 n_volume_steps; - ma_uint32 card; - ma_uint32 n_ports; - void** ports; - void* active_port; - ma_uint8 n_formats; - void** formats; -} ma_pa_source_info; + /* Don't bother asking the client for more audio data if we're just stopping the device anyway. */ + if (pDevice->alsa.breakFromMainLoop) { + return MA_FALSE; + } -typedef void (* ma_pa_context_notify_cb_t)(ma_pa_context* c, void* userdata); -typedef void (* ma_pa_sink_info_cb_t) (ma_pa_context* c, const ma_pa_sink_info* i, int eol, void* userdata); -typedef void (* ma_pa_source_info_cb_t) (ma_pa_context* c, const ma_pa_source_info* i, int eol, void* userdata); -typedef void (* ma_pa_stream_success_cb_t)(ma_pa_stream* s, int success, void* userdata); -typedef void (* ma_pa_stream_request_cb_t)(ma_pa_stream* s, size_t nbytes, void* userdata); -typedef void (* ma_pa_free_cb_t) (void* p); -#endif + ma_device__read_frames_from_client(pDevice, framesAvailable, pDevice->alsa.pIntermediaryBuffer); + ma_snd_pcm_sframes_t framesWritten = ((ma_snd_pcm_writei_proc)pDevice->pContext->alsa.snd_pcm_writei)((ma_snd_pcm_t*)pDevice->alsa.pPCM, pDevice->alsa.pIntermediaryBuffer, framesAvailable); + if (framesWritten < 0) { + if (framesWritten == -EAGAIN) { + continue; /* Just keep trying... */ + } else if (framesWritten == -EPIPE) { + /* Underrun. Just recover and try writing again. */ + if (((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCM, framesWritten, MA_TRUE) < 0) { + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to recover device after underrun.", MA_FAILED_TO_START_BACKEND_DEVICE); + return MA_FALSE; + } -typedef ma_pa_mainloop* (* ma_pa_mainloop_new_proc) (); -typedef void (* ma_pa_mainloop_free_proc) (ma_pa_mainloop* m); -typedef ma_pa_mainloop_api* (* ma_pa_mainloop_get_api_proc) (ma_pa_mainloop* m); -typedef int (* ma_pa_mainloop_iterate_proc) (ma_pa_mainloop* m, int block, int* retval); -typedef void (* ma_pa_mainloop_wakeup_proc) (ma_pa_mainloop* m); -typedef ma_pa_context* (* ma_pa_context_new_proc) (ma_pa_mainloop_api* mainloop, const char* name); -typedef void (* ma_pa_context_unref_proc) (ma_pa_context* c); -typedef int (* ma_pa_context_connect_proc) (ma_pa_context* c, const char* server, ma_pa_context_flags_t flags, const ma_pa_spawn_api* api); -typedef void (* ma_pa_context_disconnect_proc) (ma_pa_context* c); -typedef void (* ma_pa_context_set_state_callback_proc) (ma_pa_context* c, ma_pa_context_notify_cb_t cb, void* userdata); -typedef ma_pa_context_state_t (* ma_pa_context_get_state_proc) (ma_pa_context* c); -typedef ma_pa_operation* (* ma_pa_context_get_sink_info_list_proc) (ma_pa_context* c, ma_pa_sink_info_cb_t cb, void* userdata); -typedef ma_pa_operation* (* ma_pa_context_get_source_info_list_proc) (ma_pa_context* c, ma_pa_source_info_cb_t cb, void* userdata); -typedef ma_pa_operation* (* ma_pa_context_get_sink_info_by_name_proc) (ma_pa_context* c, const char* name, ma_pa_sink_info_cb_t cb, void* userdata); -typedef ma_pa_operation* (* ma_pa_context_get_source_info_by_name_proc)(ma_pa_context* c, const char* name, ma_pa_source_info_cb_t cb, void* userdata); -typedef void (* ma_pa_operation_unref_proc) (ma_pa_operation* o); -typedef ma_pa_operation_state_t (* ma_pa_operation_get_state_proc) (ma_pa_operation* o); -typedef ma_pa_channel_map* (* ma_pa_channel_map_init_extend_proc) (ma_pa_channel_map* m, unsigned channels, ma_pa_channel_map_def_t def); -typedef int (* ma_pa_channel_map_valid_proc) (const ma_pa_channel_map* m); -typedef int (* ma_pa_channel_map_compatible_proc) (const ma_pa_channel_map* m, const ma_pa_sample_spec* ss); -typedef ma_pa_stream* (* ma_pa_stream_new_proc) (ma_pa_context* c, const char* name, const ma_pa_sample_spec* ss, const ma_pa_channel_map* map); -typedef void (* ma_pa_stream_unref_proc) (ma_pa_stream* s); -typedef int (* ma_pa_stream_connect_playback_proc) (ma_pa_stream* s, const char* dev, const ma_pa_buffer_attr* attr, ma_pa_stream_flags_t flags, const ma_pa_cvolume* volume, ma_pa_stream* sync_stream); -typedef int (* ma_pa_stream_connect_record_proc) (ma_pa_stream* s, const char* dev, const ma_pa_buffer_attr* attr, ma_pa_stream_flags_t flags); -typedef int (* ma_pa_stream_disconnect_proc) (ma_pa_stream* s); -typedef ma_pa_stream_state_t (* ma_pa_stream_get_state_proc) (ma_pa_stream* s); -typedef const ma_pa_sample_spec* (* ma_pa_stream_get_sample_spec_proc) (ma_pa_stream* s); -typedef const ma_pa_channel_map* (* ma_pa_stream_get_channel_map_proc) (ma_pa_stream* s); -typedef const ma_pa_buffer_attr* (* ma_pa_stream_get_buffer_attr_proc) (ma_pa_stream* s); -typedef ma_pa_operation* (* ma_pa_stream_set_buffer_attr_proc) (ma_pa_stream* s, const ma_pa_buffer_attr* attr, ma_pa_stream_success_cb_t cb, void* userdata); -typedef const char* (* ma_pa_stream_get_device_name_proc) (ma_pa_stream* s); -typedef void (* ma_pa_stream_set_write_callback_proc) (ma_pa_stream* s, ma_pa_stream_request_cb_t cb, void* userdata); -typedef void (* ma_pa_stream_set_read_callback_proc) (ma_pa_stream* s, ma_pa_stream_request_cb_t cb, void* userdata); -typedef ma_pa_operation* (* ma_pa_stream_flush_proc) (ma_pa_stream* s, ma_pa_stream_success_cb_t cb, void* userdata); -typedef ma_pa_operation* (* ma_pa_stream_drain_proc) (ma_pa_stream* s, ma_pa_stream_success_cb_t cb, void* userdata); -typedef int (* ma_pa_stream_is_corked_proc) (ma_pa_stream* s); -typedef ma_pa_operation* (* ma_pa_stream_cork_proc) (ma_pa_stream* s, int b, ma_pa_stream_success_cb_t cb, void* userdata); -typedef ma_pa_operation* (* ma_pa_stream_trigger_proc) (ma_pa_stream* s, ma_pa_stream_success_cb_t cb, void* userdata); -typedef int (* ma_pa_stream_begin_write_proc) (ma_pa_stream* s, void** data, size_t* nbytes); -typedef int (* ma_pa_stream_write_proc) (ma_pa_stream* s, const void* data, size_t nbytes, ma_pa_free_cb_t free_cb, int64_t offset, ma_pa_seek_mode_t seek); -typedef int (* ma_pa_stream_peek_proc) (ma_pa_stream* s, const void** data, size_t* nbytes); -typedef int (* ma_pa_stream_drop_proc) (ma_pa_stream* s); -typedef size_t (* ma_pa_stream_writable_size_proc) (ma_pa_stream* s); -typedef size_t (* ma_pa_stream_readable_size_proc) (ma_pa_stream* s); + framesWritten = ((ma_snd_pcm_writei_proc)pDevice->pContext->alsa.snd_pcm_writei)((ma_snd_pcm_t*)pDevice->alsa.pPCM, pDevice->alsa.pIntermediaryBuffer, framesAvailable); + if (framesWritten < 0) { + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to write data to the internal device.", MA_FAILED_TO_SEND_DATA_TO_DEVICE); + return MA_FALSE; + } -typedef struct -{ - ma_uint32 count; - ma_uint32 capacity; - ma_device_info* pInfo; -} ma_pulse_device_enum_data; + break; /* Success. */ + } else { + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_writei() failed when writing initial data.", MA_FAILED_TO_SEND_DATA_TO_DEVICE); + return MA_FALSE; + } + } else { + break; /* Success. */ + } + } + } -ma_result ma_result_from_pulse(int result) + return MA_TRUE; +} + +static ma_bool32 ma_device_read_and_send_to_client__alsa(ma_device* pDevice) { - switch (result) { - case MA_PA_OK: return MA_SUCCESS; - case MA_PA_ERR_ACCESS: return MA_ACCESS_DENIED; - case MA_PA_ERR_INVALID: return MA_INVALID_ARGS; - case MA_PA_ERR_NOENTITY: return MA_NO_DEVICE; - default: return MA_ERROR; + MA_ASSERT(pDevice != NULL); + if (!ma_device_is_started(pDevice)) { + return MA_FALSE; } -} + if (pDevice->alsa.breakFromMainLoop) { + return MA_FALSE; + } + + ma_uint32 framesToSend = 0; + void* pBuffer = NULL; + if (pDevice->alsa.pIntermediaryBuffer == NULL) { + /* mmap. */ + ma_bool32 requiresRestart; + ma_uint32 framesAvailable = ma_device__wait_for_frames__alsa(pDevice, &requiresRestart); + if (framesAvailable == 0) { + return MA_FALSE; + } + + const ma_snd_pcm_channel_area_t* pAreas; + ma_snd_pcm_uframes_t mappedOffset; + ma_snd_pcm_uframes_t mappedFrames = framesAvailable; + while (framesAvailable > 0) { + int result = ((ma_snd_pcm_mmap_begin_proc)pDevice->pContext->alsa.snd_pcm_mmap_begin)((ma_snd_pcm_t*)pDevice->alsa.pPCM, &pAreas, &mappedOffset, &mappedFrames); + if (result < 0) { + return MA_FALSE; + } + + if (mappedFrames > 0) { + void* pBuffer = (ma_uint8*)pAreas[0].addr + ((pAreas[0].first + (mappedOffset * pAreas[0].step)) / 8); + ma_device__send_frames_to_client(pDevice, mappedFrames, pBuffer); + } + + result = ((ma_snd_pcm_mmap_commit_proc)pDevice->pContext->alsa.snd_pcm_mmap_commit)((ma_snd_pcm_t*)pDevice->alsa.pPCM, mappedOffset, mappedFrames); + if (result < 0 || (ma_snd_pcm_uframes_t)result != mappedFrames) { + ((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCM, result, MA_TRUE); + return MA_FALSE; + } -#if 0 -ma_pa_sample_format_t ma_format_to_pulse(ma_format format) -{ - if (ma_is_little_endian()) { - switch (format) { - case ma_format_s16: return MA_PA_SAMPLE_S16LE; - case ma_format_s24: return MA_PA_SAMPLE_S24LE; - case ma_format_s32: return MA_PA_SAMPLE_S32LE; - case ma_format_f32: return MA_PA_SAMPLE_FLOAT32LE; - default: break; + if (requiresRestart) { + if (((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCM) < 0) { + return MA_FALSE; + } + } + + if (framesAvailable >= mappedFrames) { + framesAvailable -= mappedFrames; + } else { + framesAvailable = 0; + } } } else { - switch (format) { - case ma_format_s16: return MA_PA_SAMPLE_S16BE; - case ma_format_s24: return MA_PA_SAMPLE_S24BE; - case ma_format_s32: return MA_PA_SAMPLE_S32BE; - case ma_format_f32: return MA_PA_SAMPLE_FLOAT32BE; - default: break; - } - } + /* readi/writei. */ + ma_snd_pcm_sframes_t framesRead = 0; + while (!pDevice->alsa.breakFromMainLoop) { + ma_uint32 framesAvailable = ma_device__wait_for_frames__alsa(pDevice, NULL); + if (framesAvailable == 0) { + continue; + } - /* Endian agnostic. */ - switch (format) { - case ma_format_u8: return MA_PA_SAMPLE_U8; - default: return MA_PA_SAMPLE_INVALID; - } -} -#endif + framesRead = ((ma_snd_pcm_readi_proc)pDevice->pContext->alsa.snd_pcm_readi)((ma_snd_pcm_t*)pDevice->alsa.pPCM, pDevice->alsa.pIntermediaryBuffer, framesAvailable); + if (framesRead < 0) { + if (framesRead == -EAGAIN) { + continue; /* Just keep trying... */ + } else if (framesRead == -EPIPE) { + /* Overrun. Just recover and try reading again. */ + if (((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCM, framesRead, MA_TRUE) < 0) { + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to recover device after overrun.", MA_FAILED_TO_START_BACKEND_DEVICE); + return MA_FALSE; + } -ma_format ma_format_from_pulse(ma_pa_sample_format_t format) -{ - if (ma_is_little_endian()) { - switch (format) { - case MA_PA_SAMPLE_S16LE: return ma_format_s16; - case MA_PA_SAMPLE_S24LE: return ma_format_s24; - case MA_PA_SAMPLE_S32LE: return ma_format_s32; - case MA_PA_SAMPLE_FLOAT32LE: return ma_format_f32; - default: break; - } - } else { - switch (format) { - case MA_PA_SAMPLE_S16BE: return ma_format_s16; - case MA_PA_SAMPLE_S24BE: return ma_format_s24; - case MA_PA_SAMPLE_S32BE: return ma_format_s32; - case MA_PA_SAMPLE_FLOAT32BE: return ma_format_f32; - default: break; + framesRead = ((ma_snd_pcm_readi_proc)pDevice->pContext->alsa.snd_pcm_readi)((ma_snd_pcm_t*)pDevice->alsa.pPCM, pDevice->alsa.pIntermediaryBuffer, framesAvailable); + if (framesRead < 0) { + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to read data from the internal device.", MA_FAILED_TO_READ_DATA_FROM_DEVICE); + return MA_FALSE; + } + + break; /* Success. */ + } else { + return MA_FALSE; + } + } else { + break; /* Success. */ + } } + + framesToSend = framesRead; + pBuffer = pDevice->alsa.pIntermediaryBuffer; } - /* Endian agnostic. */ - switch (format) { - case MA_PA_SAMPLE_U8: return ma_format_u8; - default: return ma_format_unknown; + if (framesToSend > 0) { + ma_device__send_frames_to_client(pDevice, framesToSend, pBuffer); } + + return MA_TRUE; } +#endif /* 0 */ -ma_channel ma_channel_position_from_pulse(ma_pa_channel_position_t position) +static void ma_device_uninit__alsa(ma_device* pDevice) { - switch (position) - { - case MA_PA_CHANNEL_POSITION_INVALID: return MA_CHANNEL_NONE; - case MA_PA_CHANNEL_POSITION_MONO: return MA_CHANNEL_MONO; - case MA_PA_CHANNEL_POSITION_FRONT_LEFT: return MA_CHANNEL_FRONT_LEFT; - case MA_PA_CHANNEL_POSITION_FRONT_RIGHT: return MA_CHANNEL_FRONT_RIGHT; - case MA_PA_CHANNEL_POSITION_FRONT_CENTER: return MA_CHANNEL_FRONT_CENTER; - case MA_PA_CHANNEL_POSITION_REAR_CENTER: return MA_CHANNEL_BACK_CENTER; - case MA_PA_CHANNEL_POSITION_REAR_LEFT: return MA_CHANNEL_BACK_LEFT; - case MA_PA_CHANNEL_POSITION_REAR_RIGHT: return MA_CHANNEL_BACK_RIGHT; - case MA_PA_CHANNEL_POSITION_LFE: return MA_CHANNEL_LFE; - case MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER: return MA_CHANNEL_FRONT_LEFT_CENTER; - case MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER: return MA_CHANNEL_FRONT_RIGHT_CENTER; - case MA_PA_CHANNEL_POSITION_SIDE_LEFT: return MA_CHANNEL_SIDE_LEFT; - case MA_PA_CHANNEL_POSITION_SIDE_RIGHT: return MA_CHANNEL_SIDE_RIGHT; - case MA_PA_CHANNEL_POSITION_AUX0: return MA_CHANNEL_AUX_0; - case MA_PA_CHANNEL_POSITION_AUX1: return MA_CHANNEL_AUX_1; - case MA_PA_CHANNEL_POSITION_AUX2: return MA_CHANNEL_AUX_2; - case MA_PA_CHANNEL_POSITION_AUX3: return MA_CHANNEL_AUX_3; - case MA_PA_CHANNEL_POSITION_AUX4: return MA_CHANNEL_AUX_4; - case MA_PA_CHANNEL_POSITION_AUX5: return MA_CHANNEL_AUX_5; - case MA_PA_CHANNEL_POSITION_AUX6: return MA_CHANNEL_AUX_6; - case MA_PA_CHANNEL_POSITION_AUX7: return MA_CHANNEL_AUX_7; - case MA_PA_CHANNEL_POSITION_AUX8: return MA_CHANNEL_AUX_8; - case MA_PA_CHANNEL_POSITION_AUX9: return MA_CHANNEL_AUX_9; - case MA_PA_CHANNEL_POSITION_AUX10: return MA_CHANNEL_AUX_10; - case MA_PA_CHANNEL_POSITION_AUX11: return MA_CHANNEL_AUX_11; - case MA_PA_CHANNEL_POSITION_AUX12: return MA_CHANNEL_AUX_12; - case MA_PA_CHANNEL_POSITION_AUX13: return MA_CHANNEL_AUX_13; - case MA_PA_CHANNEL_POSITION_AUX14: return MA_CHANNEL_AUX_14; - case MA_PA_CHANNEL_POSITION_AUX15: return MA_CHANNEL_AUX_15; - case MA_PA_CHANNEL_POSITION_AUX16: return MA_CHANNEL_AUX_16; - case MA_PA_CHANNEL_POSITION_AUX17: return MA_CHANNEL_AUX_17; - case MA_PA_CHANNEL_POSITION_AUX18: return MA_CHANNEL_AUX_18; - case MA_PA_CHANNEL_POSITION_AUX19: return MA_CHANNEL_AUX_19; - case MA_PA_CHANNEL_POSITION_AUX20: return MA_CHANNEL_AUX_20; - case MA_PA_CHANNEL_POSITION_AUX21: return MA_CHANNEL_AUX_21; - case MA_PA_CHANNEL_POSITION_AUX22: return MA_CHANNEL_AUX_22; - case MA_PA_CHANNEL_POSITION_AUX23: return MA_CHANNEL_AUX_23; - case MA_PA_CHANNEL_POSITION_AUX24: return MA_CHANNEL_AUX_24; - case MA_PA_CHANNEL_POSITION_AUX25: return MA_CHANNEL_AUX_25; - case MA_PA_CHANNEL_POSITION_AUX26: return MA_CHANNEL_AUX_26; - case MA_PA_CHANNEL_POSITION_AUX27: return MA_CHANNEL_AUX_27; - case MA_PA_CHANNEL_POSITION_AUX28: return MA_CHANNEL_AUX_28; - case MA_PA_CHANNEL_POSITION_AUX29: return MA_CHANNEL_AUX_29; - case MA_PA_CHANNEL_POSITION_AUX30: return MA_CHANNEL_AUX_30; - case MA_PA_CHANNEL_POSITION_AUX31: return MA_CHANNEL_AUX_31; - case MA_PA_CHANNEL_POSITION_TOP_CENTER: return MA_CHANNEL_TOP_CENTER; - case MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT: return MA_CHANNEL_TOP_FRONT_LEFT; - case MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT: return MA_CHANNEL_TOP_FRONT_RIGHT; - case MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER: return MA_CHANNEL_TOP_FRONT_CENTER; - case MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT: return MA_CHANNEL_TOP_BACK_LEFT; - case MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT: return MA_CHANNEL_TOP_BACK_RIGHT; - case MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER: return MA_CHANNEL_TOP_BACK_CENTER; - default: return MA_CHANNEL_NONE; + MA_ASSERT(pDevice != NULL); + + if ((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture) { + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture); } -} -#if 0 -ma_pa_channel_position_t ma_channel_position_to_pulse(ma_channel position) -{ - switch (position) - { - case MA_CHANNEL_NONE: return MA_PA_CHANNEL_POSITION_INVALID; - case MA_CHANNEL_FRONT_LEFT: return MA_PA_CHANNEL_POSITION_FRONT_LEFT; - case MA_CHANNEL_FRONT_RIGHT: return MA_PA_CHANNEL_POSITION_FRONT_RIGHT; - case MA_CHANNEL_FRONT_CENTER: return MA_PA_CHANNEL_POSITION_FRONT_CENTER; - case MA_CHANNEL_LFE: return MA_PA_CHANNEL_POSITION_LFE; - case MA_CHANNEL_BACK_LEFT: return MA_PA_CHANNEL_POSITION_REAR_LEFT; - case MA_CHANNEL_BACK_RIGHT: return MA_PA_CHANNEL_POSITION_REAR_RIGHT; - case MA_CHANNEL_FRONT_LEFT_CENTER: return MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER; - case MA_CHANNEL_FRONT_RIGHT_CENTER: return MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER; - case MA_CHANNEL_BACK_CENTER: return MA_PA_CHANNEL_POSITION_REAR_CENTER; - case MA_CHANNEL_SIDE_LEFT: return MA_PA_CHANNEL_POSITION_SIDE_LEFT; - case MA_CHANNEL_SIDE_RIGHT: return MA_PA_CHANNEL_POSITION_SIDE_RIGHT; - case MA_CHANNEL_TOP_CENTER: return MA_PA_CHANNEL_POSITION_TOP_CENTER; - case MA_CHANNEL_TOP_FRONT_LEFT: return MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT; - case MA_CHANNEL_TOP_FRONT_CENTER: return MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER; - case MA_CHANNEL_TOP_FRONT_RIGHT: return MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT; - case MA_CHANNEL_TOP_BACK_LEFT: return MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT; - case MA_CHANNEL_TOP_BACK_CENTER: return MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER; - case MA_CHANNEL_TOP_BACK_RIGHT: return MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT; - case MA_CHANNEL_19: return MA_PA_CHANNEL_POSITION_AUX18; - case MA_CHANNEL_20: return MA_PA_CHANNEL_POSITION_AUX19; - case MA_CHANNEL_21: return MA_PA_CHANNEL_POSITION_AUX20; - case MA_CHANNEL_22: return MA_PA_CHANNEL_POSITION_AUX21; - case MA_CHANNEL_23: return MA_PA_CHANNEL_POSITION_AUX22; - case MA_CHANNEL_24: return MA_PA_CHANNEL_POSITION_AUX23; - case MA_CHANNEL_25: return MA_PA_CHANNEL_POSITION_AUX24; - case MA_CHANNEL_26: return MA_PA_CHANNEL_POSITION_AUX25; - case MA_CHANNEL_27: return MA_PA_CHANNEL_POSITION_AUX26; - case MA_CHANNEL_28: return MA_PA_CHANNEL_POSITION_AUX27; - case MA_CHANNEL_29: return MA_PA_CHANNEL_POSITION_AUX28; - case MA_CHANNEL_30: return MA_PA_CHANNEL_POSITION_AUX29; - case MA_CHANNEL_31: return MA_PA_CHANNEL_POSITION_AUX30; - case MA_CHANNEL_32: return MA_PA_CHANNEL_POSITION_AUX31; - default: return (ma_pa_channel_position_t)position; + if ((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback) { + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback); } } -#endif -ma_result ma_wait_for_operation__pulse(ma_context* pContext, ma_pa_mainloop* pMainLoop, ma_pa_operation* pOP) +static ma_result ma_device_init_by_type__alsa(ma_context* pContext, const ma_device_config* pConfig, ma_device_type deviceType, ma_device* pDevice) { - ma_assert(pContext != NULL); - ma_assert(pMainLoop != NULL); - ma_assert(pOP != NULL); + ma_result result; + ma_snd_pcm_t* pPCM; + ma_bool32 isUsingMMap; + ma_snd_pcm_format_t formatALSA; + ma_share_mode shareMode; + ma_device_id* pDeviceID; + ma_format internalFormat; + ma_uint32 internalChannels; + ma_uint32 internalSampleRate; + ma_channel internalChannelMap[MA_MAX_CHANNELS]; + ma_uint32 internalPeriodSizeInFrames; + ma_uint32 internalPeriods; + ma_snd_pcm_hw_params_t* pHWParams; + ma_snd_pcm_sw_params_t* pSWParams; + ma_snd_pcm_uframes_t bufferBoundary; + float bufferSizeScaleFactor; - while (((ma_pa_operation_get_state_proc)pContext->pulse.pa_operation_get_state)(pOP) == MA_PA_OPERATION_RUNNING) { - int error = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)(pMainLoop, 1, NULL); - if (error < 0) { - return ma_result_from_pulse(error); - } + MA_ASSERT(pContext != NULL); + MA_ASSERT(pConfig != NULL); + MA_ASSERT(deviceType != ma_device_type_duplex); /* This function should only be called for playback _or_ capture, never duplex. */ + MA_ASSERT(pDevice != NULL); + + formatALSA = ma_convert_ma_format_to_alsa_format((deviceType == ma_device_type_capture) ? pConfig->capture.format : pConfig->playback.format); + shareMode = (deviceType == ma_device_type_capture) ? pConfig->capture.shareMode : pConfig->playback.shareMode; + pDeviceID = (deviceType == ma_device_type_capture) ? pConfig->capture.pDeviceID : pConfig->playback.pDeviceID; + + result = ma_context_open_pcm__alsa(pContext, shareMode, deviceType, pDeviceID, &pPCM); + if (result != MA_SUCCESS) { + return result; } - return MA_SUCCESS; -} + /* If using the default buffer size we may want to apply some device-specific scaling for known devices that have peculiar latency characteristics */ + bufferSizeScaleFactor = 1; + if (pDevice->usingDefaultBufferSize) { + ma_snd_pcm_info_t* pInfo = (ma_snd_pcm_info_t*)ma__calloc_from_callbacks(((ma_snd_pcm_info_sizeof_proc)pContext->alsa.snd_pcm_info_sizeof)(), &pContext->allocationCallbacks); + if (pInfo == NULL) { + return MA_OUT_OF_MEMORY; + } -ma_result ma_device__wait_for_operation__pulse(ma_device* pDevice, ma_pa_operation* pOP) -{ - ma_assert(pDevice != NULL); - ma_assert(pOP != NULL); + /* We may need to scale the size of the buffer depending on the device. */ + if (((ma_snd_pcm_info_proc)pContext->alsa.snd_pcm_info)(pPCM, pInfo) == 0) { + const char* deviceName = ((ma_snd_pcm_info_get_name_proc)pContext->alsa.snd_pcm_info_get_name)(pInfo); + if (deviceName != NULL) { + if (ma_strcmp(deviceName, "default") == 0) { + char** ppDeviceHints; + char** ppNextDeviceHint; - return ma_wait_for_operation__pulse(pDevice->pContext, (ma_pa_mainloop*)pDevice->pulse.pMainLoop, pOP); -} + /* It's the default device. We need to use DESC from snd_device_name_hint(). */ + if (((ma_snd_device_name_hint_proc)pContext->alsa.snd_device_name_hint)(-1, "pcm", (void***)&ppDeviceHints) < 0) { + ma__free_from_callbacks(pInfo, &pContext->allocationCallbacks); + return MA_NO_BACKEND; + } + ppNextDeviceHint = ppDeviceHints; + while (*ppNextDeviceHint != NULL) { + char* NAME = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "NAME"); + char* DESC = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "DESC"); + char* IOID = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "IOID"); -ma_bool32 ma_context_is_device_id_equal__pulse(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) -{ - ma_assert(pContext != NULL); - ma_assert(pID0 != NULL); - ma_assert(pID1 != NULL); - (void)pContext; + ma_bool32 foundDevice = MA_FALSE; + if ((deviceType == ma_device_type_playback && (IOID == NULL || ma_strcmp(IOID, "Output") == 0)) || + (deviceType == ma_device_type_capture && (IOID != NULL && ma_strcmp(IOID, "Input" ) == 0))) { + if (ma_strcmp(NAME, deviceName) == 0) { + bufferSizeScaleFactor = ma_find_default_buffer_size_scale__alsa(DESC); + foundDevice = MA_TRUE; + } + } - return ma_strcmp(pID0->pulse, pID1->pulse) == 0; -} + free(NAME); + free(DESC); + free(IOID); + ppNextDeviceHint += 1; + if (foundDevice) { + break; + } + } -typedef struct -{ - ma_context* pContext; - ma_enum_devices_callback_proc callback; - void* pUserData; - ma_bool32 isTerminated; -} ma_context_enumerate_devices_callback_data__pulse; + ((ma_snd_device_name_free_hint_proc)pContext->alsa.snd_device_name_free_hint)((void**)ppDeviceHints); + } else { + bufferSizeScaleFactor = ma_find_default_buffer_size_scale__alsa(deviceName); + } + } + } -void ma_context_enumerate_devices_sink_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_sink_info* pSinkInfo, int endOfList, void* pUserData) -{ - ma_context_enumerate_devices_callback_data__pulse* pData = (ma_context_enumerate_devices_callback_data__pulse*)pUserData; - ma_device_info deviceInfo; + ma__free_from_callbacks(pInfo, &pContext->allocationCallbacks); + } - ma_assert(pData != NULL); - if (endOfList || pData->isTerminated) { - return; + /* Hardware parameters. */ + pHWParams = (ma_snd_pcm_hw_params_t*)ma__calloc_from_callbacks(((ma_snd_pcm_hw_params_sizeof_proc)pContext->alsa.snd_pcm_hw_params_sizeof)(), &pContext->allocationCallbacks); + if (pHWParams == NULL) { + return MA_OUT_OF_MEMORY; } - ma_zero_object(&deviceInfo); + if (((ma_snd_pcm_hw_params_any_proc)pContext->alsa.snd_pcm_hw_params_any)(pPCM, pHWParams) < 0) { + ma__free_from_callbacks(pHWParams, &pContext->allocationCallbacks); + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to initialize hardware parameters. snd_pcm_hw_params_any() failed.", MA_FAILED_TO_CONFIGURE_BACKEND_DEVICE); + } - /* The name from PulseAudio is the ID for miniaudio. */ - if (pSinkInfo->name != NULL) { - ma_strncpy_s(deviceInfo.id.pulse, sizeof(deviceInfo.id.pulse), pSinkInfo->name, (size_t)-1); + /* MMAP Mode. Try using interleaved MMAP access. If this fails, fall back to standard readi/writei. */ + isUsingMMap = MA_FALSE; +#if 0 /* NOTE: MMAP mode temporarily disabled. */ + if (deviceType != ma_device_type_capture) { /* <-- Disabling MMAP mode for capture devices because I apparently do not have a device that supports it which means I can't test it... Contributions welcome. */ + if (!pConfig->alsa.noMMap && ma_device__is_async(pDevice)) { + if (((ma_snd_pcm_hw_params_set_access_proc)pContext->alsa.snd_pcm_hw_params_set_access)(pPCM, pHWParams, MA_SND_PCM_ACCESS_MMAP_INTERLEAVED) == 0) { + pDevice->alsa.isUsingMMap = MA_TRUE; + } + } } +#endif - /* The description from PulseAudio is the name for miniaudio. */ - if (pSinkInfo->description != NULL) { - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), pSinkInfo->description, (size_t)-1); + if (!isUsingMMap) { + if (((ma_snd_pcm_hw_params_set_access_proc)pContext->alsa.snd_pcm_hw_params_set_access)(pPCM, pHWParams, MA_SND_PCM_ACCESS_RW_INTERLEAVED) < 0) { + ma__free_from_callbacks(pHWParams, &pContext->allocationCallbacks); + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set access mode to neither SND_PCM_ACCESS_MMAP_INTERLEAVED nor SND_PCM_ACCESS_RW_INTERLEAVED. snd_pcm_hw_params_set_access() failed.", MA_FORMAT_NOT_SUPPORTED); + } } - pData->isTerminated = !pData->callback(pData->pContext, ma_device_type_playback, &deviceInfo, pData->pUserData); + /* + Most important properties first. The documentation for OSS (yes, I know this is ALSA!) recommends format, channels, then sample rate. I can't + find any documentation for ALSA specifically, so I'm going to copy the recommendation for OSS. + */ - (void)pPulseContext; /* Unused. */ -} + /* Format. */ + { + ma_snd_pcm_format_mask_t* pFormatMask; -void ma_context_enumerate_devices_source_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_source_info* pSinkInfo, int endOfList, void* pUserData) -{ - ma_context_enumerate_devices_callback_data__pulse* pData = (ma_context_enumerate_devices_callback_data__pulse*)pUserData; - ma_device_info deviceInfo; + /* Try getting every supported format first. */ + pFormatMask = (ma_snd_pcm_format_mask_t*)ma__calloc_from_callbacks(((ma_snd_pcm_format_mask_sizeof_proc)pContext->alsa.snd_pcm_format_mask_sizeof)(), &pContext->allocationCallbacks); + if (pFormatMask == NULL) { + ma__free_from_callbacks(pHWParams, &pContext->allocationCallbacks); + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); + return MA_OUT_OF_MEMORY; + } - ma_assert(pData != NULL); + ((ma_snd_pcm_hw_params_get_format_mask_proc)pContext->alsa.snd_pcm_hw_params_get_format_mask)(pHWParams, pFormatMask); - if (endOfList || pData->isTerminated) { - return; - } + /* + At this point we should have a list of supported formats, so now we need to find the best one. We first check if the requested format is + supported, and if so, use that one. If it's not supported, we just run though a list of formats and try to find the best one. + */ + if (!((ma_snd_pcm_format_mask_test_proc)pContext->alsa.snd_pcm_format_mask_test)(pFormatMask, formatALSA)) { + size_t i; - ma_zero_object(&deviceInfo); + /* The requested format is not supported so now try running through the list of formats and return the best one. */ + ma_snd_pcm_format_t preferredFormatsALSA[] = { + MA_SND_PCM_FORMAT_S16_LE, /* ma_format_s16 */ + MA_SND_PCM_FORMAT_FLOAT_LE, /* ma_format_f32 */ + MA_SND_PCM_FORMAT_S32_LE, /* ma_format_s32 */ + MA_SND_PCM_FORMAT_S24_3LE, /* ma_format_s24 */ + MA_SND_PCM_FORMAT_U8 /* ma_format_u8 */ + }; - /* The name from PulseAudio is the ID for miniaudio. */ - if (pSinkInfo->name != NULL) { - ma_strncpy_s(deviceInfo.id.pulse, sizeof(deviceInfo.id.pulse), pSinkInfo->name, (size_t)-1); - } + if (ma_is_big_endian()) { + preferredFormatsALSA[0] = MA_SND_PCM_FORMAT_S16_BE; + preferredFormatsALSA[1] = MA_SND_PCM_FORMAT_FLOAT_BE; + preferredFormatsALSA[2] = MA_SND_PCM_FORMAT_S32_BE; + preferredFormatsALSA[3] = MA_SND_PCM_FORMAT_S24_3BE; + preferredFormatsALSA[4] = MA_SND_PCM_FORMAT_U8; + } - /* The description from PulseAudio is the name for miniaudio. */ - if (pSinkInfo->description != NULL) { - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), pSinkInfo->description, (size_t)-1); - } + formatALSA = MA_SND_PCM_FORMAT_UNKNOWN; + for (i = 0; i < (sizeof(preferredFormatsALSA) / sizeof(preferredFormatsALSA[0])); ++i) { + if (((ma_snd_pcm_format_mask_test_proc)pContext->alsa.snd_pcm_format_mask_test)(pFormatMask, preferredFormatsALSA[i])) { + formatALSA = preferredFormatsALSA[i]; + break; + } + } - pData->isTerminated = !pData->callback(pData->pContext, ma_device_type_capture, &deviceInfo, pData->pUserData); + if (formatALSA == MA_SND_PCM_FORMAT_UNKNOWN) { + ma__free_from_callbacks(pHWParams, &pContext->allocationCallbacks); + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Format not supported. The device does not support any miniaudio formats.", MA_FORMAT_NOT_SUPPORTED); + } + } - (void)pPulseContext; /* Unused. */ -} + ma__free_from_callbacks(pFormatMask, &pContext->allocationCallbacks); + pFormatMask = NULL; -ma_result ma_context_enumerate_devices__pulse(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) -{ - ma_result result = MA_SUCCESS; - ma_context_enumerate_devices_callback_data__pulse callbackData; - ma_pa_operation* pOP = NULL; - ma_pa_mainloop* pMainLoop; - ma_pa_mainloop_api* pAPI; - ma_pa_context* pPulseContext; - int error; + if (((ma_snd_pcm_hw_params_set_format_proc)pContext->alsa.snd_pcm_hw_params_set_format)(pPCM, pHWParams, formatALSA) < 0) { + ma__free_from_callbacks(pHWParams, &pContext->allocationCallbacks); + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Format not supported. snd_pcm_hw_params_set_format() failed.", MA_FORMAT_NOT_SUPPORTED); + } + + internalFormat = ma_format_from_alsa(formatALSA); + if (internalFormat == ma_format_unknown) { + ma__free_from_callbacks(pHWParams, &pContext->allocationCallbacks); + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] The chosen format is not supported by miniaudio.", MA_FORMAT_NOT_SUPPORTED); + } + } - ma_assert(pContext != NULL); - ma_assert(callback != NULL); + /* Channels. */ + { + unsigned int channels = (deviceType == ma_device_type_capture) ? pConfig->capture.channels : pConfig->playback.channels; + if (((ma_snd_pcm_hw_params_set_channels_near_proc)pContext->alsa.snd_pcm_hw_params_set_channels_near)(pPCM, pHWParams, &channels) < 0) { + ma__free_from_callbacks(pHWParams, &pContext->allocationCallbacks); + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set channel count. snd_pcm_hw_params_set_channels_near() failed.", MA_FORMAT_NOT_SUPPORTED); + } + internalChannels = (ma_uint32)channels; + } - callbackData.pContext = pContext; - callbackData.callback = callback; - callbackData.pUserData = pUserData; - callbackData.isTerminated = MA_FALSE; + /* Sample Rate */ + { + unsigned int sampleRate; - pMainLoop = ((ma_pa_mainloop_new_proc)pContext->pulse.pa_mainloop_new)(); - if (pMainLoop == NULL) { - return MA_FAILED_TO_INIT_BACKEND; + /* + It appears there's either a bug in ALSA, a bug in some drivers, or I'm doing something silly; but having resampling enabled causes + problems with some device configurations when used in conjunction with MMAP access mode. To fix this problem we need to disable + resampling. + + To reproduce this problem, open the "plug:dmix" device, and set the sample rate to 44100. Internally, it looks like dmix uses a + sample rate of 48000. The hardware parameters will get set correctly with no errors, but it looks like the 44100 -> 48000 resampling + doesn't work properly - but only with MMAP access mode. You will notice skipping/crackling in the audio, and it'll run at a slightly + faster rate. + + miniaudio has built-in support for sample rate conversion (albeit low quality at the moment), so disabling resampling should be fine + for us. The only problem is that it won't be taking advantage of any kind of hardware-accelerated resampling and it won't be very + good quality until I get a chance to improve the quality of miniaudio's software sample rate conversion. + + I don't currently know if the dmix plugin is the only one with this error. Indeed, this is the only one I've been able to reproduce + this error with. In the future, we may want to restrict the disabling of resampling to only known bad plugins. + */ + ((ma_snd_pcm_hw_params_set_rate_resample_proc)pContext->alsa.snd_pcm_hw_params_set_rate_resample)(pPCM, pHWParams, 0); + + sampleRate = pConfig->sampleRate; + if (((ma_snd_pcm_hw_params_set_rate_near_proc)pContext->alsa.snd_pcm_hw_params_set_rate_near)(pPCM, pHWParams, &sampleRate, 0) < 0) { + ma__free_from_callbacks(pHWParams, &pContext->allocationCallbacks); + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Sample rate not supported. snd_pcm_hw_params_set_rate_near() failed.", MA_FORMAT_NOT_SUPPORTED); + } + internalSampleRate = (ma_uint32)sampleRate; } - pAPI = ((ma_pa_mainloop_get_api_proc)pContext->pulse.pa_mainloop_get_api)(pMainLoop); - if (pAPI == NULL) { - ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); - return MA_FAILED_TO_INIT_BACKEND; + /* Periods. */ + { + ma_uint32 periods = pConfig->periods; + if (((ma_snd_pcm_hw_params_set_periods_near_proc)pContext->alsa.snd_pcm_hw_params_set_periods_near)(pPCM, pHWParams, &periods, NULL) < 0) { + ma__free_from_callbacks(pHWParams, &pContext->allocationCallbacks); + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set period count. snd_pcm_hw_params_set_periods_near() failed.", MA_FORMAT_NOT_SUPPORTED); + } + internalPeriods = periods; } - pPulseContext = ((ma_pa_context_new_proc)pContext->pulse.pa_context_new)(pAPI, pContext->pulse.pApplicationName); - if (pPulseContext == NULL) { - ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); - return MA_FAILED_TO_INIT_BACKEND; + /* Buffer Size */ + { + ma_snd_pcm_uframes_t actualBufferSizeInFrames = pConfig->periodSizeInFrames * internalPeriods; + if (actualBufferSizeInFrames == 0) { + actualBufferSizeInFrames = ma_scale_buffer_size(ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->periodSizeInMilliseconds, internalSampleRate), bufferSizeScaleFactor) * internalPeriods; + } + + if (((ma_snd_pcm_hw_params_set_buffer_size_near_proc)pContext->alsa.snd_pcm_hw_params_set_buffer_size_near)(pPCM, pHWParams, &actualBufferSizeInFrames) < 0) { + ma__free_from_callbacks(pHWParams, &pContext->allocationCallbacks); + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set buffer size for device. snd_pcm_hw_params_set_buffer_size() failed.", MA_FORMAT_NOT_SUPPORTED); + } + internalPeriodSizeInFrames = actualBufferSizeInFrames / internalPeriods; } - error = ((ma_pa_context_connect_proc)pContext->pulse.pa_context_connect)(pPulseContext, pContext->pulse.pServerName, (pContext->pulse.tryAutoSpawn) ? 0 : MA_PA_CONTEXT_NOAUTOSPAWN, NULL); - if (error != MA_PA_OK) { - ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)(pPulseContext); - ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); - return ma_result_from_pulse(error); + /* Apply hardware parameters. */ + if (((ma_snd_pcm_hw_params_proc)pContext->alsa.snd_pcm_hw_params)(pPCM, pHWParams) < 0) { + ma__free_from_callbacks(pHWParams, &pContext->allocationCallbacks); + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set hardware parameters. snd_pcm_hw_params() failed.", MA_FAILED_TO_CONFIGURE_BACKEND_DEVICE); } - for (;;) { - ma_pa_context_state_t state = ((ma_pa_context_get_state_proc)pContext->pulse.pa_context_get_state)(pPulseContext); - if (state == MA_PA_CONTEXT_READY) { - break; /* Success. */ - } - if (state == MA_PA_CONTEXT_CONNECTING || state == MA_PA_CONTEXT_AUTHORIZING || state == MA_PA_CONTEXT_SETTING_NAME) { - error = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)(pMainLoop, 1, NULL); - if (error < 0) { - result = ma_result_from_pulse(error); - goto done; - } + ma__free_from_callbacks(pHWParams, &pContext->allocationCallbacks); + pHWParams = NULL; -#ifdef MA_DEBUG_OUTPUT - printf("[PulseAudio] pa_context_get_state() returned %d. Waiting.\n", state); -#endif - continue; /* Keep trying. */ - } - if (state == MA_PA_CONTEXT_UNCONNECTED || state == MA_PA_CONTEXT_FAILED || state == MA_PA_CONTEXT_TERMINATED) { -#ifdef MA_DEBUG_OUTPUT - printf("[PulseAudio] pa_context_get_state() returned %d. Failed.\n", state); -#endif - goto done; /* Failed. */ - } + + /* Software parameters. */ + pSWParams = (ma_snd_pcm_sw_params_t*)ma__calloc_from_callbacks(((ma_snd_pcm_sw_params_sizeof_proc)pContext->alsa.snd_pcm_sw_params_sizeof)(), &pContext->allocationCallbacks); + if (pSWParams == NULL) { + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); + return MA_OUT_OF_MEMORY; } + if (((ma_snd_pcm_sw_params_current_proc)pContext->alsa.snd_pcm_sw_params_current)(pPCM, pSWParams) != 0) { + ma__free_from_callbacks(pSWParams, &pContext->allocationCallbacks); + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to initialize software parameters. snd_pcm_sw_params_current() failed.", MA_FAILED_TO_CONFIGURE_BACKEND_DEVICE); + } - /* Playback. */ - if (!callbackData.isTerminated) { - pOP = ((ma_pa_context_get_sink_info_list_proc)pContext->pulse.pa_context_get_sink_info_list)(pPulseContext, ma_context_enumerate_devices_sink_callback__pulse, &callbackData); - if (pOP == NULL) { - result = MA_ERROR; - goto done; - } + if (((ma_snd_pcm_sw_params_set_avail_min_proc)pContext->alsa.snd_pcm_sw_params_set_avail_min)(pPCM, pSWParams, ma_prev_power_of_2(internalPeriodSizeInFrames)) != 0) { + ma__free_from_callbacks(pSWParams, &pContext->allocationCallbacks); + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_sw_params_set_avail_min() failed.", MA_FORMAT_NOT_SUPPORTED); + } - result = ma_wait_for_operation__pulse(pContext, pMainLoop, pOP); - ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); - if (result != MA_SUCCESS) { - goto done; - } + if (((ma_snd_pcm_sw_params_get_boundary_proc)pContext->alsa.snd_pcm_sw_params_get_boundary)(pSWParams, &bufferBoundary) < 0) { + bufferBoundary = internalPeriodSizeInFrames * internalPeriods; } + /*printf("TRACE: bufferBoundary=%ld\n", bufferBoundary);*/ - /* Capture. */ - if (!callbackData.isTerminated) { - pOP = ((ma_pa_context_get_source_info_list_proc)pContext->pulse.pa_context_get_source_info_list)(pPulseContext, ma_context_enumerate_devices_source_callback__pulse, &callbackData); - if (pOP == NULL) { - result = MA_ERROR; - goto done; + if (deviceType == ma_device_type_playback && !isUsingMMap) { /* Only playback devices in writei/readi mode need a start threshold. */ + /* + Subtle detail here with the start threshold. When in playback-only mode (no full-duplex) we can set the start threshold to + the size of a period. But for full-duplex we need to set it such that it is at least two periods. + */ + if (((ma_snd_pcm_sw_params_set_start_threshold_proc)pContext->alsa.snd_pcm_sw_params_set_start_threshold)(pPCM, pSWParams, internalPeriodSizeInFrames*2) != 0) { + ma__free_from_callbacks(pSWParams, &pContext->allocationCallbacks); + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set start threshold for playback device. snd_pcm_sw_params_set_start_threshold() failed.", MA_FAILED_TO_CONFIGURE_BACKEND_DEVICE); } - - result = ma_wait_for_operation__pulse(pContext, pMainLoop, pOP); - ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); - if (result != MA_SUCCESS) { - goto done; + if (((ma_snd_pcm_sw_params_set_stop_threshold_proc)pContext->alsa.snd_pcm_sw_params_set_stop_threshold)(pPCM, pSWParams, bufferBoundary) != 0) { /* Set to boundary to loop instead of stop in the event of an xrun. */ + ma__free_from_callbacks(pSWParams, &pContext->allocationCallbacks); + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set stop threshold for playback device. snd_pcm_sw_params_set_stop_threshold() failed.", MA_FAILED_TO_CONFIGURE_BACKEND_DEVICE); } } -done: - ((ma_pa_context_disconnect_proc)pContext->pulse.pa_context_disconnect)(pPulseContext); - ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)(pPulseContext); - ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); - return result; -} - + if (((ma_snd_pcm_sw_params_proc)pContext->alsa.snd_pcm_sw_params)(pPCM, pSWParams) != 0) { + ma__free_from_callbacks(pSWParams, &pContext->allocationCallbacks); + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set software parameters. snd_pcm_sw_params() failed.", MA_FAILED_TO_CONFIGURE_BACKEND_DEVICE); + } -typedef struct -{ - ma_device_info* pDeviceInfo; - ma_bool32 foundDevice; -} ma_context_get_device_info_callback_data__pulse; + ma__free_from_callbacks(pSWParams, &pContext->allocationCallbacks); + pSWParams = NULL; -void ma_context_get_device_info_sink_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_sink_info* pInfo, int endOfList, void* pUserData) -{ - ma_context_get_device_info_callback_data__pulse* pData = (ma_context_get_device_info_callback_data__pulse*)pUserData; - if (endOfList > 0) { - return; - } + /* Grab the internal channel map. For now we're not going to bother trying to change the channel map and instead just do it ourselves. */ + { + ma_snd_pcm_chmap_t* pChmap = ((ma_snd_pcm_get_chmap_proc)pContext->alsa.snd_pcm_get_chmap)(pPCM); + if (pChmap != NULL) { + ma_uint32 iChannel; - ma_assert(pData != NULL); - pData->foundDevice = MA_TRUE; + /* There are cases where the returned channel map can have a different channel count than was returned by snd_pcm_hw_params_set_channels_near(). */ + if (pChmap->channels >= internalChannels) { + /* Drop excess channels. */ + for (iChannel = 0; iChannel < internalChannels; ++iChannel) { + internalChannelMap[iChannel] = ma_convert_alsa_channel_position_to_ma_channel(pChmap->pos[iChannel]); + } + } else { + ma_uint32 i; - if (pInfo->name != NULL) { - ma_strncpy_s(pData->pDeviceInfo->id.pulse, sizeof(pData->pDeviceInfo->id.pulse), pInfo->name, (size_t)-1); - } + /* + Excess channels use defaults. Do an initial fill with defaults, overwrite the first pChmap->channels, validate to ensure there are no duplicate + channels. If validation fails, fall back to defaults. + */ + ma_bool32 isValid = MA_TRUE; - if (pInfo->description != NULL) { - ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pInfo->description, (size_t)-1); - } + /* Fill with defaults. */ + ma_get_standard_channel_map(ma_standard_channel_map_alsa, internalChannels, internalChannelMap); - pData->pDeviceInfo->minChannels = pInfo->sample_spec.channels; - pData->pDeviceInfo->maxChannels = pInfo->sample_spec.channels; - pData->pDeviceInfo->minSampleRate = pInfo->sample_spec.rate; - pData->pDeviceInfo->maxSampleRate = pInfo->sample_spec.rate; - pData->pDeviceInfo->formatCount = 1; - pData->pDeviceInfo->formats[0] = ma_format_from_pulse(pInfo->sample_spec.format); + /* Overwrite first pChmap->channels channels. */ + for (iChannel = 0; iChannel < pChmap->channels; ++iChannel) { + internalChannelMap[iChannel] = ma_convert_alsa_channel_position_to_ma_channel(pChmap->pos[iChannel]); + } - (void)pPulseContext; /* Unused. */ -} + /* Validate. */ + for (i = 0; i < internalChannels && isValid; ++i) { + ma_uint32 j; + for (j = i+1; j < internalChannels; ++j) { + if (internalChannelMap[i] == internalChannelMap[j]) { + isValid = MA_FALSE; + break; + } + } + } -void ma_context_get_device_info_source_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_source_info* pInfo, int endOfList, void* pUserData) -{ - ma_context_get_device_info_callback_data__pulse* pData = (ma_context_get_device_info_callback_data__pulse*)pUserData; + /* If our channel map is invalid, fall back to defaults. */ + if (!isValid) { + ma_get_standard_channel_map(ma_standard_channel_map_alsa, internalChannels, internalChannelMap); + } + } - if (endOfList > 0) { - return; + free(pChmap); + pChmap = NULL; + } else { + /* Could not retrieve the channel map. Fall back to a hard-coded assumption. */ + ma_get_standard_channel_map(ma_standard_channel_map_alsa, internalChannels, internalChannelMap); + } } - ma_assert(pData != NULL); - pData->foundDevice = MA_TRUE; - if (pInfo->name != NULL) { - ma_strncpy_s(pData->pDeviceInfo->id.pulse, sizeof(pData->pDeviceInfo->id.pulse), pInfo->name, (size_t)-1); + /* We're done. Prepare the device. */ + if (((ma_snd_pcm_prepare_proc)pDevice->pContext->alsa.snd_pcm_prepare)(pPCM) < 0) { + ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to prepare device.", MA_FAILED_TO_START_BACKEND_DEVICE); } - if (pInfo->description != NULL) { - ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pInfo->description, (size_t)-1); - } - pData->pDeviceInfo->minChannels = pInfo->sample_spec.channels; - pData->pDeviceInfo->maxChannels = pInfo->sample_spec.channels; - pData->pDeviceInfo->minSampleRate = pInfo->sample_spec.rate; - pData->pDeviceInfo->maxSampleRate = pInfo->sample_spec.rate; - pData->pDeviceInfo->formatCount = 1; - pData->pDeviceInfo->formats[0] = ma_format_from_pulse(pInfo->sample_spec.format); + if (deviceType == ma_device_type_capture) { + pDevice->alsa.pPCMCapture = (ma_ptr)pPCM; + pDevice->alsa.isUsingMMapCapture = isUsingMMap; + pDevice->capture.internalFormat = internalFormat; + pDevice->capture.internalChannels = internalChannels; + pDevice->capture.internalSampleRate = internalSampleRate; + ma_channel_map_copy(pDevice->capture.internalChannelMap, internalChannelMap, internalChannels); + pDevice->capture.internalPeriodSizeInFrames = internalPeriodSizeInFrames; + pDevice->capture.internalPeriods = internalPeriods; + } else { + pDevice->alsa.pPCMPlayback = (ma_ptr)pPCM; + pDevice->alsa.isUsingMMapPlayback = isUsingMMap; + pDevice->playback.internalFormat = internalFormat; + pDevice->playback.internalChannels = internalChannels; + pDevice->playback.internalSampleRate = internalSampleRate; + ma_channel_map_copy(pDevice->playback.internalChannelMap, internalChannelMap, internalChannels); + pDevice->playback.internalPeriodSizeInFrames = internalPeriodSizeInFrames; + pDevice->playback.internalPeriods = internalPeriods; + } - (void)pPulseContext; /* Unused. */ + return MA_SUCCESS; } -ma_result ma_context_get_device_info__pulse(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) +static ma_result ma_device_init__alsa(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) { - ma_result result = MA_SUCCESS; - ma_context_get_device_info_callback_data__pulse callbackData; - ma_pa_operation* pOP = NULL; - ma_pa_mainloop* pMainLoop; - ma_pa_mainloop_api* pAPI; - ma_pa_context* pPulseContext; - int error; + MA_ASSERT(pDevice != NULL); - ma_assert(pContext != NULL); + MA_ZERO_OBJECT(&pDevice->alsa); - /* No exclusive mode with the PulseAudio backend. */ - if (shareMode == ma_share_mode_exclusive) { - return MA_SHARE_MODE_NOT_SUPPORTED; + if (pConfig->deviceType == ma_device_type_loopback) { + return MA_DEVICE_TYPE_NOT_SUPPORTED; } - callbackData.pDeviceInfo = pDeviceInfo; - callbackData.foundDevice = MA_FALSE; - - pMainLoop = ((ma_pa_mainloop_new_proc)pContext->pulse.pa_mainloop_new)(); - if (pMainLoop == NULL) { - return MA_FAILED_TO_INIT_BACKEND; + if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { + ma_result result = ma_device_init_by_type__alsa(pContext, pConfig, ma_device_type_capture, pDevice); + if (result != MA_SUCCESS) { + return result; + } } - pAPI = ((ma_pa_mainloop_get_api_proc)pContext->pulse.pa_mainloop_get_api)(pMainLoop); - if (pAPI == NULL) { - ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); - return MA_FAILED_TO_INIT_BACKEND; + if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { + ma_result result = ma_device_init_by_type__alsa(pContext, pConfig, ma_device_type_playback, pDevice); + if (result != MA_SUCCESS) { + return result; + } } - pPulseContext = ((ma_pa_context_new_proc)pContext->pulse.pa_context_new)(pAPI, pContext->pulse.pApplicationName); - if (pPulseContext == NULL) { - ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); - return MA_FAILED_TO_INIT_BACKEND; - } + return MA_SUCCESS; +} - error = ((ma_pa_context_connect_proc)pContext->pulse.pa_context_connect)(pPulseContext, pContext->pulse.pServerName, 0, NULL); - if (error != MA_PA_OK) { - ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)(pPulseContext); - ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); - return ma_result_from_pulse(error); +static ma_result ma_device_read__alsa(ma_device* pDevice, void* pFramesOut, ma_uint32 frameCount, ma_uint32* pFramesRead) +{ + ma_snd_pcm_sframes_t resultALSA; + + MA_ASSERT(pDevice != NULL); + MA_ASSERT(pFramesOut != NULL); + + if (pFramesRead != NULL) { + *pFramesRead = 0; } for (;;) { - ma_pa_context_state_t state = ((ma_pa_context_get_state_proc)pContext->pulse.pa_context_get_state)(pPulseContext); - if (state == MA_PA_CONTEXT_READY) { + resultALSA = ((ma_snd_pcm_readi_proc)pDevice->pContext->alsa.snd_pcm_readi)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture, pFramesOut, frameCount); + if (resultALSA >= 0) { break; /* Success. */ - } - if (state == MA_PA_CONTEXT_CONNECTING || state == MA_PA_CONTEXT_AUTHORIZING || state == MA_PA_CONTEXT_SETTING_NAME) { - error = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)(pMainLoop, 1, NULL); - if (error < 0) { - result = ma_result_from_pulse(error); - goto done; - } + } else { + if (resultALSA == -EAGAIN) { + /*printf("TRACE: EGAIN (read)\n");*/ + continue; /* Try again. */ + } else if (resultALSA == -EPIPE) { + #if defined(MA_DEBUG_OUTPUT) + printf("TRACE: EPIPE (read)\n"); + #endif -#ifdef MA_DEBUG_OUTPUT - printf("[PulseAudio] pa_context_get_state() returned %d. Waiting.\n", state); -#endif - continue; /* Keep trying. */ - } - if (state == MA_PA_CONTEXT_UNCONNECTED || state == MA_PA_CONTEXT_FAILED || state == MA_PA_CONTEXT_TERMINATED) { -#ifdef MA_DEBUG_OUTPUT - printf("[PulseAudio] pa_context_get_state() returned %d. Failed.\n", state); -#endif - goto done; /* Failed. */ - } - } + /* Overrun. Recover and try again. If this fails we need to return an error. */ + if (((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture, resultALSA, MA_TRUE) < 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to recover device after overrun.", MA_FAILED_TO_START_BACKEND_DEVICE); + } - if (deviceType == ma_device_type_playback) { - pOP = ((ma_pa_context_get_sink_info_by_name_proc)pContext->pulse.pa_context_get_sink_info_by_name)(pPulseContext, pDeviceID->pulse, ma_context_get_device_info_sink_callback__pulse, &callbackData); - } else { - pOP = ((ma_pa_context_get_source_info_by_name_proc)pContext->pulse.pa_context_get_source_info_by_name)(pPulseContext, pDeviceID->pulse, ma_context_get_device_info_source_callback__pulse, &callbackData); + if (((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture) < 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start device after underrun.", MA_FAILED_TO_START_BACKEND_DEVICE); + } + + resultALSA = ((ma_snd_pcm_readi_proc)pDevice->pContext->alsa.snd_pcm_readi)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture, pFramesOut, frameCount); + if (resultALSA < 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to read data from the internal device.", MA_FAILED_TO_READ_DATA_FROM_DEVICE); + } + } + } } - if (pOP != NULL) { - ma_wait_for_operation__pulse(pContext, pMainLoop, pOP); - ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); - } else { - result = MA_ERROR; - goto done; + if (pFramesRead != NULL) { + *pFramesRead = resultALSA; } - if (!callbackData.foundDevice) { - result = MA_NO_DEVICE; - goto done; - } + return MA_SUCCESS; +} +static ma_result ma_device_write__alsa(ma_device* pDevice, const void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) +{ + ma_snd_pcm_sframes_t resultALSA; -done: - ((ma_pa_context_disconnect_proc)pContext->pulse.pa_context_disconnect)(pPulseContext); - ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)(pPulseContext); - ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); - return result; -} + MA_ASSERT(pDevice != NULL); + MA_ASSERT(pFrames != NULL); + if (pFramesWritten != NULL) { + *pFramesWritten = 0; + } -void ma_pulse_device_state_callback(ma_pa_context* pPulseContext, void* pUserData) -{ - ma_device* pDevice; - ma_context* pContext; + for (;;) { + resultALSA = ((ma_snd_pcm_writei_proc)pDevice->pContext->alsa.snd_pcm_writei)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback, pFrames, frameCount); + if (resultALSA >= 0) { + break; /* Success. */ + } else { + if (resultALSA == -EAGAIN) { + /*printf("TRACE: EGAIN (write)\n");*/ + continue; /* Try again. */ + } else if (resultALSA == -EPIPE) { + #if defined(MA_DEBUG_OUTPUT) + printf("TRACE: EPIPE (write)\n"); + #endif - pDevice = (ma_device*)pUserData; - ma_assert(pDevice != NULL); + /* Underrun. Recover and try again. If this fails we need to return an error. */ + if (((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback, resultALSA, MA_TRUE) < 0) { /* MA_TRUE=silent (don't print anything on error). */ + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to recover device after underrun.", MA_FAILED_TO_START_BACKEND_DEVICE); + } - pContext = pDevice->pContext; - ma_assert(pContext != NULL); + /* + In my testing I have had a situation where writei() does not automatically restart the device even though I've set it + up as such in the software parameters. What will happen is writei() will block indefinitely even though the number of + frames is well beyond the auto-start threshold. To work around this I've needed to add an explicit start here. Not sure + if this is me just being stupid and not recovering the device properly, but this definitely feels like something isn't + quite right here. + */ + if (((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback) < 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start device after underrun.", MA_FAILED_TO_START_BACKEND_DEVICE); + } - pDevice->pulse.pulseContextState = ((ma_pa_context_get_state_proc)pContext->pulse.pa_context_get_state)(pPulseContext); + resultALSA = ((ma_snd_pcm_writei_proc)pDevice->pContext->alsa.snd_pcm_writei)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback, pFrames, frameCount); + if (resultALSA < 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to write data to device after underrun.", MA_FAILED_TO_START_BACKEND_DEVICE); + } + } + } + } + + if (pFramesWritten != NULL) { + *pFramesWritten = resultALSA; + } + + return MA_SUCCESS; } -void ma_device_sink_info_callback(ma_pa_context* pPulseContext, const ma_pa_sink_info* pInfo, int endOfList, void* pUserData) +static ma_result ma_device_main_loop__alsa(ma_device* pDevice) { - ma_pa_sink_info* pInfoOut; + ma_result result = MA_SUCCESS; + ma_bool32 exitLoop = MA_FALSE; - if (endOfList > 0) { - return; + MA_ASSERT(pDevice != NULL); + + /* Capture devices need to be started immediately. */ + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + if (((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture) < 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start device in preparation for reading.", MA_FAILED_TO_START_BACKEND_DEVICE); + } } - pInfoOut = (ma_pa_sink_info*)pUserData; - ma_assert(pInfoOut != NULL); + while (ma_device__get_state(pDevice) == MA_STATE_STARTED && !exitLoop) { + switch (pDevice->type) + { + case ma_device_type_duplex: + { + if (pDevice->alsa.isUsingMMapCapture || pDevice->alsa.isUsingMMapPlayback) { + /* MMAP */ + return MA_INVALID_OPERATION; /* Not yet implemented. */ + } else { + /* readi() and writei() */ - *pInfoOut = *pInfo; + /* The process is: device_read -> convert -> callback -> convert -> device_write */ + ma_uint32 totalCapturedDeviceFramesProcessed = 0; + ma_uint32 capturedDevicePeriodSizeInFrames = ma_min(pDevice->capture.internalPeriodSizeInFrames, pDevice->playback.internalPeriodSizeInFrames); + + while (totalCapturedDeviceFramesProcessed < capturedDevicePeriodSizeInFrames) { + ma_uint8 capturedDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint8 playbackDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 capturedDeviceDataCapInFrames = sizeof(capturedDeviceData) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + ma_uint32 playbackDeviceDataCapInFrames = sizeof(playbackDeviceData) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + ma_uint32 capturedDeviceFramesRemaining; + ma_uint32 capturedDeviceFramesProcessed; + ma_uint32 capturedDeviceFramesToProcess; + ma_uint32 capturedDeviceFramesToTryProcessing = capturedDevicePeriodSizeInFrames - totalCapturedDeviceFramesProcessed; + if (capturedDeviceFramesToTryProcessing > capturedDeviceDataCapInFrames) { + capturedDeviceFramesToTryProcessing = capturedDeviceDataCapInFrames; + } - (void)pPulseContext; /* Unused. */ -} + result = ma_device_read__alsa(pDevice, capturedDeviceData, capturedDeviceFramesToTryProcessing, &capturedDeviceFramesToProcess); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } -void ma_device_source_info_callback(ma_pa_context* pPulseContext, const ma_pa_source_info* pInfo, int endOfList, void* pUserData) -{ - ma_pa_source_info* pInfoOut; + capturedDeviceFramesRemaining = capturedDeviceFramesToProcess; + capturedDeviceFramesProcessed = 0; - if (endOfList > 0) { - return; - } + for (;;) { + ma_uint8 capturedClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint8 playbackClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 capturedClientDataCapInFrames = sizeof(capturedClientData) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); + ma_uint32 playbackClientDataCapInFrames = sizeof(playbackClientData) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); + ma_uint64 capturedClientFramesToProcessThisIteration = ma_min(capturedClientDataCapInFrames, playbackClientDataCapInFrames); + ma_uint64 capturedDeviceFramesToProcessThisIteration = capturedDeviceFramesRemaining; + ma_uint8* pRunningCapturedDeviceFrames = ma_offset_ptr(capturedDeviceData, capturedDeviceFramesProcessed * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); + + /* Convert capture data from device format to client format. */ + result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningCapturedDeviceFrames, &capturedDeviceFramesToProcessThisIteration, capturedClientData, &capturedClientFramesToProcessThisIteration); + if (result != MA_SUCCESS) { + break; + } - pInfoOut = (ma_pa_source_info*)pUserData; - ma_assert(pInfoOut != NULL); + /* + If we weren't able to generate any output frames it must mean we've exhaused all of our input. The only time this would not be the case is if capturedClientData was too small + which should never be the case when it's of the size MA_DATA_CONVERTER_STACK_BUFFER_SIZE. + */ + if (capturedClientFramesToProcessThisIteration == 0) { + break; + } - *pInfoOut = *pInfo; + ma_device__on_data(pDevice, playbackClientData, capturedClientData, (ma_uint32)capturedClientFramesToProcessThisIteration); /* Safe cast .*/ - (void)pPulseContext; /* Unused. */ -} + capturedDeviceFramesProcessed += (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */ + capturedDeviceFramesRemaining -= (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */ -void ma_device_sink_name_callback(ma_pa_context* pPulseContext, const ma_pa_sink_info* pInfo, int endOfList, void* pUserData) -{ - ma_device* pDevice; + /* At this point the playbackClientData buffer should be holding data that needs to be written to the device. */ + for (;;) { + ma_uint64 convertedClientFrameCount = capturedClientFramesToProcessThisIteration; + ma_uint64 convertedDeviceFrameCount = playbackDeviceDataCapInFrames; + result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, playbackClientData, &convertedClientFrameCount, playbackDeviceData, &convertedDeviceFrameCount); + if (result != MA_SUCCESS) { + break; + } - if (endOfList > 0) { - return; - } + result = ma_device_write__alsa(pDevice, playbackDeviceData, (ma_uint32)convertedDeviceFrameCount, NULL); /* Safe cast. */ + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } - pDevice = (ma_device*)pUserData; - ma_assert(pDevice != NULL); + capturedClientFramesToProcessThisIteration -= (ma_uint32)convertedClientFrameCount; /* Safe cast. */ + if (capturedClientFramesToProcessThisIteration == 0) { + break; + } + } - ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), pInfo->description, (size_t)-1); + /* In case an error happened from ma_device_write__alsa()... */ + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } + } - (void)pPulseContext; /* Unused. */ -} + totalCapturedDeviceFramesProcessed += capturedDeviceFramesProcessed; + } + } + } break; -void ma_device_source_name_callback(ma_pa_context* pPulseContext, const ma_pa_source_info* pInfo, int endOfList, void* pUserData) -{ - ma_device* pDevice; + case ma_device_type_capture: + { + if (pDevice->alsa.isUsingMMapCapture) { + /* MMAP */ + return MA_INVALID_OPERATION; /* Not yet implemented. */ + } else { + /* readi() */ - if (endOfList > 0) { - return; - } + /* We read in chunks of the period size, but use a stack allocated buffer for the intermediary. */ + ma_uint8 intermediaryBuffer[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + ma_uint32 periodSizeInFrames = pDevice->capture.internalPeriodSizeInFrames; + ma_uint32 framesReadThisPeriod = 0; + while (framesReadThisPeriod < periodSizeInFrames) { + ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesReadThisPeriod; + ma_uint32 framesProcessed; + ma_uint32 framesToReadThisIteration = framesRemainingInPeriod; + if (framesToReadThisIteration > intermediaryBufferSizeInFrames) { + framesToReadThisIteration = intermediaryBufferSizeInFrames; + } - pDevice = (ma_device*)pUserData; - ma_assert(pDevice != NULL); + result = ma_device_read__alsa(pDevice, intermediaryBuffer, framesToReadThisIteration, &framesProcessed); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } - ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), pInfo->description, (size_t)-1); + ma_device__send_frames_to_client(pDevice, framesProcessed, intermediaryBuffer); - (void)pPulseContext; /* Unused. */ -} + framesReadThisPeriod += framesProcessed; + } + } + } break; -void ma_device_uninit__pulse(ma_device* pDevice) -{ - ma_context* pContext; + case ma_device_type_playback: + { + if (pDevice->alsa.isUsingMMapPlayback) { + /* MMAP */ + return MA_INVALID_OPERATION; /* Not yet implemented. */ + } else { + /* writei() */ - ma_assert(pDevice != NULL); + /* We write in chunks of the period size, but use a stack allocated buffer for the intermediary. */ + ma_uint8 intermediaryBuffer[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + ma_uint32 periodSizeInFrames = pDevice->playback.internalPeriodSizeInFrames; + ma_uint32 framesWrittenThisPeriod = 0; + while (framesWrittenThisPeriod < periodSizeInFrames) { + ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesWrittenThisPeriod; + ma_uint32 framesProcessed; + ma_uint32 framesToWriteThisIteration = framesRemainingInPeriod; + if (framesToWriteThisIteration > intermediaryBufferSizeInFrames) { + framesToWriteThisIteration = intermediaryBufferSizeInFrames; + } - pContext = pDevice->pContext; - ma_assert(pContext != NULL); + ma_device__read_frames_from_client(pDevice, framesToWriteThisIteration, intermediaryBuffer); - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - ((ma_pa_stream_disconnect_proc)pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamCapture); - ((ma_pa_stream_unref_proc)pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamCapture); + result = ma_device_write__alsa(pDevice, intermediaryBuffer, framesToWriteThisIteration, &framesProcessed); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } + + framesWrittenThisPeriod += framesProcessed; + } + } + } break; + + /* To silence a warning. Will never hit this. */ + case ma_device_type_loopback: + default: break; + } } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - ((ma_pa_stream_disconnect_proc)pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); - ((ma_pa_stream_unref_proc)pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); + + /* Here is where the device needs to be stopped. */ + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + ((ma_snd_pcm_drain_proc)pDevice->pContext->alsa.snd_pcm_drain)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture); + + /* We need to prepare the device again, otherwise we won't be able to restart the device. */ + if (((ma_snd_pcm_prepare_proc)pDevice->pContext->alsa.snd_pcm_prepare)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture) < 0) { + #ifdef MA_DEBUG_OUTPUT + printf("[ALSA] Failed to prepare capture device after stopping.\n"); + #endif + } } - ((ma_pa_context_disconnect_proc)pContext->pulse.pa_context_disconnect)((ma_pa_context*)pDevice->pulse.pPulseContext); - ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)((ma_pa_context*)pDevice->pulse.pPulseContext); - ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)pDevice->pulse.pMainLoop); -} + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + ((ma_snd_pcm_drain_proc)pDevice->pContext->alsa.snd_pcm_drain)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback); -ma_pa_buffer_attr ma_device__pa_buffer_attr_new(ma_uint32 bufferSizeInFrames, ma_uint32 periods, const ma_pa_sample_spec* ss) -{ - ma_pa_buffer_attr attr; - attr.maxlength = bufferSizeInFrames * ma_get_bytes_per_sample(ma_format_from_pulse(ss->format)) * ss->channels; - attr.tlength = attr.maxlength / periods; - attr.prebuf = (ma_uint32)-1; - attr.minreq = (ma_uint32)-1; - attr.fragsize = attr.maxlength / periods; + /* We need to prepare the device again, otherwise we won't be able to restart the device. */ + if (((ma_snd_pcm_prepare_proc)pDevice->pContext->alsa.snd_pcm_prepare)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback) < 0) { + #ifdef MA_DEBUG_OUTPUT + printf("[ALSA] Failed to prepare playback device after stopping.\n"); + #endif + } + } - return attr; + return result; } -ma_pa_stream* ma_device__pa_stream_new__pulse(ma_device* pDevice, const char* pStreamName, const ma_pa_sample_spec* ss, const ma_pa_channel_map* cmap) +static ma_result ma_context_uninit__alsa(ma_context* pContext) { - static int g_StreamCounter = 0; - char actualStreamName[256]; + MA_ASSERT(pContext != NULL); + MA_ASSERT(pContext->backend == ma_backend_alsa); - if (pStreamName != NULL) { - ma_strncpy_s(actualStreamName, sizeof(actualStreamName), pStreamName, (size_t)-1); - } else { - ma_strcpy_s(actualStreamName, sizeof(actualStreamName), "miniaudio:"); - ma_itoa_s(g_StreamCounter, actualStreamName + 8, sizeof(actualStreamName)-8, 10); /* 8 = strlen("miniaudio:") */ - } - g_StreamCounter += 1; + /* Clean up memory for memory leak checkers. */ + ((ma_snd_config_update_free_global_proc)pContext->alsa.snd_config_update_free_global)(); - return ((ma_pa_stream_new_proc)pDevice->pContext->pulse.pa_stream_new)((ma_pa_context*)pDevice->pulse.pPulseContext, actualStreamName, ss, cmap); -} +#ifndef MA_NO_RUNTIME_LINKING + ma_dlclose(pContext, pContext->alsa.asoundSO); +#endif -ma_result ma_device_init__pulse(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) -{ - ma_result result = MA_SUCCESS; - int error = 0; - const char* devPlayback = NULL; - const char* devCapture = NULL; - ma_uint32 bufferSizeInMilliseconds; - ma_pa_sink_info sinkInfo; - ma_pa_source_info sourceInfo; - ma_pa_operation* pOP = NULL; - ma_pa_sample_spec ss; - ma_pa_channel_map cmap; - ma_pa_buffer_attr attr; - const ma_pa_sample_spec* pActualSS = NULL; - const ma_pa_channel_map* pActualCMap = NULL; - const ma_pa_buffer_attr* pActualAttr = NULL; - ma_uint32 iChannel; - ma_pa_stream_flags_t streamFlags; + ma_mutex_uninit(&pContext->alsa.internalDeviceEnumLock); - ma_assert(pDevice != NULL); - ma_zero_object(&pDevice->pulse); + return MA_SUCCESS; +} - if (pConfig->deviceType == ma_device_type_loopback) { - return MA_DEVICE_TYPE_NOT_SUPPORTED; - } +static ma_result ma_context_init__alsa(const ma_context_config* pConfig, ma_context* pContext) +{ +#ifndef MA_NO_RUNTIME_LINKING + const char* libasoundNames[] = { + "libasound.so.2", + "libasound.so" + }; + size_t i; - /* No exclusive mode with the PulseAudio backend. */ - if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) || - ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) { - return MA_SHARE_MODE_NOT_SUPPORTED; + for (i = 0; i < ma_countof(libasoundNames); ++i) { + pContext->alsa.asoundSO = ma_dlopen(pContext, libasoundNames[i]); + if (pContext->alsa.asoundSO != NULL) { + break; + } } - if ((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.pDeviceID != NULL) { - devPlayback = pConfig->playback.pDeviceID->pulse; - } - if ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.pDeviceID != NULL) { - devCapture = pConfig->capture.pDeviceID->pulse; + if (pContext->alsa.asoundSO == NULL) { +#ifdef MA_DEBUG_OUTPUT + printf("[ALSA] Failed to open shared object.\n"); +#endif + return MA_NO_BACKEND; } - bufferSizeInMilliseconds = pConfig->bufferSizeInMilliseconds; - if (bufferSizeInMilliseconds == 0) { - bufferSizeInMilliseconds = ma_calculate_buffer_size_in_milliseconds_from_frames(pConfig->bufferSizeInFrames, pConfig->sampleRate); - } + pContext->alsa.snd_pcm_open = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_open"); + pContext->alsa.snd_pcm_close = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_close"); + pContext->alsa.snd_pcm_hw_params_sizeof = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_sizeof"); + pContext->alsa.snd_pcm_hw_params_any = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_any"); + pContext->alsa.snd_pcm_hw_params_set_format = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_format"); + pContext->alsa.snd_pcm_hw_params_set_format_first = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_format_first"); + pContext->alsa.snd_pcm_hw_params_get_format_mask = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_format_mask"); + pContext->alsa.snd_pcm_hw_params_set_channels_near = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_channels_near"); + pContext->alsa.snd_pcm_hw_params_set_rate_resample = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_rate_resample"); + pContext->alsa.snd_pcm_hw_params_set_rate_near = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_rate_near"); + pContext->alsa.snd_pcm_hw_params_set_buffer_size_near = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_buffer_size_near"); + pContext->alsa.snd_pcm_hw_params_set_periods_near = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_periods_near"); + pContext->alsa.snd_pcm_hw_params_set_access = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_access"); + pContext->alsa.snd_pcm_hw_params_get_format = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_format"); + pContext->alsa.snd_pcm_hw_params_get_channels = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_channels"); + pContext->alsa.snd_pcm_hw_params_get_channels_min = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_channels_min"); + pContext->alsa.snd_pcm_hw_params_get_channels_max = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_channels_max"); + pContext->alsa.snd_pcm_hw_params_get_rate = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_rate"); + pContext->alsa.snd_pcm_hw_params_get_rate_min = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_rate_min"); + pContext->alsa.snd_pcm_hw_params_get_rate_max = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_rate_max"); + pContext->alsa.snd_pcm_hw_params_get_buffer_size = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_buffer_size"); + pContext->alsa.snd_pcm_hw_params_get_periods = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_periods"); + pContext->alsa.snd_pcm_hw_params_get_access = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_access"); + pContext->alsa.snd_pcm_hw_params = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params"); + pContext->alsa.snd_pcm_sw_params_sizeof = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_sizeof"); + pContext->alsa.snd_pcm_sw_params_current = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_current"); + pContext->alsa.snd_pcm_sw_params_get_boundary = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_get_boundary"); + pContext->alsa.snd_pcm_sw_params_set_avail_min = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_set_avail_min"); + pContext->alsa.snd_pcm_sw_params_set_start_threshold = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_set_start_threshold"); + pContext->alsa.snd_pcm_sw_params_set_stop_threshold = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_set_stop_threshold"); + pContext->alsa.snd_pcm_sw_params = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params"); + pContext->alsa.snd_pcm_format_mask_sizeof = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_format_mask_sizeof"); + pContext->alsa.snd_pcm_format_mask_test = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_format_mask_test"); + pContext->alsa.snd_pcm_get_chmap = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_get_chmap"); + pContext->alsa.snd_pcm_state = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_state"); + pContext->alsa.snd_pcm_prepare = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_prepare"); + pContext->alsa.snd_pcm_start = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_start"); + pContext->alsa.snd_pcm_drop = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_drop"); + pContext->alsa.snd_pcm_drain = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_drain"); + pContext->alsa.snd_device_name_hint = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_device_name_hint"); + pContext->alsa.snd_device_name_get_hint = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_device_name_get_hint"); + pContext->alsa.snd_card_get_index = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_card_get_index"); + pContext->alsa.snd_device_name_free_hint = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_device_name_free_hint"); + pContext->alsa.snd_pcm_mmap_begin = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_mmap_begin"); + pContext->alsa.snd_pcm_mmap_commit = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_mmap_commit"); + pContext->alsa.snd_pcm_recover = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_recover"); + pContext->alsa.snd_pcm_readi = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_readi"); + pContext->alsa.snd_pcm_writei = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_writei"); + pContext->alsa.snd_pcm_avail = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_avail"); + pContext->alsa.snd_pcm_avail_update = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_avail_update"); + pContext->alsa.snd_pcm_wait = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_wait"); + pContext->alsa.snd_pcm_info = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_info"); + pContext->alsa.snd_pcm_info_sizeof = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_info_sizeof"); + pContext->alsa.snd_pcm_info_get_name = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_info_get_name"); + pContext->alsa.snd_config_update_free_global = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_config_update_free_global"); +#else + /* The system below is just for type safety. */ + ma_snd_pcm_open_proc _snd_pcm_open = snd_pcm_open; + ma_snd_pcm_close_proc _snd_pcm_close = snd_pcm_close; + ma_snd_pcm_hw_params_sizeof_proc _snd_pcm_hw_params_sizeof = snd_pcm_hw_params_sizeof; + ma_snd_pcm_hw_params_any_proc _snd_pcm_hw_params_any = snd_pcm_hw_params_any; + ma_snd_pcm_hw_params_set_format_proc _snd_pcm_hw_params_set_format = snd_pcm_hw_params_set_format; + ma_snd_pcm_hw_params_set_format_first_proc _snd_pcm_hw_params_set_format_first = snd_pcm_hw_params_set_format_first; + ma_snd_pcm_hw_params_get_format_mask_proc _snd_pcm_hw_params_get_format_mask = snd_pcm_hw_params_get_format_mask; + ma_snd_pcm_hw_params_set_channels_near_proc _snd_pcm_hw_params_set_channels_near = snd_pcm_hw_params_set_channels_near; + ma_snd_pcm_hw_params_set_rate_resample_proc _snd_pcm_hw_params_set_rate_resample = snd_pcm_hw_params_set_rate_resample; + ma_snd_pcm_hw_params_set_rate_near_proc _snd_pcm_hw_params_set_rate_near = snd_pcm_hw_params_set_rate_near; + ma_snd_pcm_hw_params_set_buffer_size_near_proc _snd_pcm_hw_params_set_buffer_size_near = snd_pcm_hw_params_set_buffer_size_near; + ma_snd_pcm_hw_params_set_periods_near_proc _snd_pcm_hw_params_set_periods_near = snd_pcm_hw_params_set_periods_near; + ma_snd_pcm_hw_params_set_access_proc _snd_pcm_hw_params_set_access = snd_pcm_hw_params_set_access; + ma_snd_pcm_hw_params_get_format_proc _snd_pcm_hw_params_get_format = snd_pcm_hw_params_get_format; + ma_snd_pcm_hw_params_get_channels_proc _snd_pcm_hw_params_get_channels = snd_pcm_hw_params_get_channels; + ma_snd_pcm_hw_params_get_channels_min_proc _snd_pcm_hw_params_get_channels_min = snd_pcm_hw_params_get_channels_min; + ma_snd_pcm_hw_params_get_channels_max_proc _snd_pcm_hw_params_get_channels_max = snd_pcm_hw_params_get_channels_max; + ma_snd_pcm_hw_params_get_rate_proc _snd_pcm_hw_params_get_rate = snd_pcm_hw_params_get_rate; + ma_snd_pcm_hw_params_get_rate_min_proc _snd_pcm_hw_params_get_rate_min = snd_pcm_hw_params_get_rate_min; + ma_snd_pcm_hw_params_get_rate_max_proc _snd_pcm_hw_params_get_rate_max = snd_pcm_hw_params_get_rate_max; + ma_snd_pcm_hw_params_get_buffer_size_proc _snd_pcm_hw_params_get_buffer_size = snd_pcm_hw_params_get_buffer_size; + ma_snd_pcm_hw_params_get_periods_proc _snd_pcm_hw_params_get_periods = snd_pcm_hw_params_get_periods; + ma_snd_pcm_hw_params_get_access_proc _snd_pcm_hw_params_get_access = snd_pcm_hw_params_get_access; + ma_snd_pcm_hw_params_proc _snd_pcm_hw_params = snd_pcm_hw_params; + ma_snd_pcm_sw_params_sizeof_proc _snd_pcm_sw_params_sizeof = snd_pcm_sw_params_sizeof; + ma_snd_pcm_sw_params_current_proc _snd_pcm_sw_params_current = snd_pcm_sw_params_current; + ma_snd_pcm_sw_params_get_boundary_proc _snd_pcm_sw_params_get_boundary = snd_pcm_sw_params_get_boundary; + ma_snd_pcm_sw_params_set_avail_min_proc _snd_pcm_sw_params_set_avail_min = snd_pcm_sw_params_set_avail_min; + ma_snd_pcm_sw_params_set_start_threshold_proc _snd_pcm_sw_params_set_start_threshold = snd_pcm_sw_params_set_start_threshold; + ma_snd_pcm_sw_params_set_stop_threshold_proc _snd_pcm_sw_params_set_stop_threshold = snd_pcm_sw_params_set_stop_threshold; + ma_snd_pcm_sw_params_proc _snd_pcm_sw_params = snd_pcm_sw_params; + ma_snd_pcm_format_mask_sizeof_proc _snd_pcm_format_mask_sizeof = snd_pcm_format_mask_sizeof; + ma_snd_pcm_format_mask_test_proc _snd_pcm_format_mask_test = snd_pcm_format_mask_test; + ma_snd_pcm_get_chmap_proc _snd_pcm_get_chmap = snd_pcm_get_chmap; + ma_snd_pcm_state_proc _snd_pcm_state = snd_pcm_state; + ma_snd_pcm_prepare_proc _snd_pcm_prepare = snd_pcm_prepare; + ma_snd_pcm_start_proc _snd_pcm_start = snd_pcm_start; + ma_snd_pcm_drop_proc _snd_pcm_drop = snd_pcm_drop; + ma_snd_pcm_drain_proc _snd_pcm_drain = snd_pcm_drain; + ma_snd_device_name_hint_proc _snd_device_name_hint = snd_device_name_hint; + ma_snd_device_name_get_hint_proc _snd_device_name_get_hint = snd_device_name_get_hint; + ma_snd_card_get_index_proc _snd_card_get_index = snd_card_get_index; + ma_snd_device_name_free_hint_proc _snd_device_name_free_hint = snd_device_name_free_hint; + ma_snd_pcm_mmap_begin_proc _snd_pcm_mmap_begin = snd_pcm_mmap_begin; + ma_snd_pcm_mmap_commit_proc _snd_pcm_mmap_commit = snd_pcm_mmap_commit; + ma_snd_pcm_recover_proc _snd_pcm_recover = snd_pcm_recover; + ma_snd_pcm_readi_proc _snd_pcm_readi = snd_pcm_readi; + ma_snd_pcm_writei_proc _snd_pcm_writei = snd_pcm_writei; + ma_snd_pcm_avail_proc _snd_pcm_avail = snd_pcm_avail; + ma_snd_pcm_avail_update_proc _snd_pcm_avail_update = snd_pcm_avail_update; + ma_snd_pcm_wait_proc _snd_pcm_wait = snd_pcm_wait; + ma_snd_pcm_info_proc _snd_pcm_info = snd_pcm_info; + ma_snd_pcm_info_sizeof_proc _snd_pcm_info_sizeof = snd_pcm_info_sizeof; + ma_snd_pcm_info_get_name_proc _snd_pcm_info_get_name = snd_pcm_info_get_name; + ma_snd_config_update_free_global_proc _snd_config_update_free_global = snd_config_update_free_global; - pDevice->pulse.pMainLoop = ((ma_pa_mainloop_new_proc)pContext->pulse.pa_mainloop_new)(); - if (pDevice->pulse.pMainLoop == NULL) { - result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create main loop for device.", MA_FAILED_TO_INIT_BACKEND); - goto on_error0; - } + pContext->alsa.snd_pcm_open = (ma_proc)_snd_pcm_open; + pContext->alsa.snd_pcm_close = (ma_proc)_snd_pcm_close; + pContext->alsa.snd_pcm_hw_params_sizeof = (ma_proc)_snd_pcm_hw_params_sizeof; + pContext->alsa.snd_pcm_hw_params_any = (ma_proc)_snd_pcm_hw_params_any; + pContext->alsa.snd_pcm_hw_params_set_format = (ma_proc)_snd_pcm_hw_params_set_format; + pContext->alsa.snd_pcm_hw_params_set_format_first = (ma_proc)_snd_pcm_hw_params_set_format_first; + pContext->alsa.snd_pcm_hw_params_get_format_mask = (ma_proc)_snd_pcm_hw_params_get_format_mask; + pContext->alsa.snd_pcm_hw_params_set_channels_near = (ma_proc)_snd_pcm_hw_params_set_channels_near; + pContext->alsa.snd_pcm_hw_params_set_rate_resample = (ma_proc)_snd_pcm_hw_params_set_rate_resample; + pContext->alsa.snd_pcm_hw_params_set_rate_near = (ma_proc)_snd_pcm_hw_params_set_rate_near; + pContext->alsa.snd_pcm_hw_params_set_buffer_size_near = (ma_proc)_snd_pcm_hw_params_set_buffer_size_near; + pContext->alsa.snd_pcm_hw_params_set_periods_near = (ma_proc)_snd_pcm_hw_params_set_periods_near; + pContext->alsa.snd_pcm_hw_params_set_access = (ma_proc)_snd_pcm_hw_params_set_access; + pContext->alsa.snd_pcm_hw_params_get_format = (ma_proc)_snd_pcm_hw_params_get_format; + pContext->alsa.snd_pcm_hw_params_get_channels = (ma_proc)_snd_pcm_hw_params_get_channels; + pContext->alsa.snd_pcm_hw_params_get_channels_min = (ma_proc)_snd_pcm_hw_params_get_channels_min; + pContext->alsa.snd_pcm_hw_params_get_channels_max = (ma_proc)_snd_pcm_hw_params_get_channels_max; + pContext->alsa.snd_pcm_hw_params_get_rate = (ma_proc)_snd_pcm_hw_params_get_rate; + pContext->alsa.snd_pcm_hw_params_get_buffer_size = (ma_proc)_snd_pcm_hw_params_get_buffer_size; + pContext->alsa.snd_pcm_hw_params_get_periods = (ma_proc)_snd_pcm_hw_params_get_periods; + pContext->alsa.snd_pcm_hw_params_get_access = (ma_proc)_snd_pcm_hw_params_get_access; + pContext->alsa.snd_pcm_hw_params = (ma_proc)_snd_pcm_hw_params; + pContext->alsa.snd_pcm_sw_params_sizeof = (ma_proc)_snd_pcm_sw_params_sizeof; + pContext->alsa.snd_pcm_sw_params_current = (ma_proc)_snd_pcm_sw_params_current; + pContext->alsa.snd_pcm_sw_params_get_boundary = (ma_proc)_snd_pcm_sw_params_get_boundary; + pContext->alsa.snd_pcm_sw_params_set_avail_min = (ma_proc)_snd_pcm_sw_params_set_avail_min; + pContext->alsa.snd_pcm_sw_params_set_start_threshold = (ma_proc)_snd_pcm_sw_params_set_start_threshold; + pContext->alsa.snd_pcm_sw_params_set_stop_threshold = (ma_proc)_snd_pcm_sw_params_set_stop_threshold; + pContext->alsa.snd_pcm_sw_params = (ma_proc)_snd_pcm_sw_params; + pContext->alsa.snd_pcm_format_mask_sizeof = (ma_proc)_snd_pcm_format_mask_sizeof; + pContext->alsa.snd_pcm_format_mask_test = (ma_proc)_snd_pcm_format_mask_test; + pContext->alsa.snd_pcm_get_chmap = (ma_proc)_snd_pcm_get_chmap; + pContext->alsa.snd_pcm_state = (ma_proc)_snd_pcm_state; + pContext->alsa.snd_pcm_prepare = (ma_proc)_snd_pcm_prepare; + pContext->alsa.snd_pcm_start = (ma_proc)_snd_pcm_start; + pContext->alsa.snd_pcm_drop = (ma_proc)_snd_pcm_drop; + pContext->alsa.snd_pcm_drain = (ma_proc)_snd_pcm_drain; + pContext->alsa.snd_device_name_hint = (ma_proc)_snd_device_name_hint; + pContext->alsa.snd_device_name_get_hint = (ma_proc)_snd_device_name_get_hint; + pContext->alsa.snd_card_get_index = (ma_proc)_snd_card_get_index; + pContext->alsa.snd_device_name_free_hint = (ma_proc)_snd_device_name_free_hint; + pContext->alsa.snd_pcm_mmap_begin = (ma_proc)_snd_pcm_mmap_begin; + pContext->alsa.snd_pcm_mmap_commit = (ma_proc)_snd_pcm_mmap_commit; + pContext->alsa.snd_pcm_recover = (ma_proc)_snd_pcm_recover; + pContext->alsa.snd_pcm_readi = (ma_proc)_snd_pcm_readi; + pContext->alsa.snd_pcm_writei = (ma_proc)_snd_pcm_writei; + pContext->alsa.snd_pcm_avail = (ma_proc)_snd_pcm_avail; + pContext->alsa.snd_pcm_avail_update = (ma_proc)_snd_pcm_avail_update; + pContext->alsa.snd_pcm_wait = (ma_proc)_snd_pcm_wait; + pContext->alsa.snd_pcm_info = (ma_proc)_snd_pcm_info; + pContext->alsa.snd_pcm_info_sizeof = (ma_proc)_snd_pcm_info_sizeof; + pContext->alsa.snd_pcm_info_get_name = (ma_proc)_snd_pcm_info_get_name; + pContext->alsa.snd_config_update_free_global = (ma_proc)_snd_config_update_free_global; +#endif - pDevice->pulse.pAPI = ((ma_pa_mainloop_get_api_proc)pContext->pulse.pa_mainloop_get_api)((ma_pa_mainloop*)pDevice->pulse.pMainLoop); - if (pDevice->pulse.pAPI == NULL) { - result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to retrieve PulseAudio main loop.", MA_FAILED_TO_INIT_BACKEND); - goto on_error1; - } + pContext->alsa.useVerboseDeviceEnumeration = pConfig->alsa.useVerboseDeviceEnumeration; - pDevice->pulse.pPulseContext = ((ma_pa_context_new_proc)pContext->pulse.pa_context_new)((ma_pa_mainloop_api*)pDevice->pulse.pAPI, pContext->pulse.pApplicationName); - if (pDevice->pulse.pPulseContext == NULL) { - result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create PulseAudio context for device.", MA_FAILED_TO_INIT_BACKEND); - goto on_error1; + if (ma_mutex_init(pContext, &pContext->alsa.internalDeviceEnumLock) != MA_SUCCESS) { + ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[ALSA] WARNING: Failed to initialize mutex for internal device enumeration.", MA_ERROR); } - error = ((ma_pa_context_connect_proc)pContext->pulse.pa_context_connect)((ma_pa_context*)pDevice->pulse.pPulseContext, pContext->pulse.pServerName, (pContext->pulse.tryAutoSpawn) ? 0 : MA_PA_CONTEXT_NOAUTOSPAWN, NULL); - if (error != MA_PA_OK) { - result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to connect PulseAudio context.", ma_result_from_pulse(error)); - goto on_error2; - } + pContext->onUninit = ma_context_uninit__alsa; + pContext->onDeviceIDEqual = ma_context_is_device_id_equal__alsa; + pContext->onEnumDevices = ma_context_enumerate_devices__alsa; + pContext->onGetDeviceInfo = ma_context_get_device_info__alsa; + pContext->onDeviceInit = ma_device_init__alsa; + pContext->onDeviceUninit = ma_device_uninit__alsa; + pContext->onDeviceStart = NULL; /* Not used. Started in the main loop. */ + pContext->onDeviceStop = NULL; /* Not used. Started in the main loop. */ + pContext->onDeviceMainLoop = ma_device_main_loop__alsa; + return MA_SUCCESS; +} +#endif /* ALSA */ - pDevice->pulse.pulseContextState = MA_PA_CONTEXT_UNCONNECTED; - ((ma_pa_context_set_state_callback_proc)pContext->pulse.pa_context_set_state_callback)((ma_pa_context*)pDevice->pulse.pPulseContext, ma_pulse_device_state_callback, pDevice); - /* Wait for PulseAudio to get itself ready before returning. */ - for (;;) { - if (pDevice->pulse.pulseContextState == MA_PA_CONTEXT_READY) { - break; - } - /* An error may have occurred. */ - if (pDevice->pulse.pulseContextState == MA_PA_CONTEXT_FAILED || pDevice->pulse.pulseContextState == MA_PA_CONTEXT_TERMINATED) { - result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] An error occurred while connecting the PulseAudio context.", MA_ERROR); - goto on_error3; - } +/****************************************************************************** - error = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pDevice->pulse.pMainLoop, 1, NULL); - if (error < 0) { - result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] The PulseAudio main loop returned an error while connecting the PulseAudio context.", ma_result_from_pulse(error)); - goto on_error3; - } - } +PulseAudio Backend - if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { - pOP = ((ma_pa_context_get_source_info_by_name_proc)pContext->pulse.pa_context_get_source_info_by_name)((ma_pa_context*)pDevice->pulse.pPulseContext, devCapture, ma_device_source_info_callback, &sourceInfo); - if (pOP != NULL) { - ma_device__wait_for_operation__pulse(pDevice, pOP); - ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); - } else { - result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to retrieve source info for capture device.", ma_result_from_pulse(error)); - goto on_error3; - } +******************************************************************************/ +#ifdef MA_HAS_PULSEAUDIO +/* +It is assumed pulseaudio.h is available when compile-time linking is being used. We use this for type safety when using +compile time linking (we don't have this luxury when using runtime linking without headers). - ss = sourceInfo.sample_spec; - cmap = sourceInfo.channel_map; +When using compile time linking, each of our ma_* equivalents should use the sames types as defined by the header. The +reason for this is that it allow us to take advantage of proper type safety. +*/ +#ifdef MA_NO_RUNTIME_LINKING +#include - pDevice->capture.internalBufferSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(bufferSizeInMilliseconds, ss.rate); - pDevice->capture.internalPeriods = pConfig->periods; +#define MA_PA_OK PA_OK +#define MA_PA_ERR_ACCESS PA_ERR_ACCESS +#define MA_PA_ERR_INVALID PA_ERR_INVALID +#define MA_PA_ERR_NOENTITY PA_ERR_NOENTITY - attr = ma_device__pa_buffer_attr_new(pDevice->capture.internalBufferSizeInFrames, pConfig->periods, &ss); - #ifdef MA_DEBUG_OUTPUT - printf("[PulseAudio] Capture attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; internalBufferSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDevice->capture.internalBufferSizeInFrames); - #endif +#define MA_PA_CHANNELS_MAX PA_CHANNELS_MAX +#define MA_PA_RATE_MAX PA_RATE_MAX - pDevice->pulse.pStreamCapture = ma_device__pa_stream_new__pulse(pDevice, pConfig->pulse.pStreamNameCapture, &ss, &cmap); - if (pDevice->pulse.pStreamCapture == NULL) { - result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create PulseAudio capture stream.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - goto on_error3; - } +typedef pa_context_flags_t ma_pa_context_flags_t; +#define MA_PA_CONTEXT_NOFLAGS PA_CONTEXT_NOFLAGS +#define MA_PA_CONTEXT_NOAUTOSPAWN PA_CONTEXT_NOAUTOSPAWN +#define MA_PA_CONTEXT_NOFAIL PA_CONTEXT_NOFAIL - streamFlags = MA_PA_STREAM_START_CORKED | MA_PA_STREAM_FIX_FORMAT | MA_PA_STREAM_FIX_RATE | MA_PA_STREAM_FIX_CHANNELS; - if (devCapture != NULL) { - streamFlags |= MA_PA_STREAM_DONT_MOVE; - } +typedef pa_stream_flags_t ma_pa_stream_flags_t; +#define MA_PA_STREAM_NOFLAGS PA_STREAM_NOFLAGS +#define MA_PA_STREAM_START_CORKED PA_STREAM_START_CORKED +#define MA_PA_STREAM_INTERPOLATE_TIMING PA_STREAM_INTERPOLATE_TIMING +#define MA_PA_STREAM_NOT_MONOTONIC PA_STREAM_NOT_MONOTONIC +#define MA_PA_STREAM_AUTO_TIMING_UPDATE PA_STREAM_AUTO_TIMING_UPDATE +#define MA_PA_STREAM_NO_REMAP_CHANNELS PA_STREAM_NO_REMAP_CHANNELS +#define MA_PA_STREAM_NO_REMIX_CHANNELS PA_STREAM_NO_REMIX_CHANNELS +#define MA_PA_STREAM_FIX_FORMAT PA_STREAM_FIX_FORMAT +#define MA_PA_STREAM_FIX_RATE PA_STREAM_FIX_RATE +#define MA_PA_STREAM_FIX_CHANNELS PA_STREAM_FIX_CHANNELS +#define MA_PA_STREAM_DONT_MOVE PA_STREAM_DONT_MOVE +#define MA_PA_STREAM_VARIABLE_RATE PA_STREAM_VARIABLE_RATE +#define MA_PA_STREAM_PEAK_DETECT PA_STREAM_PEAK_DETECT +#define MA_PA_STREAM_START_MUTED PA_STREAM_START_MUTED +#define MA_PA_STREAM_ADJUST_LATENCY PA_STREAM_ADJUST_LATENCY +#define MA_PA_STREAM_EARLY_REQUESTS PA_STREAM_EARLY_REQUESTS +#define MA_PA_STREAM_DONT_INHIBIT_AUTO_SUSPEND PA_STREAM_DONT_INHIBIT_AUTO_SUSPEND +#define MA_PA_STREAM_START_UNMUTED PA_STREAM_START_UNMUTED +#define MA_PA_STREAM_FAIL_ON_SUSPEND PA_STREAM_FAIL_ON_SUSPEND +#define MA_PA_STREAM_RELATIVE_VOLUME PA_STREAM_RELATIVE_VOLUME +#define MA_PA_STREAM_PASSTHROUGH PA_STREAM_PASSTHROUGH - error = ((ma_pa_stream_connect_record_proc)pContext->pulse.pa_stream_connect_record)((ma_pa_stream*)pDevice->pulse.pStreamCapture, devCapture, &attr, streamFlags); - if (error != MA_PA_OK) { - result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to connect PulseAudio capture stream.", ma_result_from_pulse(error)); - goto on_error4; - } +typedef pa_sink_flags_t ma_pa_sink_flags_t; +#define MA_PA_SINK_NOFLAGS PA_SINK_NOFLAGS +#define MA_PA_SINK_HW_VOLUME_CTRL PA_SINK_HW_VOLUME_CTRL +#define MA_PA_SINK_LATENCY PA_SINK_LATENCY +#define MA_PA_SINK_HARDWARE PA_SINK_HARDWARE +#define MA_PA_SINK_NETWORK PA_SINK_NETWORK +#define MA_PA_SINK_HW_MUTE_CTRL PA_SINK_HW_MUTE_CTRL +#define MA_PA_SINK_DECIBEL_VOLUME PA_SINK_DECIBEL_VOLUME +#define MA_PA_SINK_FLAT_VOLUME PA_SINK_FLAT_VOLUME +#define MA_PA_SINK_DYNAMIC_LATENCY PA_SINK_DYNAMIC_LATENCY +#define MA_PA_SINK_SET_FORMATS PA_SINK_SET_FORMATS - while (((ma_pa_stream_get_state_proc)pContext->pulse.pa_stream_get_state)((ma_pa_stream*)pDevice->pulse.pStreamCapture) != MA_PA_STREAM_READY) { - error = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pDevice->pulse.pMainLoop, 1, NULL); - if (error < 0) { - result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] The PulseAudio main loop returned an error while connecting the PulseAudio capture stream.", ma_result_from_pulse(error)); - goto on_error5; - } - } +typedef pa_source_flags_t ma_pa_source_flags_t; +#define MA_PA_SOURCE_NOFLAGS PA_SOURCE_NOFLAGS +#define MA_PA_SOURCE_HW_VOLUME_CTRL PA_SOURCE_HW_VOLUME_CTRL +#define MA_PA_SOURCE_LATENCY PA_SOURCE_LATENCY +#define MA_PA_SOURCE_HARDWARE PA_SOURCE_HARDWARE +#define MA_PA_SOURCE_NETWORK PA_SOURCE_NETWORK +#define MA_PA_SOURCE_HW_MUTE_CTRL PA_SOURCE_HW_MUTE_CTRL +#define MA_PA_SOURCE_DECIBEL_VOLUME PA_SOURCE_DECIBEL_VOLUME +#define MA_PA_SOURCE_DYNAMIC_LATENCY PA_SOURCE_DYNAMIC_LATENCY +#define MA_PA_SOURCE_FLAT_VOLUME PA_SOURCE_FLAT_VOLUME - /* Internal format. */ - pActualSS = ((ma_pa_stream_get_sample_spec_proc)pContext->pulse.pa_stream_get_sample_spec)((ma_pa_stream*)pDevice->pulse.pStreamCapture); - if (pActualSS != NULL) { - /* If anything has changed between the requested and the actual sample spec, we need to update the buffer. */ - if (ss.format != pActualSS->format || ss.channels != pActualSS->channels || ss.rate != pActualSS->rate) { - attr = ma_device__pa_buffer_attr_new(pDevice->capture.internalBufferSizeInFrames, pConfig->periods, pActualSS); +typedef pa_context_state_t ma_pa_context_state_t; +#define MA_PA_CONTEXT_UNCONNECTED PA_CONTEXT_UNCONNECTED +#define MA_PA_CONTEXT_CONNECTING PA_CONTEXT_CONNECTING +#define MA_PA_CONTEXT_AUTHORIZING PA_CONTEXT_AUTHORIZING +#define MA_PA_CONTEXT_SETTING_NAME PA_CONTEXT_SETTING_NAME +#define MA_PA_CONTEXT_READY PA_CONTEXT_READY +#define MA_PA_CONTEXT_FAILED PA_CONTEXT_FAILED +#define MA_PA_CONTEXT_TERMINATED PA_CONTEXT_TERMINATED - pOP = ((ma_pa_stream_set_buffer_attr_proc)pContext->pulse.pa_stream_set_buffer_attr)((ma_pa_stream*)pDevice->pulse.pStreamCapture, &attr, NULL, NULL); - if (pOP != NULL) { - ma_device__wait_for_operation__pulse(pDevice, pOP); - ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); - } - } +typedef pa_stream_state_t ma_pa_stream_state_t; +#define MA_PA_STREAM_UNCONNECTED PA_STREAM_UNCONNECTED +#define MA_PA_STREAM_CREATING PA_STREAM_CREATING +#define MA_PA_STREAM_READY PA_STREAM_READY +#define MA_PA_STREAM_FAILED PA_STREAM_FAILED +#define MA_PA_STREAM_TERMINATED PA_STREAM_TERMINATED - ss = *pActualSS; - } +typedef pa_operation_state_t ma_pa_operation_state_t; +#define MA_PA_OPERATION_RUNNING PA_OPERATION_RUNNING +#define MA_PA_OPERATION_DONE PA_OPERATION_DONE +#define MA_PA_OPERATION_CANCELLED PA_OPERATION_CANCELLED - pDevice->capture.internalFormat = ma_format_from_pulse(ss.format); - pDevice->capture.internalChannels = ss.channels; - pDevice->capture.internalSampleRate = ss.rate; +typedef pa_sink_state_t ma_pa_sink_state_t; +#define MA_PA_SINK_INVALID_STATE PA_SINK_INVALID_STATE +#define MA_PA_SINK_RUNNING PA_SINK_RUNNING +#define MA_PA_SINK_IDLE PA_SINK_IDLE +#define MA_PA_SINK_SUSPENDED PA_SINK_SUSPENDED - /* Internal channel map. */ - pActualCMap = ((ma_pa_stream_get_channel_map_proc)pContext->pulse.pa_stream_get_channel_map)((ma_pa_stream*)pDevice->pulse.pStreamCapture); - if (pActualCMap != NULL) { - cmap = *pActualCMap; - } - for (iChannel = 0; iChannel < pDevice->capture.internalChannels; ++iChannel) { - pDevice->capture.internalChannelMap[iChannel] = ma_channel_position_from_pulse(cmap.map[iChannel]); - } +typedef pa_source_state_t ma_pa_source_state_t; +#define MA_PA_SOURCE_INVALID_STATE PA_SOURCE_INVALID_STATE +#define MA_PA_SOURCE_RUNNING PA_SOURCE_RUNNING +#define MA_PA_SOURCE_IDLE PA_SOURCE_IDLE +#define MA_PA_SOURCE_SUSPENDED PA_SOURCE_SUSPENDED - /* Buffer. */ - pActualAttr = ((ma_pa_stream_get_buffer_attr_proc)pContext->pulse.pa_stream_get_buffer_attr)((ma_pa_stream*)pDevice->pulse.pStreamCapture); - if (pActualAttr != NULL) { - attr = *pActualAttr; - } - pDevice->capture.internalBufferSizeInFrames = attr.maxlength / (ma_get_bytes_per_sample(pDevice->capture.internalFormat) * pDevice->capture.internalChannels); - pDevice->capture.internalPeriods = attr.maxlength / attr.fragsize; - #ifdef MA_DEBUG_OUTPUT - printf("[PulseAudio] Capture actual attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; internalBufferSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDevice->capture.internalBufferSizeInFrames); - #endif +typedef pa_seek_mode_t ma_pa_seek_mode_t; +#define MA_PA_SEEK_RELATIVE PA_SEEK_RELATIVE +#define MA_PA_SEEK_ABSOLUTE PA_SEEK_ABSOLUTE +#define MA_PA_SEEK_RELATIVE_ON_READ PA_SEEK_RELATIVE_ON_READ +#define MA_PA_SEEK_RELATIVE_END PA_SEEK_RELATIVE_END - /* Name. */ - devCapture = ((ma_pa_stream_get_device_name_proc)pContext->pulse.pa_stream_get_device_name)((ma_pa_stream*)pDevice->pulse.pStreamCapture); - if (devCapture != NULL) { - ma_pa_operation* pOP = ((ma_pa_context_get_source_info_by_name_proc)pContext->pulse.pa_context_get_source_info_by_name)((ma_pa_context*)pDevice->pulse.pPulseContext, devCapture, ma_device_source_name_callback, pDevice); - if (pOP != NULL) { - ma_device__wait_for_operation__pulse(pDevice, pOP); - ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); - } - } - } +typedef pa_channel_position_t ma_pa_channel_position_t; +#define MA_PA_CHANNEL_POSITION_INVALID PA_CHANNEL_POSITION_INVALID +#define MA_PA_CHANNEL_POSITION_MONO PA_CHANNEL_POSITION_MONO +#define MA_PA_CHANNEL_POSITION_FRONT_LEFT PA_CHANNEL_POSITION_FRONT_LEFT +#define MA_PA_CHANNEL_POSITION_FRONT_RIGHT PA_CHANNEL_POSITION_FRONT_RIGHT +#define MA_PA_CHANNEL_POSITION_FRONT_CENTER PA_CHANNEL_POSITION_FRONT_CENTER +#define MA_PA_CHANNEL_POSITION_REAR_CENTER PA_CHANNEL_POSITION_REAR_CENTER +#define MA_PA_CHANNEL_POSITION_REAR_LEFT PA_CHANNEL_POSITION_REAR_LEFT +#define MA_PA_CHANNEL_POSITION_REAR_RIGHT PA_CHANNEL_POSITION_REAR_RIGHT +#define MA_PA_CHANNEL_POSITION_LFE PA_CHANNEL_POSITION_LFE +#define MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER +#define MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER +#define MA_PA_CHANNEL_POSITION_SIDE_LEFT PA_CHANNEL_POSITION_SIDE_LEFT +#define MA_PA_CHANNEL_POSITION_SIDE_RIGHT PA_CHANNEL_POSITION_SIDE_RIGHT +#define MA_PA_CHANNEL_POSITION_AUX0 PA_CHANNEL_POSITION_AUX0 +#define MA_PA_CHANNEL_POSITION_AUX1 PA_CHANNEL_POSITION_AUX1 +#define MA_PA_CHANNEL_POSITION_AUX2 PA_CHANNEL_POSITION_AUX2 +#define MA_PA_CHANNEL_POSITION_AUX3 PA_CHANNEL_POSITION_AUX3 +#define MA_PA_CHANNEL_POSITION_AUX4 PA_CHANNEL_POSITION_AUX4 +#define MA_PA_CHANNEL_POSITION_AUX5 PA_CHANNEL_POSITION_AUX5 +#define MA_PA_CHANNEL_POSITION_AUX6 PA_CHANNEL_POSITION_AUX6 +#define MA_PA_CHANNEL_POSITION_AUX7 PA_CHANNEL_POSITION_AUX7 +#define MA_PA_CHANNEL_POSITION_AUX8 PA_CHANNEL_POSITION_AUX8 +#define MA_PA_CHANNEL_POSITION_AUX9 PA_CHANNEL_POSITION_AUX9 +#define MA_PA_CHANNEL_POSITION_AUX10 PA_CHANNEL_POSITION_AUX10 +#define MA_PA_CHANNEL_POSITION_AUX11 PA_CHANNEL_POSITION_AUX11 +#define MA_PA_CHANNEL_POSITION_AUX12 PA_CHANNEL_POSITION_AUX12 +#define MA_PA_CHANNEL_POSITION_AUX13 PA_CHANNEL_POSITION_AUX13 +#define MA_PA_CHANNEL_POSITION_AUX14 PA_CHANNEL_POSITION_AUX14 +#define MA_PA_CHANNEL_POSITION_AUX15 PA_CHANNEL_POSITION_AUX15 +#define MA_PA_CHANNEL_POSITION_AUX16 PA_CHANNEL_POSITION_AUX16 +#define MA_PA_CHANNEL_POSITION_AUX17 PA_CHANNEL_POSITION_AUX17 +#define MA_PA_CHANNEL_POSITION_AUX18 PA_CHANNEL_POSITION_AUX18 +#define MA_PA_CHANNEL_POSITION_AUX19 PA_CHANNEL_POSITION_AUX19 +#define MA_PA_CHANNEL_POSITION_AUX20 PA_CHANNEL_POSITION_AUX20 +#define MA_PA_CHANNEL_POSITION_AUX21 PA_CHANNEL_POSITION_AUX21 +#define MA_PA_CHANNEL_POSITION_AUX22 PA_CHANNEL_POSITION_AUX22 +#define MA_PA_CHANNEL_POSITION_AUX23 PA_CHANNEL_POSITION_AUX23 +#define MA_PA_CHANNEL_POSITION_AUX24 PA_CHANNEL_POSITION_AUX24 +#define MA_PA_CHANNEL_POSITION_AUX25 PA_CHANNEL_POSITION_AUX25 +#define MA_PA_CHANNEL_POSITION_AUX26 PA_CHANNEL_POSITION_AUX26 +#define MA_PA_CHANNEL_POSITION_AUX27 PA_CHANNEL_POSITION_AUX27 +#define MA_PA_CHANNEL_POSITION_AUX28 PA_CHANNEL_POSITION_AUX28 +#define MA_PA_CHANNEL_POSITION_AUX29 PA_CHANNEL_POSITION_AUX29 +#define MA_PA_CHANNEL_POSITION_AUX30 PA_CHANNEL_POSITION_AUX30 +#define MA_PA_CHANNEL_POSITION_AUX31 PA_CHANNEL_POSITION_AUX31 +#define MA_PA_CHANNEL_POSITION_TOP_CENTER PA_CHANNEL_POSITION_TOP_CENTER +#define MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT PA_CHANNEL_POSITION_TOP_FRONT_LEFT +#define MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT PA_CHANNEL_POSITION_TOP_FRONT_RIGHT +#define MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER PA_CHANNEL_POSITION_TOP_FRONT_CENTER +#define MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT PA_CHANNEL_POSITION_TOP_REAR_LEFT +#define MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT PA_CHANNEL_POSITION_TOP_REAR_RIGHT +#define MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER PA_CHANNEL_POSITION_TOP_REAR_CENTER +#define MA_PA_CHANNEL_POSITION_LEFT PA_CHANNEL_POSITION_LEFT +#define MA_PA_CHANNEL_POSITION_RIGHT PA_CHANNEL_POSITION_RIGHT +#define MA_PA_CHANNEL_POSITION_CENTER PA_CHANNEL_POSITION_CENTER +#define MA_PA_CHANNEL_POSITION_SUBWOOFER PA_CHANNEL_POSITION_SUBWOOFER - if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { - pOP = ((ma_pa_context_get_sink_info_by_name_proc)pContext->pulse.pa_context_get_sink_info_by_name)((ma_pa_context*)pDevice->pulse.pPulseContext, devPlayback, ma_device_sink_info_callback, &sinkInfo); - if (pOP != NULL) { - ma_device__wait_for_operation__pulse(pDevice, pOP); - ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); - } else { - result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to retrieve sink info for playback device.", ma_result_from_pulse(error)); - goto on_error3; - } +typedef pa_channel_map_def_t ma_pa_channel_map_def_t; +#define MA_PA_CHANNEL_MAP_AIFF PA_CHANNEL_MAP_AIFF +#define MA_PA_CHANNEL_MAP_ALSA PA_CHANNEL_MAP_ALSA +#define MA_PA_CHANNEL_MAP_AUX PA_CHANNEL_MAP_AUX +#define MA_PA_CHANNEL_MAP_WAVEEX PA_CHANNEL_MAP_WAVEEX +#define MA_PA_CHANNEL_MAP_OSS PA_CHANNEL_MAP_OSS +#define MA_PA_CHANNEL_MAP_DEFAULT PA_CHANNEL_MAP_DEFAULT - ss = sinkInfo.sample_spec; - cmap = sinkInfo.channel_map; +typedef pa_sample_format_t ma_pa_sample_format_t; +#define MA_PA_SAMPLE_INVALID PA_SAMPLE_INVALID +#define MA_PA_SAMPLE_U8 PA_SAMPLE_U8 +#define MA_PA_SAMPLE_ALAW PA_SAMPLE_ALAW +#define MA_PA_SAMPLE_ULAW PA_SAMPLE_ULAW +#define MA_PA_SAMPLE_S16LE PA_SAMPLE_S16LE +#define MA_PA_SAMPLE_S16BE PA_SAMPLE_S16BE +#define MA_PA_SAMPLE_FLOAT32LE PA_SAMPLE_FLOAT32LE +#define MA_PA_SAMPLE_FLOAT32BE PA_SAMPLE_FLOAT32BE +#define MA_PA_SAMPLE_S32LE PA_SAMPLE_S32LE +#define MA_PA_SAMPLE_S32BE PA_SAMPLE_S32BE +#define MA_PA_SAMPLE_S24LE PA_SAMPLE_S24LE +#define MA_PA_SAMPLE_S24BE PA_SAMPLE_S24BE +#define MA_PA_SAMPLE_S24_32LE PA_SAMPLE_S24_32LE +#define MA_PA_SAMPLE_S24_32BE PA_SAMPLE_S24_32BE - pDevice->playback.internalBufferSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(bufferSizeInMilliseconds, ss.rate); - pDevice->playback.internalPeriods = pConfig->periods; +typedef pa_mainloop ma_pa_mainloop; +typedef pa_mainloop_api ma_pa_mainloop_api; +typedef pa_context ma_pa_context; +typedef pa_operation ma_pa_operation; +typedef pa_stream ma_pa_stream; +typedef pa_spawn_api ma_pa_spawn_api; +typedef pa_buffer_attr ma_pa_buffer_attr; +typedef pa_channel_map ma_pa_channel_map; +typedef pa_cvolume ma_pa_cvolume; +typedef pa_sample_spec ma_pa_sample_spec; +typedef pa_sink_info ma_pa_sink_info; +typedef pa_source_info ma_pa_source_info; - attr = ma_device__pa_buffer_attr_new(pDevice->playback.internalBufferSizeInFrames, pConfig->periods, &ss); - #ifdef MA_DEBUG_OUTPUT - printf("[PulseAudio] Playback attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; internalBufferSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDevice->playback.internalBufferSizeInFrames); - #endif +typedef pa_context_notify_cb_t ma_pa_context_notify_cb_t; +typedef pa_sink_info_cb_t ma_pa_sink_info_cb_t; +typedef pa_source_info_cb_t ma_pa_source_info_cb_t; +typedef pa_stream_success_cb_t ma_pa_stream_success_cb_t; +typedef pa_stream_request_cb_t ma_pa_stream_request_cb_t; +typedef pa_free_cb_t ma_pa_free_cb_t; +#else +#define MA_PA_OK 0 +#define MA_PA_ERR_ACCESS 1 +#define MA_PA_ERR_INVALID 2 +#define MA_PA_ERR_NOENTITY 5 - pDevice->pulse.pStreamPlayback = ma_device__pa_stream_new__pulse(pDevice, pConfig->pulse.pStreamNamePlayback, &ss, &cmap); - if (pDevice->pulse.pStreamPlayback == NULL) { - result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create PulseAudio playback stream.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - goto on_error3; - } +#define MA_PA_CHANNELS_MAX 32 +#define MA_PA_RATE_MAX 384000 - streamFlags = MA_PA_STREAM_START_CORKED | MA_PA_STREAM_FIX_FORMAT | MA_PA_STREAM_FIX_RATE | MA_PA_STREAM_FIX_CHANNELS; - if (devPlayback != NULL) { - streamFlags |= MA_PA_STREAM_DONT_MOVE; - } +typedef int ma_pa_context_flags_t; +#define MA_PA_CONTEXT_NOFLAGS 0x00000000 +#define MA_PA_CONTEXT_NOAUTOSPAWN 0x00000001 +#define MA_PA_CONTEXT_NOFAIL 0x00000002 - error = ((ma_pa_stream_connect_playback_proc)pContext->pulse.pa_stream_connect_playback)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, devPlayback, &attr, streamFlags, NULL, NULL); - if (error != MA_PA_OK) { - result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to connect PulseAudio playback stream.", ma_result_from_pulse(error)); - goto on_error6; - } +typedef int ma_pa_stream_flags_t; +#define MA_PA_STREAM_NOFLAGS 0x00000000 +#define MA_PA_STREAM_START_CORKED 0x00000001 +#define MA_PA_STREAM_INTERPOLATE_TIMING 0x00000002 +#define MA_PA_STREAM_NOT_MONOTONIC 0x00000004 +#define MA_PA_STREAM_AUTO_TIMING_UPDATE 0x00000008 +#define MA_PA_STREAM_NO_REMAP_CHANNELS 0x00000010 +#define MA_PA_STREAM_NO_REMIX_CHANNELS 0x00000020 +#define MA_PA_STREAM_FIX_FORMAT 0x00000040 +#define MA_PA_STREAM_FIX_RATE 0x00000080 +#define MA_PA_STREAM_FIX_CHANNELS 0x00000100 +#define MA_PA_STREAM_DONT_MOVE 0x00000200 +#define MA_PA_STREAM_VARIABLE_RATE 0x00000400 +#define MA_PA_STREAM_PEAK_DETECT 0x00000800 +#define MA_PA_STREAM_START_MUTED 0x00001000 +#define MA_PA_STREAM_ADJUST_LATENCY 0x00002000 +#define MA_PA_STREAM_EARLY_REQUESTS 0x00004000 +#define MA_PA_STREAM_DONT_INHIBIT_AUTO_SUSPEND 0x00008000 +#define MA_PA_STREAM_START_UNMUTED 0x00010000 +#define MA_PA_STREAM_FAIL_ON_SUSPEND 0x00020000 +#define MA_PA_STREAM_RELATIVE_VOLUME 0x00040000 +#define MA_PA_STREAM_PASSTHROUGH 0x00080000 - while (((ma_pa_stream_get_state_proc)pContext->pulse.pa_stream_get_state)((ma_pa_stream*)pDevice->pulse.pStreamPlayback) != MA_PA_STREAM_READY) { - error = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pDevice->pulse.pMainLoop, 1, NULL); - if (error < 0) { - result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] The PulseAudio main loop returned an error while connecting the PulseAudio playback stream.", ma_result_from_pulse(error)); - goto on_error7; - } - } +typedef int ma_pa_sink_flags_t; +#define MA_PA_SINK_NOFLAGS 0x00000000 +#define MA_PA_SINK_HW_VOLUME_CTRL 0x00000001 +#define MA_PA_SINK_LATENCY 0x00000002 +#define MA_PA_SINK_HARDWARE 0x00000004 +#define MA_PA_SINK_NETWORK 0x00000008 +#define MA_PA_SINK_HW_MUTE_CTRL 0x00000010 +#define MA_PA_SINK_DECIBEL_VOLUME 0x00000020 +#define MA_PA_SINK_FLAT_VOLUME 0x00000040 +#define MA_PA_SINK_DYNAMIC_LATENCY 0x00000080 +#define MA_PA_SINK_SET_FORMATS 0x00000100 - /* Internal format. */ - pActualSS = ((ma_pa_stream_get_sample_spec_proc)pContext->pulse.pa_stream_get_sample_spec)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); - if (pActualSS != NULL) { - /* If anything has changed between the requested and the actual sample spec, we need to update the buffer. */ - if (ss.format != pActualSS->format || ss.channels != pActualSS->channels || ss.rate != pActualSS->rate) { - attr = ma_device__pa_buffer_attr_new(pDevice->playback.internalBufferSizeInFrames, pConfig->periods, pActualSS); +typedef int ma_pa_source_flags_t; +#define MA_PA_SOURCE_NOFLAGS 0x00000000 +#define MA_PA_SOURCE_HW_VOLUME_CTRL 0x00000001 +#define MA_PA_SOURCE_LATENCY 0x00000002 +#define MA_PA_SOURCE_HARDWARE 0x00000004 +#define MA_PA_SOURCE_NETWORK 0x00000008 +#define MA_PA_SOURCE_HW_MUTE_CTRL 0x00000010 +#define MA_PA_SOURCE_DECIBEL_VOLUME 0x00000020 +#define MA_PA_SOURCE_DYNAMIC_LATENCY 0x00000040 +#define MA_PA_SOURCE_FLAT_VOLUME 0x00000080 - pOP = ((ma_pa_stream_set_buffer_attr_proc)pContext->pulse.pa_stream_set_buffer_attr)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, &attr, NULL, NULL); - if (pOP != NULL) { - ma_device__wait_for_operation__pulse(pDevice, pOP); - ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); - } - } +typedef int ma_pa_context_state_t; +#define MA_PA_CONTEXT_UNCONNECTED 0 +#define MA_PA_CONTEXT_CONNECTING 1 +#define MA_PA_CONTEXT_AUTHORIZING 2 +#define MA_PA_CONTEXT_SETTING_NAME 3 +#define MA_PA_CONTEXT_READY 4 +#define MA_PA_CONTEXT_FAILED 5 +#define MA_PA_CONTEXT_TERMINATED 6 - ss = *pActualSS; - } +typedef int ma_pa_stream_state_t; +#define MA_PA_STREAM_UNCONNECTED 0 +#define MA_PA_STREAM_CREATING 1 +#define MA_PA_STREAM_READY 2 +#define MA_PA_STREAM_FAILED 3 +#define MA_PA_STREAM_TERMINATED 4 - pDevice->playback.internalFormat = ma_format_from_pulse(ss.format); - pDevice->playback.internalChannels = ss.channels; - pDevice->playback.internalSampleRate = ss.rate; +typedef int ma_pa_operation_state_t; +#define MA_PA_OPERATION_RUNNING 0 +#define MA_PA_OPERATION_DONE 1 +#define MA_PA_OPERATION_CANCELLED 2 - /* Internal channel map. */ - pActualCMap = ((ma_pa_stream_get_channel_map_proc)pContext->pulse.pa_stream_get_channel_map)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); - if (pActualCMap != NULL) { - cmap = *pActualCMap; - } - for (iChannel = 0; iChannel < pDevice->playback.internalChannels; ++iChannel) { - pDevice->playback.internalChannelMap[iChannel] = ma_channel_position_from_pulse(cmap.map[iChannel]); - } +typedef int ma_pa_sink_state_t; +#define MA_PA_SINK_INVALID_STATE -1 +#define MA_PA_SINK_RUNNING 0 +#define MA_PA_SINK_IDLE 1 +#define MA_PA_SINK_SUSPENDED 2 - /* Buffer. */ - pActualAttr = ((ma_pa_stream_get_buffer_attr_proc)pContext->pulse.pa_stream_get_buffer_attr)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); - if (pActualAttr != NULL) { - attr = *pActualAttr; - } - pDevice->playback.internalBufferSizeInFrames = attr.maxlength / (ma_get_bytes_per_sample(pDevice->playback.internalFormat) * pDevice->playback.internalChannels); - pDevice->playback.internalPeriods = /*pConfig->periods;*/attr.maxlength / attr.tlength; - #ifdef MA_DEBUG_OUTPUT - printf("[PulseAudio] Playback actual attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; internalBufferSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDevice->playback.internalBufferSizeInFrames); - #endif +typedef int ma_pa_source_state_t; +#define MA_PA_SOURCE_INVALID_STATE -1 +#define MA_PA_SOURCE_RUNNING 0 +#define MA_PA_SOURCE_IDLE 1 +#define MA_PA_SOURCE_SUSPENDED 2 - /* Name. */ - devPlayback = ((ma_pa_stream_get_device_name_proc)pContext->pulse.pa_stream_get_device_name)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); - if (devPlayback != NULL) { - ma_pa_operation* pOP = ((ma_pa_context_get_sink_info_by_name_proc)pContext->pulse.pa_context_get_sink_info_by_name)((ma_pa_context*)pDevice->pulse.pPulseContext, devPlayback, ma_device_sink_name_callback, pDevice); - if (pOP != NULL) { - ma_device__wait_for_operation__pulse(pDevice, pOP); - ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); - } - } - } +typedef int ma_pa_seek_mode_t; +#define MA_PA_SEEK_RELATIVE 0 +#define MA_PA_SEEK_ABSOLUTE 1 +#define MA_PA_SEEK_RELATIVE_ON_READ 2 +#define MA_PA_SEEK_RELATIVE_END 3 - return MA_SUCCESS; +typedef int ma_pa_channel_position_t; +#define MA_PA_CHANNEL_POSITION_INVALID -1 +#define MA_PA_CHANNEL_POSITION_MONO 0 +#define MA_PA_CHANNEL_POSITION_FRONT_LEFT 1 +#define MA_PA_CHANNEL_POSITION_FRONT_RIGHT 2 +#define MA_PA_CHANNEL_POSITION_FRONT_CENTER 3 +#define MA_PA_CHANNEL_POSITION_REAR_CENTER 4 +#define MA_PA_CHANNEL_POSITION_REAR_LEFT 5 +#define MA_PA_CHANNEL_POSITION_REAR_RIGHT 6 +#define MA_PA_CHANNEL_POSITION_LFE 7 +#define MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER 8 +#define MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER 9 +#define MA_PA_CHANNEL_POSITION_SIDE_LEFT 10 +#define MA_PA_CHANNEL_POSITION_SIDE_RIGHT 11 +#define MA_PA_CHANNEL_POSITION_AUX0 12 +#define MA_PA_CHANNEL_POSITION_AUX1 13 +#define MA_PA_CHANNEL_POSITION_AUX2 14 +#define MA_PA_CHANNEL_POSITION_AUX3 15 +#define MA_PA_CHANNEL_POSITION_AUX4 16 +#define MA_PA_CHANNEL_POSITION_AUX5 17 +#define MA_PA_CHANNEL_POSITION_AUX6 18 +#define MA_PA_CHANNEL_POSITION_AUX7 19 +#define MA_PA_CHANNEL_POSITION_AUX8 20 +#define MA_PA_CHANNEL_POSITION_AUX9 21 +#define MA_PA_CHANNEL_POSITION_AUX10 22 +#define MA_PA_CHANNEL_POSITION_AUX11 23 +#define MA_PA_CHANNEL_POSITION_AUX12 24 +#define MA_PA_CHANNEL_POSITION_AUX13 25 +#define MA_PA_CHANNEL_POSITION_AUX14 26 +#define MA_PA_CHANNEL_POSITION_AUX15 27 +#define MA_PA_CHANNEL_POSITION_AUX16 28 +#define MA_PA_CHANNEL_POSITION_AUX17 29 +#define MA_PA_CHANNEL_POSITION_AUX18 30 +#define MA_PA_CHANNEL_POSITION_AUX19 31 +#define MA_PA_CHANNEL_POSITION_AUX20 32 +#define MA_PA_CHANNEL_POSITION_AUX21 33 +#define MA_PA_CHANNEL_POSITION_AUX22 34 +#define MA_PA_CHANNEL_POSITION_AUX23 35 +#define MA_PA_CHANNEL_POSITION_AUX24 36 +#define MA_PA_CHANNEL_POSITION_AUX25 37 +#define MA_PA_CHANNEL_POSITION_AUX26 38 +#define MA_PA_CHANNEL_POSITION_AUX27 39 +#define MA_PA_CHANNEL_POSITION_AUX28 40 +#define MA_PA_CHANNEL_POSITION_AUX29 41 +#define MA_PA_CHANNEL_POSITION_AUX30 42 +#define MA_PA_CHANNEL_POSITION_AUX31 43 +#define MA_PA_CHANNEL_POSITION_TOP_CENTER 44 +#define MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT 45 +#define MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT 46 +#define MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER 47 +#define MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT 48 +#define MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT 49 +#define MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER 50 +#define MA_PA_CHANNEL_POSITION_LEFT MA_PA_CHANNEL_POSITION_FRONT_LEFT +#define MA_PA_CHANNEL_POSITION_RIGHT MA_PA_CHANNEL_POSITION_FRONT_RIGHT +#define MA_PA_CHANNEL_POSITION_CENTER MA_PA_CHANNEL_POSITION_FRONT_CENTER +#define MA_PA_CHANNEL_POSITION_SUBWOOFER MA_PA_CHANNEL_POSITION_LFE +typedef int ma_pa_channel_map_def_t; +#define MA_PA_CHANNEL_MAP_AIFF 0 +#define MA_PA_CHANNEL_MAP_ALSA 1 +#define MA_PA_CHANNEL_MAP_AUX 2 +#define MA_PA_CHANNEL_MAP_WAVEEX 3 +#define MA_PA_CHANNEL_MAP_OSS 4 +#define MA_PA_CHANNEL_MAP_DEFAULT MA_PA_CHANNEL_MAP_AIFF -on_error7: - if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { - ((ma_pa_stream_disconnect_proc)pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); - } -on_error6: - if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { - ((ma_pa_stream_unref_proc)pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); - } -on_error5: - if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { - ((ma_pa_stream_disconnect_proc)pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamCapture); - } -on_error4: - if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { - ((ma_pa_stream_unref_proc)pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamCapture); - } -on_error3: ((ma_pa_context_disconnect_proc)pContext->pulse.pa_context_disconnect)((ma_pa_context*)pDevice->pulse.pPulseContext); -on_error2: ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)((ma_pa_context*)pDevice->pulse.pPulseContext); -on_error1: ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)pDevice->pulse.pMainLoop); -on_error0: - return result; -} +typedef int ma_pa_sample_format_t; +#define MA_PA_SAMPLE_INVALID -1 +#define MA_PA_SAMPLE_U8 0 +#define MA_PA_SAMPLE_ALAW 1 +#define MA_PA_SAMPLE_ULAW 2 +#define MA_PA_SAMPLE_S16LE 3 +#define MA_PA_SAMPLE_S16BE 4 +#define MA_PA_SAMPLE_FLOAT32LE 5 +#define MA_PA_SAMPLE_FLOAT32BE 6 +#define MA_PA_SAMPLE_S32LE 7 +#define MA_PA_SAMPLE_S32BE 8 +#define MA_PA_SAMPLE_S24LE 9 +#define MA_PA_SAMPLE_S24BE 10 +#define MA_PA_SAMPLE_S24_32LE 11 +#define MA_PA_SAMPLE_S24_32BE 12 +typedef struct ma_pa_mainloop ma_pa_mainloop; +typedef struct ma_pa_mainloop_api ma_pa_mainloop_api; +typedef struct ma_pa_context ma_pa_context; +typedef struct ma_pa_operation ma_pa_operation; +typedef struct ma_pa_stream ma_pa_stream; +typedef struct ma_pa_spawn_api ma_pa_spawn_api; -void ma_pulse_operation_complete_callback(ma_pa_stream* pStream, int success, void* pUserData) +typedef struct { - ma_bool32* pIsSuccessful = (ma_bool32*)pUserData; - ma_assert(pIsSuccessful != NULL); - - *pIsSuccessful = (ma_bool32)success; - - (void)pStream; /* Unused. */ -} + ma_uint32 maxlength; + ma_uint32 tlength; + ma_uint32 prebuf; + ma_uint32 minreq; + ma_uint32 fragsize; +} ma_pa_buffer_attr; -ma_result ma_device__cork_stream__pulse(ma_device* pDevice, ma_device_type deviceType, int cork) +typedef struct { - ma_context* pContext = pDevice->pContext; - ma_bool32 wasSuccessful; - ma_pa_stream* pStream; - ma_pa_operation* pOP; - ma_result result; - - /* This should not be called with a duplex device type. */ - if (deviceType == ma_device_type_duplex) { - return MA_INVALID_ARGS; - } - - wasSuccessful = MA_FALSE; - - pStream = (ma_pa_stream*)((deviceType == ma_device_type_capture) ? pDevice->pulse.pStreamCapture : pDevice->pulse.pStreamPlayback); - ma_assert(pStream != NULL); - - pOP = ((ma_pa_stream_cork_proc)pContext->pulse.pa_stream_cork)(pStream, cork, ma_pulse_operation_complete_callback, &wasSuccessful); - if (pOP == NULL) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to cork PulseAudio stream.", (cork == 0) ? MA_FAILED_TO_START_BACKEND_DEVICE : MA_FAILED_TO_STOP_BACKEND_DEVICE); - } - - result = ma_device__wait_for_operation__pulse(pDevice, pOP); - ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); - - if (result != MA_SUCCESS) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] An error occurred while waiting for the PulseAudio stream to cork.", result); - } + ma_uint8 channels; + ma_pa_channel_position_t map[MA_PA_CHANNELS_MAX]; +} ma_pa_channel_map; - if (!wasSuccessful) { - if (cork) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to stop PulseAudio stream.", MA_FAILED_TO_STOP_BACKEND_DEVICE); - } else { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to start PulseAudio stream.", MA_FAILED_TO_START_BACKEND_DEVICE); - } - } +typedef struct +{ + ma_uint8 channels; + ma_uint32 values[MA_PA_CHANNELS_MAX]; +} ma_pa_cvolume; - return MA_SUCCESS; -} +typedef struct +{ + ma_pa_sample_format_t format; + ma_uint32 rate; + ma_uint8 channels; +} ma_pa_sample_spec; -ma_result ma_device_stop__pulse(ma_device* pDevice) +typedef struct { - ma_result result; - ma_bool32 wasSuccessful; - ma_pa_operation* pOP; + const char* name; + ma_uint32 index; + const char* description; + ma_pa_sample_spec sample_spec; + ma_pa_channel_map channel_map; + ma_uint32 owner_module; + ma_pa_cvolume volume; + int mute; + ma_uint32 monitor_source; + const char* monitor_source_name; + ma_uint64 latency; + const char* driver; + ma_pa_sink_flags_t flags; + void* proplist; + ma_uint64 configured_latency; + ma_uint32 base_volume; + ma_pa_sink_state_t state; + ma_uint32 n_volume_steps; + ma_uint32 card; + ma_uint32 n_ports; + void** ports; + void* active_port; + ma_uint8 n_formats; + void** formats; +} ma_pa_sink_info; - ma_assert(pDevice != NULL); +typedef struct +{ + const char *name; + ma_uint32 index; + const char *description; + ma_pa_sample_spec sample_spec; + ma_pa_channel_map channel_map; + ma_uint32 owner_module; + ma_pa_cvolume volume; + int mute; + ma_uint32 monitor_of_sink; + const char *monitor_of_sink_name; + ma_uint64 latency; + const char *driver; + ma_pa_source_flags_t flags; + void* proplist; + ma_uint64 configured_latency; + ma_uint32 base_volume; + ma_pa_source_state_t state; + ma_uint32 n_volume_steps; + ma_uint32 card; + ma_uint32 n_ports; + void** ports; + void* active_port; + ma_uint8 n_formats; + void** formats; +} ma_pa_source_info; - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - result = ma_device__cork_stream__pulse(pDevice, ma_device_type_capture, 1); - if (result != MA_SUCCESS) { - return result; - } - } +typedef void (* ma_pa_context_notify_cb_t)(ma_pa_context* c, void* userdata); +typedef void (* ma_pa_sink_info_cb_t) (ma_pa_context* c, const ma_pa_sink_info* i, int eol, void* userdata); +typedef void (* ma_pa_source_info_cb_t) (ma_pa_context* c, const ma_pa_source_info* i, int eol, void* userdata); +typedef void (* ma_pa_stream_success_cb_t)(ma_pa_stream* s, int success, void* userdata); +typedef void (* ma_pa_stream_request_cb_t)(ma_pa_stream* s, size_t nbytes, void* userdata); +typedef void (* ma_pa_free_cb_t) (void* p); +#endif - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - /* The stream needs to be drained if it's a playback device. */ - pOP = ((ma_pa_stream_drain_proc)pDevice->pContext->pulse.pa_stream_drain)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, ma_pulse_operation_complete_callback, &wasSuccessful); - if (pOP != NULL) { - ma_device__wait_for_operation__pulse(pDevice, pOP); - ((ma_pa_operation_unref_proc)pDevice->pContext->pulse.pa_operation_unref)(pOP); - } - result = ma_device__cork_stream__pulse(pDevice, ma_device_type_playback, 1); - if (result != MA_SUCCESS) { - return result; - } - } +typedef ma_pa_mainloop* (* ma_pa_mainloop_new_proc) (); +typedef void (* ma_pa_mainloop_free_proc) (ma_pa_mainloop* m); +typedef ma_pa_mainloop_api* (* ma_pa_mainloop_get_api_proc) (ma_pa_mainloop* m); +typedef int (* ma_pa_mainloop_iterate_proc) (ma_pa_mainloop* m, int block, int* retval); +typedef void (* ma_pa_mainloop_wakeup_proc) (ma_pa_mainloop* m); +typedef ma_pa_context* (* ma_pa_context_new_proc) (ma_pa_mainloop_api* mainloop, const char* name); +typedef void (* ma_pa_context_unref_proc) (ma_pa_context* c); +typedef int (* ma_pa_context_connect_proc) (ma_pa_context* c, const char* server, ma_pa_context_flags_t flags, const ma_pa_spawn_api* api); +typedef void (* ma_pa_context_disconnect_proc) (ma_pa_context* c); +typedef void (* ma_pa_context_set_state_callback_proc) (ma_pa_context* c, ma_pa_context_notify_cb_t cb, void* userdata); +typedef ma_pa_context_state_t (* ma_pa_context_get_state_proc) (ma_pa_context* c); +typedef ma_pa_operation* (* ma_pa_context_get_sink_info_list_proc) (ma_pa_context* c, ma_pa_sink_info_cb_t cb, void* userdata); +typedef ma_pa_operation* (* ma_pa_context_get_source_info_list_proc) (ma_pa_context* c, ma_pa_source_info_cb_t cb, void* userdata); +typedef ma_pa_operation* (* ma_pa_context_get_sink_info_by_name_proc) (ma_pa_context* c, const char* name, ma_pa_sink_info_cb_t cb, void* userdata); +typedef ma_pa_operation* (* ma_pa_context_get_source_info_by_name_proc)(ma_pa_context* c, const char* name, ma_pa_source_info_cb_t cb, void* userdata); +typedef void (* ma_pa_operation_unref_proc) (ma_pa_operation* o); +typedef ma_pa_operation_state_t (* ma_pa_operation_get_state_proc) (ma_pa_operation* o); +typedef ma_pa_channel_map* (* ma_pa_channel_map_init_extend_proc) (ma_pa_channel_map* m, unsigned channels, ma_pa_channel_map_def_t def); +typedef int (* ma_pa_channel_map_valid_proc) (const ma_pa_channel_map* m); +typedef int (* ma_pa_channel_map_compatible_proc) (const ma_pa_channel_map* m, const ma_pa_sample_spec* ss); +typedef ma_pa_stream* (* ma_pa_stream_new_proc) (ma_pa_context* c, const char* name, const ma_pa_sample_spec* ss, const ma_pa_channel_map* map); +typedef void (* ma_pa_stream_unref_proc) (ma_pa_stream* s); +typedef int (* ma_pa_stream_connect_playback_proc) (ma_pa_stream* s, const char* dev, const ma_pa_buffer_attr* attr, ma_pa_stream_flags_t flags, const ma_pa_cvolume* volume, ma_pa_stream* sync_stream); +typedef int (* ma_pa_stream_connect_record_proc) (ma_pa_stream* s, const char* dev, const ma_pa_buffer_attr* attr, ma_pa_stream_flags_t flags); +typedef int (* ma_pa_stream_disconnect_proc) (ma_pa_stream* s); +typedef ma_pa_stream_state_t (* ma_pa_stream_get_state_proc) (ma_pa_stream* s); +typedef const ma_pa_sample_spec* (* ma_pa_stream_get_sample_spec_proc) (ma_pa_stream* s); +typedef const ma_pa_channel_map* (* ma_pa_stream_get_channel_map_proc) (ma_pa_stream* s); +typedef const ma_pa_buffer_attr* (* ma_pa_stream_get_buffer_attr_proc) (ma_pa_stream* s); +typedef ma_pa_operation* (* ma_pa_stream_set_buffer_attr_proc) (ma_pa_stream* s, const ma_pa_buffer_attr* attr, ma_pa_stream_success_cb_t cb, void* userdata); +typedef const char* (* ma_pa_stream_get_device_name_proc) (ma_pa_stream* s); +typedef void (* ma_pa_stream_set_write_callback_proc) (ma_pa_stream* s, ma_pa_stream_request_cb_t cb, void* userdata); +typedef void (* ma_pa_stream_set_read_callback_proc) (ma_pa_stream* s, ma_pa_stream_request_cb_t cb, void* userdata); +typedef ma_pa_operation* (* ma_pa_stream_flush_proc) (ma_pa_stream* s, ma_pa_stream_success_cb_t cb, void* userdata); +typedef ma_pa_operation* (* ma_pa_stream_drain_proc) (ma_pa_stream* s, ma_pa_stream_success_cb_t cb, void* userdata); +typedef int (* ma_pa_stream_is_corked_proc) (ma_pa_stream* s); +typedef ma_pa_operation* (* ma_pa_stream_cork_proc) (ma_pa_stream* s, int b, ma_pa_stream_success_cb_t cb, void* userdata); +typedef ma_pa_operation* (* ma_pa_stream_trigger_proc) (ma_pa_stream* s, ma_pa_stream_success_cb_t cb, void* userdata); +typedef int (* ma_pa_stream_begin_write_proc) (ma_pa_stream* s, void** data, size_t* nbytes); +typedef int (* ma_pa_stream_write_proc) (ma_pa_stream* s, const void* data, size_t nbytes, ma_pa_free_cb_t free_cb, int64_t offset, ma_pa_seek_mode_t seek); +typedef int (* ma_pa_stream_peek_proc) (ma_pa_stream* s, const void** data, size_t* nbytes); +typedef int (* ma_pa_stream_drop_proc) (ma_pa_stream* s); +typedef size_t (* ma_pa_stream_writable_size_proc) (ma_pa_stream* s); +typedef size_t (* ma_pa_stream_readable_size_proc) (ma_pa_stream* s); - return MA_SUCCESS; -} +typedef struct +{ + ma_uint32 count; + ma_uint32 capacity; + ma_device_info* pInfo; +} ma_pulse_device_enum_data; -ma_result ma_device_write__pulse(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) +static ma_result ma_result_from_pulse(int result) { - ma_uint32 totalFramesWritten; - - ma_assert(pDevice != NULL); - ma_assert(pPCMFrames != NULL); - ma_assert(frameCount > 0); - - if (pFramesWritten != NULL) { - *pFramesWritten = 0; + switch (result) { + case MA_PA_OK: return MA_SUCCESS; + case MA_PA_ERR_ACCESS: return MA_ACCESS_DENIED; + case MA_PA_ERR_INVALID: return MA_INVALID_ARGS; + case MA_PA_ERR_NOENTITY: return MA_NO_DEVICE; + default: return MA_ERROR; } +} - totalFramesWritten = 0; - while (totalFramesWritten < frameCount) { - if (ma_device__get_state(pDevice) != MA_STATE_STARTED) { - return MA_DEVICE_NOT_STARTED; +#if 0 +static ma_pa_sample_format_t ma_format_to_pulse(ma_format format) +{ + if (ma_is_little_endian()) { + switch (format) { + case ma_format_s16: return MA_PA_SAMPLE_S16LE; + case ma_format_s24: return MA_PA_SAMPLE_S24LE; + case ma_format_s32: return MA_PA_SAMPLE_S32LE; + case ma_format_f32: return MA_PA_SAMPLE_FLOAT32LE; + default: break; } - - /* Place the data into the mapped buffer if we have one. */ - if (pDevice->pulse.pMappedBufferPlayback != NULL && pDevice->pulse.mappedBufferFramesRemainingPlayback > 0) { - ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - ma_uint32 mappedBufferFramesConsumed = pDevice->pulse.mappedBufferFramesCapacityPlayback - pDevice->pulse.mappedBufferFramesRemainingPlayback; - - void* pDst = (ma_uint8*)pDevice->pulse.pMappedBufferPlayback + (mappedBufferFramesConsumed * bpf); - const void* pSrc = (const ma_uint8*)pPCMFrames + (totalFramesWritten * bpf); - ma_uint32 framesToCopy = ma_min(pDevice->pulse.mappedBufferFramesRemainingPlayback, (frameCount - totalFramesWritten)); - ma_copy_memory(pDst, pSrc, framesToCopy * bpf); - - pDevice->pulse.mappedBufferFramesRemainingPlayback -= framesToCopy; - totalFramesWritten += framesToCopy; + } else { + switch (format) { + case ma_format_s16: return MA_PA_SAMPLE_S16BE; + case ma_format_s24: return MA_PA_SAMPLE_S24BE; + case ma_format_s32: return MA_PA_SAMPLE_S32BE; + case ma_format_f32: return MA_PA_SAMPLE_FLOAT32BE; + default: break; } + } - /* - Getting here means we've run out of data in the currently mapped chunk. We need to write this to the device and then try - mapping another chunk. If this fails we need to wait for space to become available. - */ - if (pDevice->pulse.mappedBufferFramesCapacityPlayback > 0 && pDevice->pulse.mappedBufferFramesRemainingPlayback == 0) { - size_t nbytes = pDevice->pulse.mappedBufferFramesCapacityPlayback * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - - int error = ((ma_pa_stream_write_proc)pDevice->pContext->pulse.pa_stream_write)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, pDevice->pulse.pMappedBufferPlayback, nbytes, NULL, 0, MA_PA_SEEK_RELATIVE); - if (error < 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to write data to the PulseAudio stream.", ma_result_from_pulse(error)); - } - - pDevice->pulse.pMappedBufferPlayback = NULL; - pDevice->pulse.mappedBufferFramesRemainingPlayback = 0; - pDevice->pulse.mappedBufferFramesCapacityPlayback = 0; - } + /* Endian agnostic. */ + switch (format) { + case ma_format_u8: return MA_PA_SAMPLE_U8; + default: return MA_PA_SAMPLE_INVALID; + } +} +#endif - ma_assert(totalFramesWritten <= frameCount); - if (totalFramesWritten == frameCount) { - break; +static ma_format ma_format_from_pulse(ma_pa_sample_format_t format) +{ + if (ma_is_little_endian()) { + switch (format) { + case MA_PA_SAMPLE_S16LE: return ma_format_s16; + case MA_PA_SAMPLE_S24LE: return ma_format_s24; + case MA_PA_SAMPLE_S32LE: return ma_format_s32; + case MA_PA_SAMPLE_FLOAT32LE: return ma_format_f32; + default: break; } - - /* Getting here means we need to map a new buffer. If we don't have enough space we need to wait for more. */ - for (;;) { - size_t writableSizeInBytes; - - /* If the device has been corked, don't try to continue. */ - if (((ma_pa_stream_is_corked_proc)pDevice->pContext->pulse.pa_stream_is_corked)((ma_pa_stream*)pDevice->pulse.pStreamPlayback)) { - break; - } - - writableSizeInBytes = ((ma_pa_stream_writable_size_proc)pDevice->pContext->pulse.pa_stream_writable_size)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); - if (writableSizeInBytes != (size_t)-1) { - /*size_t periodSizeInBytes = (pDevice->playback.internalBufferSizeInFrames / pDevice->playback.internalPeriods) * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);*/ - if (writableSizeInBytes > 0) { - /* Data is avaialable. */ - size_t bytesToMap = writableSizeInBytes; - int error = ((ma_pa_stream_begin_write_proc)pDevice->pContext->pulse.pa_stream_begin_write)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, &pDevice->pulse.pMappedBufferPlayback, &bytesToMap); - if (error < 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to map write buffer.", ma_result_from_pulse(error)); - } - - pDevice->pulse.mappedBufferFramesCapacityPlayback = bytesToMap / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - pDevice->pulse.mappedBufferFramesRemainingPlayback = pDevice->pulse.mappedBufferFramesCapacityPlayback; - - break; - } else { - /* No data available. Need to wait for more. */ - int error = ((ma_pa_mainloop_iterate_proc)pDevice->pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pDevice->pulse.pMainLoop, 1, NULL); - if (error < 0) { - return ma_result_from_pulse(error); - } - - continue; - } - } else { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to query the stream's writable size.", MA_ERROR); - } + } else { + switch (format) { + case MA_PA_SAMPLE_S16BE: return ma_format_s16; + case MA_PA_SAMPLE_S24BE: return ma_format_s24; + case MA_PA_SAMPLE_S32BE: return ma_format_s32; + case MA_PA_SAMPLE_FLOAT32BE: return ma_format_f32; + default: break; } } - if (pFramesWritten != NULL) { - *pFramesWritten = totalFramesWritten; + /* Endian agnostic. */ + switch (format) { + case MA_PA_SAMPLE_U8: return ma_format_u8; + default: return ma_format_unknown; } - - return MA_SUCCESS; } -ma_result ma_device_read__pulse(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead) +static ma_channel ma_channel_position_from_pulse(ma_pa_channel_position_t position) { - ma_uint32 totalFramesRead; - - ma_assert(pDevice != NULL); - ma_assert(pPCMFrames != NULL); - ma_assert(frameCount > 0); - - if (pFramesRead != NULL) { - *pFramesRead = 0; + switch (position) + { + case MA_PA_CHANNEL_POSITION_INVALID: return MA_CHANNEL_NONE; + case MA_PA_CHANNEL_POSITION_MONO: return MA_CHANNEL_MONO; + case MA_PA_CHANNEL_POSITION_FRONT_LEFT: return MA_CHANNEL_FRONT_LEFT; + case MA_PA_CHANNEL_POSITION_FRONT_RIGHT: return MA_CHANNEL_FRONT_RIGHT; + case MA_PA_CHANNEL_POSITION_FRONT_CENTER: return MA_CHANNEL_FRONT_CENTER; + case MA_PA_CHANNEL_POSITION_REAR_CENTER: return MA_CHANNEL_BACK_CENTER; + case MA_PA_CHANNEL_POSITION_REAR_LEFT: return MA_CHANNEL_BACK_LEFT; + case MA_PA_CHANNEL_POSITION_REAR_RIGHT: return MA_CHANNEL_BACK_RIGHT; + case MA_PA_CHANNEL_POSITION_LFE: return MA_CHANNEL_LFE; + case MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER: return MA_CHANNEL_FRONT_LEFT_CENTER; + case MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER: return MA_CHANNEL_FRONT_RIGHT_CENTER; + case MA_PA_CHANNEL_POSITION_SIDE_LEFT: return MA_CHANNEL_SIDE_LEFT; + case MA_PA_CHANNEL_POSITION_SIDE_RIGHT: return MA_CHANNEL_SIDE_RIGHT; + case MA_PA_CHANNEL_POSITION_AUX0: return MA_CHANNEL_AUX_0; + case MA_PA_CHANNEL_POSITION_AUX1: return MA_CHANNEL_AUX_1; + case MA_PA_CHANNEL_POSITION_AUX2: return MA_CHANNEL_AUX_2; + case MA_PA_CHANNEL_POSITION_AUX3: return MA_CHANNEL_AUX_3; + case MA_PA_CHANNEL_POSITION_AUX4: return MA_CHANNEL_AUX_4; + case MA_PA_CHANNEL_POSITION_AUX5: return MA_CHANNEL_AUX_5; + case MA_PA_CHANNEL_POSITION_AUX6: return MA_CHANNEL_AUX_6; + case MA_PA_CHANNEL_POSITION_AUX7: return MA_CHANNEL_AUX_7; + case MA_PA_CHANNEL_POSITION_AUX8: return MA_CHANNEL_AUX_8; + case MA_PA_CHANNEL_POSITION_AUX9: return MA_CHANNEL_AUX_9; + case MA_PA_CHANNEL_POSITION_AUX10: return MA_CHANNEL_AUX_10; + case MA_PA_CHANNEL_POSITION_AUX11: return MA_CHANNEL_AUX_11; + case MA_PA_CHANNEL_POSITION_AUX12: return MA_CHANNEL_AUX_12; + case MA_PA_CHANNEL_POSITION_AUX13: return MA_CHANNEL_AUX_13; + case MA_PA_CHANNEL_POSITION_AUX14: return MA_CHANNEL_AUX_14; + case MA_PA_CHANNEL_POSITION_AUX15: return MA_CHANNEL_AUX_15; + case MA_PA_CHANNEL_POSITION_AUX16: return MA_CHANNEL_AUX_16; + case MA_PA_CHANNEL_POSITION_AUX17: return MA_CHANNEL_AUX_17; + case MA_PA_CHANNEL_POSITION_AUX18: return MA_CHANNEL_AUX_18; + case MA_PA_CHANNEL_POSITION_AUX19: return MA_CHANNEL_AUX_19; + case MA_PA_CHANNEL_POSITION_AUX20: return MA_CHANNEL_AUX_20; + case MA_PA_CHANNEL_POSITION_AUX21: return MA_CHANNEL_AUX_21; + case MA_PA_CHANNEL_POSITION_AUX22: return MA_CHANNEL_AUX_22; + case MA_PA_CHANNEL_POSITION_AUX23: return MA_CHANNEL_AUX_23; + case MA_PA_CHANNEL_POSITION_AUX24: return MA_CHANNEL_AUX_24; + case MA_PA_CHANNEL_POSITION_AUX25: return MA_CHANNEL_AUX_25; + case MA_PA_CHANNEL_POSITION_AUX26: return MA_CHANNEL_AUX_26; + case MA_PA_CHANNEL_POSITION_AUX27: return MA_CHANNEL_AUX_27; + case MA_PA_CHANNEL_POSITION_AUX28: return MA_CHANNEL_AUX_28; + case MA_PA_CHANNEL_POSITION_AUX29: return MA_CHANNEL_AUX_29; + case MA_PA_CHANNEL_POSITION_AUX30: return MA_CHANNEL_AUX_30; + case MA_PA_CHANNEL_POSITION_AUX31: return MA_CHANNEL_AUX_31; + case MA_PA_CHANNEL_POSITION_TOP_CENTER: return MA_CHANNEL_TOP_CENTER; + case MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT: return MA_CHANNEL_TOP_FRONT_LEFT; + case MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT: return MA_CHANNEL_TOP_FRONT_RIGHT; + case MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER: return MA_CHANNEL_TOP_FRONT_CENTER; + case MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT: return MA_CHANNEL_TOP_BACK_LEFT; + case MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT: return MA_CHANNEL_TOP_BACK_RIGHT; + case MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER: return MA_CHANNEL_TOP_BACK_CENTER; + default: return MA_CHANNEL_NONE; } +} - totalFramesRead = 0; - while (totalFramesRead < frameCount) { - if (ma_device__get_state(pDevice) != MA_STATE_STARTED) { - return MA_DEVICE_NOT_STARTED; - } - - /* If a buffer is mapped we need to write to that first. Once it's consumed we reset the event and unmap it. */ - if (pDevice->pulse.pMappedBufferCapture != NULL && pDevice->pulse.mappedBufferFramesRemainingCapture > 0) { - ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 mappedBufferFramesConsumed = pDevice->pulse.mappedBufferFramesCapacityCapture - pDevice->pulse.mappedBufferFramesRemainingCapture; - - ma_uint32 framesToCopy = ma_min(pDevice->pulse.mappedBufferFramesRemainingCapture, (frameCount - totalFramesRead)); - void* pDst = (ma_uint8*)pPCMFrames + (totalFramesRead * bpf); +#if 0 +static ma_pa_channel_position_t ma_channel_position_to_pulse(ma_channel position) +{ + switch (position) + { + case MA_CHANNEL_NONE: return MA_PA_CHANNEL_POSITION_INVALID; + case MA_CHANNEL_FRONT_LEFT: return MA_PA_CHANNEL_POSITION_FRONT_LEFT; + case MA_CHANNEL_FRONT_RIGHT: return MA_PA_CHANNEL_POSITION_FRONT_RIGHT; + case MA_CHANNEL_FRONT_CENTER: return MA_PA_CHANNEL_POSITION_FRONT_CENTER; + case MA_CHANNEL_LFE: return MA_PA_CHANNEL_POSITION_LFE; + case MA_CHANNEL_BACK_LEFT: return MA_PA_CHANNEL_POSITION_REAR_LEFT; + case MA_CHANNEL_BACK_RIGHT: return MA_PA_CHANNEL_POSITION_REAR_RIGHT; + case MA_CHANNEL_FRONT_LEFT_CENTER: return MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER; + case MA_CHANNEL_FRONT_RIGHT_CENTER: return MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER; + case MA_CHANNEL_BACK_CENTER: return MA_PA_CHANNEL_POSITION_REAR_CENTER; + case MA_CHANNEL_SIDE_LEFT: return MA_PA_CHANNEL_POSITION_SIDE_LEFT; + case MA_CHANNEL_SIDE_RIGHT: return MA_PA_CHANNEL_POSITION_SIDE_RIGHT; + case MA_CHANNEL_TOP_CENTER: return MA_PA_CHANNEL_POSITION_TOP_CENTER; + case MA_CHANNEL_TOP_FRONT_LEFT: return MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT; + case MA_CHANNEL_TOP_FRONT_CENTER: return MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER; + case MA_CHANNEL_TOP_FRONT_RIGHT: return MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT; + case MA_CHANNEL_TOP_BACK_LEFT: return MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT; + case MA_CHANNEL_TOP_BACK_CENTER: return MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER; + case MA_CHANNEL_TOP_BACK_RIGHT: return MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT; + case MA_CHANNEL_19: return MA_PA_CHANNEL_POSITION_AUX18; + case MA_CHANNEL_20: return MA_PA_CHANNEL_POSITION_AUX19; + case MA_CHANNEL_21: return MA_PA_CHANNEL_POSITION_AUX20; + case MA_CHANNEL_22: return MA_PA_CHANNEL_POSITION_AUX21; + case MA_CHANNEL_23: return MA_PA_CHANNEL_POSITION_AUX22; + case MA_CHANNEL_24: return MA_PA_CHANNEL_POSITION_AUX23; + case MA_CHANNEL_25: return MA_PA_CHANNEL_POSITION_AUX24; + case MA_CHANNEL_26: return MA_PA_CHANNEL_POSITION_AUX25; + case MA_CHANNEL_27: return MA_PA_CHANNEL_POSITION_AUX26; + case MA_CHANNEL_28: return MA_PA_CHANNEL_POSITION_AUX27; + case MA_CHANNEL_29: return MA_PA_CHANNEL_POSITION_AUX28; + case MA_CHANNEL_30: return MA_PA_CHANNEL_POSITION_AUX29; + case MA_CHANNEL_31: return MA_PA_CHANNEL_POSITION_AUX30; + case MA_CHANNEL_32: return MA_PA_CHANNEL_POSITION_AUX31; + default: return (ma_pa_channel_position_t)position; + } +} +#endif - /* - This little bit of logic here is specifically for PulseAudio and it's hole management. The buffer pointer will be set to NULL - when the current fragment is a hole. For a hole we just output silence. - */ - if (pDevice->pulse.pMappedBufferCapture != NULL) { - const void* pSrc = (const ma_uint8*)pDevice->pulse.pMappedBufferCapture + (mappedBufferFramesConsumed * bpf); - ma_copy_memory(pDst, pSrc, framesToCopy * bpf); - } else { - ma_zero_memory(pDst, framesToCopy * bpf); - } +static ma_result ma_wait_for_operation__pulse(ma_context* pContext, ma_pa_mainloop* pMainLoop, ma_pa_operation* pOP) +{ + MA_ASSERT(pContext != NULL); + MA_ASSERT(pMainLoop != NULL); + MA_ASSERT(pOP != NULL); - pDevice->pulse.mappedBufferFramesRemainingCapture -= framesToCopy; - totalFramesRead += framesToCopy; + while (((ma_pa_operation_get_state_proc)pContext->pulse.pa_operation_get_state)(pOP) == MA_PA_OPERATION_RUNNING) { + int error = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)(pMainLoop, 1, NULL); + if (error < 0) { + return ma_result_from_pulse(error); } + } - /* - Getting here means we've run out of data in the currently mapped chunk. We need to drop this from the device and then try - mapping another chunk. If this fails we need to wait for data to become available. - */ - if (pDevice->pulse.mappedBufferFramesCapacityCapture > 0 && pDevice->pulse.mappedBufferFramesRemainingCapture == 0) { - int error = ((ma_pa_stream_drop_proc)pDevice->pContext->pulse.pa_stream_drop)((ma_pa_stream*)pDevice->pulse.pStreamCapture); - if (error != 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to drop fragment.", ma_result_from_pulse(error)); - } + return MA_SUCCESS; +} - pDevice->pulse.pMappedBufferCapture = NULL; - pDevice->pulse.mappedBufferFramesRemainingCapture = 0; - pDevice->pulse.mappedBufferFramesCapacityCapture = 0; - } +static ma_result ma_device__wait_for_operation__pulse(ma_device* pDevice, ma_pa_operation* pOP) +{ + MA_ASSERT(pDevice != NULL); + MA_ASSERT(pOP != NULL); - ma_assert(totalFramesRead <= frameCount); - if (totalFramesRead == frameCount) { - break; - } + return ma_wait_for_operation__pulse(pDevice->pContext, (ma_pa_mainloop*)pDevice->pulse.pMainLoop, pOP); +} - /* Getting here means we need to map a new buffer. If we don't have enough data we wait for more. */ - for (;;) { - size_t readableSizeInBytes; - if (ma_device__get_state(pDevice) != MA_STATE_STARTED) { - break; - } +static ma_bool32 ma_context_is_device_id_equal__pulse(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) +{ + MA_ASSERT(pContext != NULL); + MA_ASSERT(pID0 != NULL); + MA_ASSERT(pID1 != NULL); + (void)pContext; - /* If the device has been corked, don't try to continue. */ - if (((ma_pa_stream_is_corked_proc)pDevice->pContext->pulse.pa_stream_is_corked)((ma_pa_stream*)pDevice->pulse.pStreamCapture)) { - break; - } + return ma_strcmp(pID0->pulse, pID1->pulse) == 0; +} - readableSizeInBytes = ((ma_pa_stream_readable_size_proc)pDevice->pContext->pulse.pa_stream_readable_size)((ma_pa_stream*)pDevice->pulse.pStreamCapture); - if (readableSizeInBytes != (size_t)-1) { - /*size_t periodSizeInBytes = (pDevice->capture.internalBufferSizeInFrames / pDevice->capture.internalPeriods) * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);*/ - if (readableSizeInBytes > 0) { - /* Data is avaialable. */ - size_t bytesMapped = (size_t)-1; - int error = ((ma_pa_stream_peek_proc)pDevice->pContext->pulse.pa_stream_peek)((ma_pa_stream*)pDevice->pulse.pStreamCapture, &pDevice->pulse.pMappedBufferCapture, &bytesMapped); - if (error < 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to peek capture buffer.", ma_result_from_pulse(error)); - } - if (pDevice->pulse.pMappedBufferCapture == NULL && bytesMapped == 0) { - /* Nothing available. This shouldn't happen because we checked earlier with pa_stream_readable_size(). I'm going to throw an error in this case. */ - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Nothing available after peeking capture buffer.", MA_ERROR); - } +typedef struct +{ + ma_context* pContext; + ma_enum_devices_callback_proc callback; + void* pUserData; + ma_bool32 isTerminated; +} ma_context_enumerate_devices_callback_data__pulse; - pDevice->pulse.mappedBufferFramesCapacityCapture = bytesMapped / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - pDevice->pulse.mappedBufferFramesRemainingCapture = pDevice->pulse.mappedBufferFramesCapacityCapture; +static void ma_context_enumerate_devices_sink_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_sink_info* pSinkInfo, int endOfList, void* pUserData) +{ + ma_context_enumerate_devices_callback_data__pulse* pData = (ma_context_enumerate_devices_callback_data__pulse*)pUserData; + ma_device_info deviceInfo; - break; - } else { - /* No data available. Need to wait for more. */ + MA_ASSERT(pData != NULL); - /* - I have had reports of a deadlock in this part of the code. I have reproduced this when using the "Built-in Audio Analogue Stereo" device without - an actual microphone connected. I'm experimenting here by not blocking in pa_mainloop_iterate() and instead sleep for a bit when there are no - dispatches. - */ - int error = ((ma_pa_mainloop_iterate_proc)pDevice->pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pDevice->pulse.pMainLoop, 0, NULL); - if (error < 0) { - return ma_result_from_pulse(error); - } + if (endOfList || pData->isTerminated) { + return; + } - /* Sleep for a bit if nothing was dispatched. */ - if (error == 0) { - ma_sleep(1); - } + MA_ZERO_OBJECT(&deviceInfo); - continue; - } - } else { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to query the stream's readable size.", MA_ERROR); - } - } + /* The name from PulseAudio is the ID for miniaudio. */ + if (pSinkInfo->name != NULL) { + ma_strncpy_s(deviceInfo.id.pulse, sizeof(deviceInfo.id.pulse), pSinkInfo->name, (size_t)-1); } - if (pFramesRead != NULL) { - *pFramesRead = totalFramesRead; + /* The description from PulseAudio is the name for miniaudio. */ + if (pSinkInfo->description != NULL) { + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), pSinkInfo->description, (size_t)-1); } - return MA_SUCCESS; + pData->isTerminated = !pData->callback(pData->pContext, ma_device_type_playback, &deviceInfo, pData->pUserData); + + (void)pPulseContext; /* Unused. */ } -ma_result ma_device_main_loop__pulse(ma_device* pDevice) +static void ma_context_enumerate_devices_source_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_source_info* pSinkInfo, int endOfList, void* pUserData) { - ma_result result = MA_SUCCESS; - ma_bool32 exitLoop = MA_FALSE; + ma_context_enumerate_devices_callback_data__pulse* pData = (ma_context_enumerate_devices_callback_data__pulse*)pUserData; + ma_device_info deviceInfo; - ma_assert(pDevice != NULL); + MA_ASSERT(pData != NULL); - /* The stream needs to be uncorked first. We do this at the top for both capture and playback for PulseAudio. */ - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - result = ma_device__cork_stream__pulse(pDevice, ma_device_type_capture, 0); - if (result != MA_SUCCESS) { - return result; - } + if (endOfList || pData->isTerminated) { + return; } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - result = ma_device__cork_stream__pulse(pDevice, ma_device_type_playback, 0); - if (result != MA_SUCCESS) { - return result; - } + + MA_ZERO_OBJECT(&deviceInfo); + + /* The name from PulseAudio is the ID for miniaudio. */ + if (pSinkInfo->name != NULL) { + ma_strncpy_s(deviceInfo.id.pulse, sizeof(deviceInfo.id.pulse), pSinkInfo->name, (size_t)-1); } + /* The description from PulseAudio is the name for miniaudio. */ + if (pSinkInfo->description != NULL) { + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), pSinkInfo->description, (size_t)-1); + } - while (ma_device__get_state(pDevice) == MA_STATE_STARTED && !exitLoop) { - switch (pDevice->type) - { - case ma_device_type_duplex: - { - /* The process is: device_read -> convert -> callback -> convert -> device_write */ - ma_uint8 capturedDeviceData[8192]; - ma_uint8 playbackDeviceData[8192]; - ma_uint32 capturedDeviceDataCapInFrames = sizeof(capturedDeviceData) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 playbackDeviceDataCapInFrames = sizeof(playbackDeviceData) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + pData->isTerminated = !pData->callback(pData->pContext, ma_device_type_capture, &deviceInfo, pData->pUserData); - ma_uint32 totalFramesProcessed = 0; - ma_uint32 periodSizeInFrames = ma_min(pDevice->capture.internalBufferSizeInFrames/pDevice->capture.internalPeriods, pDevice->playback.internalBufferSizeInFrames/pDevice->playback.internalPeriods); - - while (totalFramesProcessed < periodSizeInFrames) { - ma_uint32 framesRemaining = periodSizeInFrames - totalFramesProcessed; - ma_uint32 framesProcessed; - ma_uint32 framesToProcess = framesRemaining; - if (framesToProcess > capturedDeviceDataCapInFrames) { - framesToProcess = capturedDeviceDataCapInFrames; - } + (void)pPulseContext; /* Unused. */ +} - result = ma_device_read__pulse(pDevice, capturedDeviceData, framesToProcess, &framesProcessed); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } +static ma_result ma_context_enumerate_devices__pulse(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +{ + ma_result result = MA_SUCCESS; + ma_context_enumerate_devices_callback_data__pulse callbackData; + ma_pa_operation* pOP = NULL; + ma_pa_mainloop* pMainLoop; + ma_pa_mainloop_api* pAPI; + ma_pa_context* pPulseContext; + int error; - pDevice->capture._dspFrameCount = framesToProcess; - pDevice->capture._dspFrames = capturedDeviceData; + MA_ASSERT(pContext != NULL); + MA_ASSERT(callback != NULL); - for (;;) { - ma_uint8 capturedData[8192]; - ma_uint8 playbackData[8192]; - ma_uint32 capturedDataCapInFrames = sizeof(capturedData) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); - ma_uint32 playbackDataCapInFrames = sizeof(playbackData) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); - - ma_uint32 capturedFramesToTryProcessing = ma_min(capturedDataCapInFrames, playbackDataCapInFrames); - ma_uint32 capturedFramesToProcess = (ma_uint32)ma_pcm_converter_read(&pDevice->capture.converter, capturedData, capturedFramesToTryProcessing); - if (capturedFramesToProcess == 0) { - break; /* Don't fire the data callback with zero frames. */ - } + callbackData.pContext = pContext; + callbackData.callback = callback; + callbackData.pUserData = pUserData; + callbackData.isTerminated = MA_FALSE; - ma_device__on_data(pDevice, playbackData, capturedData, capturedFramesToProcess); + pMainLoop = ((ma_pa_mainloop_new_proc)pContext->pulse.pa_mainloop_new)(); + if (pMainLoop == NULL) { + return MA_FAILED_TO_INIT_BACKEND; + } - /* At this point the playbackData buffer should be holding data that needs to be written to the device. */ - pDevice->playback._dspFrameCount = capturedFramesToProcess; - pDevice->playback._dspFrames = playbackData; - for (;;) { - ma_uint32 playbackDeviceFramesCount = (ma_uint32)ma_pcm_converter_read(&pDevice->playback.converter, playbackDeviceData, playbackDeviceDataCapInFrames); - if (playbackDeviceFramesCount == 0) { - break; - } + pAPI = ((ma_pa_mainloop_get_api_proc)pContext->pulse.pa_mainloop_get_api)(pMainLoop); + if (pAPI == NULL) { + ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); + return MA_FAILED_TO_INIT_BACKEND; + } - result = ma_device_write__pulse(pDevice, playbackDeviceData, playbackDeviceFramesCount, NULL); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } + pPulseContext = ((ma_pa_context_new_proc)pContext->pulse.pa_context_new)(pAPI, pContext->pulse.pApplicationName); + if (pPulseContext == NULL) { + ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); + return MA_FAILED_TO_INIT_BACKEND; + } - if (playbackDeviceFramesCount < playbackDeviceDataCapInFrames) { - break; - } - } + error = ((ma_pa_context_connect_proc)pContext->pulse.pa_context_connect)(pPulseContext, pContext->pulse.pServerName, (pContext->pulse.tryAutoSpawn) ? 0 : MA_PA_CONTEXT_NOAUTOSPAWN, NULL); + if (error != MA_PA_OK) { + ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)(pPulseContext); + ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); + return ma_result_from_pulse(error); + } - if (capturedFramesToProcess < capturedFramesToTryProcessing) { - break; - } + for (;;) { + ma_pa_context_state_t state = ((ma_pa_context_get_state_proc)pContext->pulse.pa_context_get_state)(pPulseContext); + if (state == MA_PA_CONTEXT_READY) { + break; /* Success. */ + } + if (state == MA_PA_CONTEXT_CONNECTING || state == MA_PA_CONTEXT_AUTHORIZING || state == MA_PA_CONTEXT_SETTING_NAME) { + error = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)(pMainLoop, 1, NULL); + if (error < 0) { + result = ma_result_from_pulse(error); + goto done; + } - /* In case an error happened from ma_device_write2__alsa()... */ - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } - } +#ifdef MA_DEBUG_OUTPUT + printf("[PulseAudio] pa_context_get_state() returned %d. Waiting.\n", state); +#endif + continue; /* Keep trying. */ + } + if (state == MA_PA_CONTEXT_UNCONNECTED || state == MA_PA_CONTEXT_FAILED || state == MA_PA_CONTEXT_TERMINATED) { +#ifdef MA_DEBUG_OUTPUT + printf("[PulseAudio] pa_context_get_state() returned %d. Failed.\n", state); +#endif + goto done; /* Failed. */ + } + } - totalFramesProcessed += framesProcessed; - } - } break; - case ma_device_type_capture: - { - ma_uint8 intermediaryBuffer[8192]; - ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 periodSizeInFrames = pDevice->capture.internalBufferSizeInFrames / pDevice->capture.internalPeriods; - ma_uint32 framesReadThisPeriod = 0; - while (framesReadThisPeriod < periodSizeInFrames) { - ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesReadThisPeriod; - ma_uint32 framesProcessed; - ma_uint32 framesToReadThisIteration = framesRemainingInPeriod; - if (framesToReadThisIteration > intermediaryBufferSizeInFrames) { - framesToReadThisIteration = intermediaryBufferSizeInFrames; - } + /* Playback. */ + if (!callbackData.isTerminated) { + pOP = ((ma_pa_context_get_sink_info_list_proc)pContext->pulse.pa_context_get_sink_info_list)(pPulseContext, ma_context_enumerate_devices_sink_callback__pulse, &callbackData); + if (pOP == NULL) { + result = MA_ERROR; + goto done; + } - result = ma_device_read__pulse(pDevice, intermediaryBuffer, framesToReadThisIteration, &framesProcessed); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } + result = ma_wait_for_operation__pulse(pContext, pMainLoop, pOP); + ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); + if (result != MA_SUCCESS) { + goto done; + } + } + + + /* Capture. */ + if (!callbackData.isTerminated) { + pOP = ((ma_pa_context_get_source_info_list_proc)pContext->pulse.pa_context_get_source_info_list)(pPulseContext, ma_context_enumerate_devices_source_callback__pulse, &callbackData); + if (pOP == NULL) { + result = MA_ERROR; + goto done; + } + + result = ma_wait_for_operation__pulse(pContext, pMainLoop, pOP); + ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); + if (result != MA_SUCCESS) { + goto done; + } + } + +done: + ((ma_pa_context_disconnect_proc)pContext->pulse.pa_context_disconnect)(pPulseContext); + ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)(pPulseContext); + ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); + return result; +} - ma_device__send_frames_to_client(pDevice, framesProcessed, intermediaryBuffer); - framesReadThisPeriod += framesProcessed; - } - } break; +typedef struct +{ + ma_device_info* pDeviceInfo; + ma_bool32 foundDevice; +} ma_context_get_device_info_callback_data__pulse; - case ma_device_type_playback: - { - ma_uint8 intermediaryBuffer[8192]; - ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - ma_uint32 periodSizeInFrames = pDevice->playback.internalBufferSizeInFrames / pDevice->playback.internalPeriods; - ma_uint32 framesWrittenThisPeriod = 0; - while (framesWrittenThisPeriod < periodSizeInFrames) { - ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesWrittenThisPeriod; - ma_uint32 framesProcessed; - ma_uint32 framesToWriteThisIteration = framesRemainingInPeriod; - if (framesToWriteThisIteration > intermediaryBufferSizeInFrames) { - framesToWriteThisIteration = intermediaryBufferSizeInFrames; - } +static void ma_context_get_device_info_sink_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_sink_info* pInfo, int endOfList, void* pUserData) +{ + ma_context_get_device_info_callback_data__pulse* pData = (ma_context_get_device_info_callback_data__pulse*)pUserData; - ma_device__read_frames_from_client(pDevice, framesToWriteThisIteration, intermediaryBuffer); + if (endOfList > 0) { + return; + } - result = ma_device_write__pulse(pDevice, intermediaryBuffer, framesToWriteThisIteration, &framesProcessed); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } + MA_ASSERT(pData != NULL); + pData->foundDevice = MA_TRUE; - framesWrittenThisPeriod += framesProcessed; - } - } break; + if (pInfo->name != NULL) { + ma_strncpy_s(pData->pDeviceInfo->id.pulse, sizeof(pData->pDeviceInfo->id.pulse), pInfo->name, (size_t)-1); + } - /* To silence a warning. Will never hit this. */ - case ma_device_type_loopback: - default: break; - } + if (pInfo->description != NULL) { + ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pInfo->description, (size_t)-1); } - /* Here is where the device needs to be stopped. */ - ma_device_stop__pulse(pDevice); + pData->pDeviceInfo->minChannels = pInfo->sample_spec.channels; + pData->pDeviceInfo->maxChannels = pInfo->sample_spec.channels; + pData->pDeviceInfo->minSampleRate = pInfo->sample_spec.rate; + pData->pDeviceInfo->maxSampleRate = pInfo->sample_spec.rate; + pData->pDeviceInfo->formatCount = 1; + pData->pDeviceInfo->formats[0] = ma_format_from_pulse(pInfo->sample_spec.format); - return result; + (void)pPulseContext; /* Unused. */ } - -ma_result ma_context_uninit__pulse(ma_context* pContext) +static void ma_context_get_device_info_source_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_source_info* pInfo, int endOfList, void* pUserData) { - ma_assert(pContext != NULL); - ma_assert(pContext->backend == ma_backend_pulseaudio); + ma_context_get_device_info_callback_data__pulse* pData = (ma_context_get_device_info_callback_data__pulse*)pUserData; - ma_free(pContext->pulse.pServerName); - pContext->pulse.pServerName = NULL; + if (endOfList > 0) { + return; + } - ma_free(pContext->pulse.pApplicationName); - pContext->pulse.pApplicationName = NULL; + MA_ASSERT(pData != NULL); + pData->foundDevice = MA_TRUE; -#ifndef MA_NO_RUNTIME_LINKING - ma_dlclose(pContext, pContext->pulse.pulseSO); -#endif + if (pInfo->name != NULL) { + ma_strncpy_s(pData->pDeviceInfo->id.pulse, sizeof(pData->pDeviceInfo->id.pulse), pInfo->name, (size_t)-1); + } - return MA_SUCCESS; + if (pInfo->description != NULL) { + ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pInfo->description, (size_t)-1); + } + + pData->pDeviceInfo->minChannels = pInfo->sample_spec.channels; + pData->pDeviceInfo->maxChannels = pInfo->sample_spec.channels; + pData->pDeviceInfo->minSampleRate = pInfo->sample_spec.rate; + pData->pDeviceInfo->maxSampleRate = pInfo->sample_spec.rate; + pData->pDeviceInfo->formatCount = 1; + pData->pDeviceInfo->formats[0] = ma_format_from_pulse(pInfo->sample_spec.format); + + (void)pPulseContext; /* Unused. */ } -ma_result ma_context_init__pulse(const ma_context_config* pConfig, ma_context* pContext) +static ma_result ma_context_get_device_info__pulse(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) { -#ifndef MA_NO_RUNTIME_LINKING - const char* libpulseNames[] = { - "libpulse.so", - "libpulse.so.0" - }; - size_t i; + ma_result result = MA_SUCCESS; + ma_context_get_device_info_callback_data__pulse callbackData; + ma_pa_operation* pOP = NULL; + ma_pa_mainloop* pMainLoop; + ma_pa_mainloop_api* pAPI; + ma_pa_context* pPulseContext; + int error; - for (i = 0; i < ma_countof(libpulseNames); ++i) { - pContext->pulse.pulseSO = ma_dlopen(pContext, libpulseNames[i]); - if (pContext->pulse.pulseSO != NULL) { - break; - } - } + MA_ASSERT(pContext != NULL); - if (pContext->pulse.pulseSO == NULL) { - return MA_NO_BACKEND; + /* No exclusive mode with the PulseAudio backend. */ + if (shareMode == ma_share_mode_exclusive) { + return MA_SHARE_MODE_NOT_SUPPORTED; } - pContext->pulse.pa_mainloop_new = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_new"); - pContext->pulse.pa_mainloop_free = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_free"); - pContext->pulse.pa_mainloop_get_api = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_get_api"); - pContext->pulse.pa_mainloop_iterate = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_iterate"); - pContext->pulse.pa_mainloop_wakeup = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_wakeup"); - pContext->pulse.pa_context_new = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_new"); - pContext->pulse.pa_context_unref = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_unref"); - pContext->pulse.pa_context_connect = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_connect"); - pContext->pulse.pa_context_disconnect = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_disconnect"); - pContext->pulse.pa_context_set_state_callback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_set_state_callback"); - pContext->pulse.pa_context_get_state = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_state"); - pContext->pulse.pa_context_get_sink_info_list = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_sink_info_list"); - pContext->pulse.pa_context_get_source_info_list = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_source_info_list"); - pContext->pulse.pa_context_get_sink_info_by_name = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_sink_info_by_name"); - pContext->pulse.pa_context_get_source_info_by_name = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_source_info_by_name"); - pContext->pulse.pa_operation_unref = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_operation_unref"); - pContext->pulse.pa_operation_get_state = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_operation_get_state"); - pContext->pulse.pa_channel_map_init_extend = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_channel_map_init_extend"); - pContext->pulse.pa_channel_map_valid = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_channel_map_valid"); - pContext->pulse.pa_channel_map_compatible = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_channel_map_compatible"); - pContext->pulse.pa_stream_new = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_new"); - pContext->pulse.pa_stream_unref = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_unref"); - pContext->pulse.pa_stream_connect_playback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_connect_playback"); - pContext->pulse.pa_stream_connect_record = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_connect_record"); - pContext->pulse.pa_stream_disconnect = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_disconnect"); - pContext->pulse.pa_stream_get_state = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_state"); - pContext->pulse.pa_stream_get_sample_spec = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_sample_spec"); - pContext->pulse.pa_stream_get_channel_map = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_channel_map"); - pContext->pulse.pa_stream_get_buffer_attr = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_buffer_attr"); - pContext->pulse.pa_stream_set_buffer_attr = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_set_buffer_attr"); - pContext->pulse.pa_stream_get_device_name = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_device_name"); - pContext->pulse.pa_stream_set_write_callback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_set_write_callback"); - pContext->pulse.pa_stream_set_read_callback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_set_read_callback"); - pContext->pulse.pa_stream_flush = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_flush"); - pContext->pulse.pa_stream_drain = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_drain"); - pContext->pulse.pa_stream_is_corked = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_is_corked"); - pContext->pulse.pa_stream_cork = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_cork"); - pContext->pulse.pa_stream_trigger = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_trigger"); - pContext->pulse.pa_stream_begin_write = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_begin_write"); - pContext->pulse.pa_stream_write = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_write"); - pContext->pulse.pa_stream_peek = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_peek"); - pContext->pulse.pa_stream_drop = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_drop"); - pContext->pulse.pa_stream_writable_size = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_writable_size"); - pContext->pulse.pa_stream_readable_size = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_readable_size"); -#else - /* This strange assignment system is just for type safety. */ - ma_pa_mainloop_new_proc _pa_mainloop_new = pa_mainloop_new; - ma_pa_mainloop_free_proc _pa_mainloop_free = pa_mainloop_free; - ma_pa_mainloop_get_api_proc _pa_mainloop_get_api = pa_mainloop_get_api; - ma_pa_mainloop_iterate_proc _pa_mainloop_iterate = pa_mainloop_iterate; - ma_pa_mainloop_wakeup_proc _pa_mainloop_wakeup = pa_mainloop_wakeup; - ma_pa_context_new_proc _pa_context_new = pa_context_new; - ma_pa_context_unref_proc _pa_context_unref = pa_context_unref; - ma_pa_context_connect_proc _pa_context_connect = pa_context_connect; - ma_pa_context_disconnect_proc _pa_context_disconnect = pa_context_disconnect; - ma_pa_context_set_state_callback_proc _pa_context_set_state_callback = pa_context_set_state_callback; - ma_pa_context_get_state_proc _pa_context_get_state = pa_context_get_state; - ma_pa_context_get_sink_info_list_proc _pa_context_get_sink_info_list = pa_context_get_sink_info_list; - ma_pa_context_get_source_info_list_proc _pa_context_get_source_info_list = pa_context_get_source_info_list; - ma_pa_context_get_sink_info_by_name_proc _pa_context_get_sink_info_by_name = pa_context_get_sink_info_by_name; - ma_pa_context_get_source_info_by_name_proc _pa_context_get_source_info_by_name= pa_context_get_source_info_by_name; - ma_pa_operation_unref_proc _pa_operation_unref = pa_operation_unref; - ma_pa_operation_get_state_proc _pa_operation_get_state = pa_operation_get_state; - ma_pa_channel_map_init_extend_proc _pa_channel_map_init_extend = pa_channel_map_init_extend; - ma_pa_channel_map_valid_proc _pa_channel_map_valid = pa_channel_map_valid; - ma_pa_channel_map_compatible_proc _pa_channel_map_compatible = pa_channel_map_compatible; - ma_pa_stream_new_proc _pa_stream_new = pa_stream_new; - ma_pa_stream_unref_proc _pa_stream_unref = pa_stream_unref; - ma_pa_stream_connect_playback_proc _pa_stream_connect_playback = pa_stream_connect_playback; - ma_pa_stream_connect_record_proc _pa_stream_connect_record = pa_stream_connect_record; - ma_pa_stream_disconnect_proc _pa_stream_disconnect = pa_stream_disconnect; - ma_pa_stream_get_state_proc _pa_stream_get_state = pa_stream_get_state; - ma_pa_stream_get_sample_spec_proc _pa_stream_get_sample_spec = pa_stream_get_sample_spec; - ma_pa_stream_get_channel_map_proc _pa_stream_get_channel_map = pa_stream_get_channel_map; - ma_pa_stream_get_buffer_attr_proc _pa_stream_get_buffer_attr = pa_stream_get_buffer_attr; - ma_pa_stream_set_buffer_attr_proc _pa_stream_set_buffer_attr = pa_stream_set_buffer_attr; - ma_pa_stream_get_device_name_proc _pa_stream_get_device_name = pa_stream_get_device_name; - ma_pa_stream_set_write_callback_proc _pa_stream_set_write_callback = pa_stream_set_write_callback; - ma_pa_stream_set_read_callback_proc _pa_stream_set_read_callback = pa_stream_set_read_callback; - ma_pa_stream_flush_proc _pa_stream_flush = pa_stream_flush; - ma_pa_stream_drain_proc _pa_stream_drain = pa_stream_drain; - ma_pa_stream_is_corked_proc _pa_stream_is_corked = pa_stream_is_corked; - ma_pa_stream_cork_proc _pa_stream_cork = pa_stream_cork; - ma_pa_stream_trigger_proc _pa_stream_trigger = pa_stream_trigger; - ma_pa_stream_begin_write_proc _pa_stream_begin_write = pa_stream_begin_write; - ma_pa_stream_write_proc _pa_stream_write = pa_stream_write; - ma_pa_stream_peek_proc _pa_stream_peek = pa_stream_peek; - ma_pa_stream_drop_proc _pa_stream_drop = pa_stream_drop; - ma_pa_stream_writable_size_proc _pa_stream_writable_size = pa_stream_writable_size; - ma_pa_stream_readable_size_proc _pa_stream_readable_size = pa_stream_readable_size; + callbackData.pDeviceInfo = pDeviceInfo; + callbackData.foundDevice = MA_FALSE; - pContext->pulse.pa_mainloop_new = (ma_proc)_pa_mainloop_new; - pContext->pulse.pa_mainloop_free = (ma_proc)_pa_mainloop_free; - pContext->pulse.pa_mainloop_get_api = (ma_proc)_pa_mainloop_get_api; - pContext->pulse.pa_mainloop_iterate = (ma_proc)_pa_mainloop_iterate; - pContext->pulse.pa_mainloop_wakeup = (ma_proc)_pa_mainloop_wakeup; - pContext->pulse.pa_context_new = (ma_proc)_pa_context_new; - pContext->pulse.pa_context_unref = (ma_proc)_pa_context_unref; - pContext->pulse.pa_context_connect = (ma_proc)_pa_context_connect; - pContext->pulse.pa_context_disconnect = (ma_proc)_pa_context_disconnect; - pContext->pulse.pa_context_set_state_callback = (ma_proc)_pa_context_set_state_callback; - pContext->pulse.pa_context_get_state = (ma_proc)_pa_context_get_state; - pContext->pulse.pa_context_get_sink_info_list = (ma_proc)_pa_context_get_sink_info_list; - pContext->pulse.pa_context_get_source_info_list = (ma_proc)_pa_context_get_source_info_list; - pContext->pulse.pa_context_get_sink_info_by_name = (ma_proc)_pa_context_get_sink_info_by_name; - pContext->pulse.pa_context_get_source_info_by_name = (ma_proc)_pa_context_get_source_info_by_name; - pContext->pulse.pa_operation_unref = (ma_proc)_pa_operation_unref; - pContext->pulse.pa_operation_get_state = (ma_proc)_pa_operation_get_state; - pContext->pulse.pa_channel_map_init_extend = (ma_proc)_pa_channel_map_init_extend; - pContext->pulse.pa_channel_map_valid = (ma_proc)_pa_channel_map_valid; - pContext->pulse.pa_channel_map_compatible = (ma_proc)_pa_channel_map_compatible; - pContext->pulse.pa_stream_new = (ma_proc)_pa_stream_new; - pContext->pulse.pa_stream_unref = (ma_proc)_pa_stream_unref; - pContext->pulse.pa_stream_connect_playback = (ma_proc)_pa_stream_connect_playback; - pContext->pulse.pa_stream_connect_record = (ma_proc)_pa_stream_connect_record; - pContext->pulse.pa_stream_disconnect = (ma_proc)_pa_stream_disconnect; - pContext->pulse.pa_stream_get_state = (ma_proc)_pa_stream_get_state; - pContext->pulse.pa_stream_get_sample_spec = (ma_proc)_pa_stream_get_sample_spec; - pContext->pulse.pa_stream_get_channel_map = (ma_proc)_pa_stream_get_channel_map; - pContext->pulse.pa_stream_get_buffer_attr = (ma_proc)_pa_stream_get_buffer_attr; - pContext->pulse.pa_stream_set_buffer_attr = (ma_proc)_pa_stream_set_buffer_attr; - pContext->pulse.pa_stream_get_device_name = (ma_proc)_pa_stream_get_device_name; - pContext->pulse.pa_stream_set_write_callback = (ma_proc)_pa_stream_set_write_callback; - pContext->pulse.pa_stream_set_read_callback = (ma_proc)_pa_stream_set_read_callback; - pContext->pulse.pa_stream_flush = (ma_proc)_pa_stream_flush; - pContext->pulse.pa_stream_drain = (ma_proc)_pa_stream_drain; - pContext->pulse.pa_stream_is_corked = (ma_proc)_pa_stream_is_corked; - pContext->pulse.pa_stream_cork = (ma_proc)_pa_stream_cork; - pContext->pulse.pa_stream_trigger = (ma_proc)_pa_stream_trigger; - pContext->pulse.pa_stream_begin_write = (ma_proc)_pa_stream_begin_write; - pContext->pulse.pa_stream_write = (ma_proc)_pa_stream_write; - pContext->pulse.pa_stream_peek = (ma_proc)_pa_stream_peek; - pContext->pulse.pa_stream_drop = (ma_proc)_pa_stream_drop; - pContext->pulse.pa_stream_writable_size = (ma_proc)_pa_stream_writable_size; - pContext->pulse.pa_stream_readable_size = (ma_proc)_pa_stream_readable_size; -#endif + pMainLoop = ((ma_pa_mainloop_new_proc)pContext->pulse.pa_mainloop_new)(); + if (pMainLoop == NULL) { + return MA_FAILED_TO_INIT_BACKEND; + } - pContext->onUninit = ma_context_uninit__pulse; - pContext->onDeviceIDEqual = ma_context_is_device_id_equal__pulse; - pContext->onEnumDevices = ma_context_enumerate_devices__pulse; - pContext->onGetDeviceInfo = ma_context_get_device_info__pulse; - pContext->onDeviceInit = ma_device_init__pulse; - pContext->onDeviceUninit = ma_device_uninit__pulse; - pContext->onDeviceStart = NULL; - pContext->onDeviceStop = NULL; - pContext->onDeviceMainLoop = ma_device_main_loop__pulse; + pAPI = ((ma_pa_mainloop_get_api_proc)pContext->pulse.pa_mainloop_get_api)(pMainLoop); + if (pAPI == NULL) { + ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); + return MA_FAILED_TO_INIT_BACKEND; + } - if (pConfig->pulse.pApplicationName) { - pContext->pulse.pApplicationName = ma_copy_string(pConfig->pulse.pApplicationName); + pPulseContext = ((ma_pa_context_new_proc)pContext->pulse.pa_context_new)(pAPI, pContext->pulse.pApplicationName); + if (pPulseContext == NULL) { + ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); + return MA_FAILED_TO_INIT_BACKEND; } - if (pConfig->pulse.pServerName) { - pContext->pulse.pServerName = ma_copy_string(pConfig->pulse.pServerName); + + error = ((ma_pa_context_connect_proc)pContext->pulse.pa_context_connect)(pPulseContext, pContext->pulse.pServerName, 0, NULL); + if (error != MA_PA_OK) { + ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)(pPulseContext); + ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); + return ma_result_from_pulse(error); } - pContext->pulse.tryAutoSpawn = pConfig->pulse.tryAutoSpawn; - - /* - Although we have found the libpulse library, it doesn't necessarily mean PulseAudio is useable. We need to initialize - and connect a dummy PulseAudio context to test PulseAudio's usability. - */ - { - ma_pa_mainloop* pMainLoop; - ma_pa_mainloop_api* pAPI; - ma_pa_context* pPulseContext; - int error; - pMainLoop = ((ma_pa_mainloop_new_proc)pContext->pulse.pa_mainloop_new)(); - if (pMainLoop == NULL) { - ma_free(pContext->pulse.pServerName); - ma_free(pContext->pulse.pApplicationName); - #ifndef MA_NO_RUNTIME_LINKING - ma_dlclose(pContext, pContext->pulse.pulseSO); - #endif - return MA_NO_BACKEND; + for (;;) { + ma_pa_context_state_t state = ((ma_pa_context_get_state_proc)pContext->pulse.pa_context_get_state)(pPulseContext); + if (state == MA_PA_CONTEXT_READY) { + break; /* Success. */ } + if (state == MA_PA_CONTEXT_CONNECTING || state == MA_PA_CONTEXT_AUTHORIZING || state == MA_PA_CONTEXT_SETTING_NAME) { + error = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)(pMainLoop, 1, NULL); + if (error < 0) { + result = ma_result_from_pulse(error); + goto done; + } - pAPI = ((ma_pa_mainloop_get_api_proc)pContext->pulse.pa_mainloop_get_api)(pMainLoop); - if (pAPI == NULL) { - ma_free(pContext->pulse.pServerName); - ma_free(pContext->pulse.pApplicationName); - ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); - #ifndef MA_NO_RUNTIME_LINKING - ma_dlclose(pContext, pContext->pulse.pulseSO); - #endif - return MA_NO_BACKEND; +#ifdef MA_DEBUG_OUTPUT + printf("[PulseAudio] pa_context_get_state() returned %d. Waiting.\n", state); +#endif + continue; /* Keep trying. */ } - - pPulseContext = ((ma_pa_context_new_proc)pContext->pulse.pa_context_new)(pAPI, pContext->pulse.pApplicationName); - if (pPulseContext == NULL) { - ma_free(pContext->pulse.pServerName); - ma_free(pContext->pulse.pApplicationName); - ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); - #ifndef MA_NO_RUNTIME_LINKING - ma_dlclose(pContext, pContext->pulse.pulseSO); - #endif - return MA_NO_BACKEND; + if (state == MA_PA_CONTEXT_UNCONNECTED || state == MA_PA_CONTEXT_FAILED || state == MA_PA_CONTEXT_TERMINATED) { +#ifdef MA_DEBUG_OUTPUT + printf("[PulseAudio] pa_context_get_state() returned %d. Failed.\n", state); +#endif + goto done; /* Failed. */ } + } - error = ((ma_pa_context_connect_proc)pContext->pulse.pa_context_connect)(pPulseContext, pContext->pulse.pServerName, 0, NULL); - if (error != MA_PA_OK) { - ma_free(pContext->pulse.pServerName); - ma_free(pContext->pulse.pApplicationName); - ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)(pPulseContext); - ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); - #ifndef MA_NO_RUNTIME_LINKING - ma_dlclose(pContext, pContext->pulse.pulseSO); - #endif - return MA_NO_BACKEND; - } + if (deviceType == ma_device_type_playback) { + pOP = ((ma_pa_context_get_sink_info_by_name_proc)pContext->pulse.pa_context_get_sink_info_by_name)(pPulseContext, pDeviceID->pulse, ma_context_get_device_info_sink_callback__pulse, &callbackData); + } else { + pOP = ((ma_pa_context_get_source_info_by_name_proc)pContext->pulse.pa_context_get_source_info_by_name)(pPulseContext, pDeviceID->pulse, ma_context_get_device_info_source_callback__pulse, &callbackData); + } - ((ma_pa_context_disconnect_proc)pContext->pulse.pa_context_disconnect)(pPulseContext); - ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)(pPulseContext); - ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); + if (pOP != NULL) { + ma_wait_for_operation__pulse(pContext, pMainLoop, pOP); + ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); + } else { + result = MA_ERROR; + goto done; } - return MA_SUCCESS; + if (!callbackData.foundDevice) { + result = MA_NO_DEVICE; + goto done; + } + + +done: + ((ma_pa_context_disconnect_proc)pContext->pulse.pa_context_disconnect)(pPulseContext); + ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)(pPulseContext); + ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); + return result; } -#endif -/****************************************************************************** +static void ma_pulse_device_state_callback(ma_pa_context* pPulseContext, void* pUserData) +{ + ma_device* pDevice; + ma_context* pContext; -JACK Backend + pDevice = (ma_device*)pUserData; + MA_ASSERT(pDevice != NULL); -******************************************************************************/ -#ifdef MA_HAS_JACK + pContext = pDevice->pContext; + MA_ASSERT(pContext != NULL); -/* It is assumed jack.h is available when compile-time linking is being used. */ -#ifdef MA_NO_RUNTIME_LINKING -#include + pDevice->pulse.pulseContextState = ((ma_pa_context_get_state_proc)pContext->pulse.pa_context_get_state)(pPulseContext); +} -typedef jack_nframes_t ma_jack_nframes_t; -typedef jack_options_t ma_jack_options_t; -typedef jack_status_t ma_jack_status_t; -typedef jack_client_t ma_jack_client_t; -typedef jack_port_t ma_jack_port_t; -typedef JackProcessCallback ma_JackProcessCallback; -typedef JackBufferSizeCallback ma_JackBufferSizeCallback; -typedef JackShutdownCallback ma_JackShutdownCallback; -#define MA_JACK_DEFAULT_AUDIO_TYPE JACK_DEFAULT_AUDIO_TYPE -#define ma_JackNoStartServer JackNoStartServer -#define ma_JackPortIsInput JackPortIsInput -#define ma_JackPortIsOutput JackPortIsOutput -#define ma_JackPortIsPhysical JackPortIsPhysical -#else -typedef ma_uint32 ma_jack_nframes_t; -typedef int ma_jack_options_t; -typedef int ma_jack_status_t; -typedef struct ma_jack_client_t ma_jack_client_t; -typedef struct ma_jack_port_t ma_jack_port_t; -typedef int (* ma_JackProcessCallback) (ma_jack_nframes_t nframes, void* arg); -typedef int (* ma_JackBufferSizeCallback)(ma_jack_nframes_t nframes, void* arg); -typedef void (* ma_JackShutdownCallback) (void* arg); -#define MA_JACK_DEFAULT_AUDIO_TYPE "32 bit float mono audio" -#define ma_JackNoStartServer 1 -#define ma_JackPortIsInput 1 -#define ma_JackPortIsOutput 2 -#define ma_JackPortIsPhysical 4 -#endif +void ma_device_sink_info_callback(ma_pa_context* pPulseContext, const ma_pa_sink_info* pInfo, int endOfList, void* pUserData) +{ + ma_pa_sink_info* pInfoOut; -typedef ma_jack_client_t* (* ma_jack_client_open_proc) (const char* client_name, ma_jack_options_t options, ma_jack_status_t* status, ...); -typedef int (* ma_jack_client_close_proc) (ma_jack_client_t* client); -typedef int (* ma_jack_client_name_size_proc) (); -typedef int (* ma_jack_set_process_callback_proc) (ma_jack_client_t* client, ma_JackProcessCallback process_callback, void* arg); -typedef int (* ma_jack_set_buffer_size_callback_proc)(ma_jack_client_t* client, ma_JackBufferSizeCallback bufsize_callback, void* arg); -typedef void (* ma_jack_on_shutdown_proc) (ma_jack_client_t* client, ma_JackShutdownCallback function, void* arg); -typedef ma_jack_nframes_t (* ma_jack_get_sample_rate_proc) (ma_jack_client_t* client); -typedef ma_jack_nframes_t (* ma_jack_get_buffer_size_proc) (ma_jack_client_t* client); -typedef const char** (* ma_jack_get_ports_proc) (ma_jack_client_t* client, const char* port_name_pattern, const char* type_name_pattern, unsigned long flags); -typedef int (* ma_jack_activate_proc) (ma_jack_client_t* client); -typedef int (* ma_jack_deactivate_proc) (ma_jack_client_t* client); -typedef int (* ma_jack_connect_proc) (ma_jack_client_t* client, const char* source_port, const char* destination_port); -typedef ma_jack_port_t* (* ma_jack_port_register_proc) (ma_jack_client_t* client, const char* port_name, const char* port_type, unsigned long flags, unsigned long buffer_size); -typedef const char* (* ma_jack_port_name_proc) (const ma_jack_port_t* port); -typedef void* (* ma_jack_port_get_buffer_proc) (ma_jack_port_t* port, ma_jack_nframes_t nframes); -typedef void (* ma_jack_free_proc) (void* ptr); + if (endOfList > 0) { + return; + } -ma_result ma_context_open_client__jack(ma_context* pContext, ma_jack_client_t** ppClient) + pInfoOut = (ma_pa_sink_info*)pUserData; + MA_ASSERT(pInfoOut != NULL); + + *pInfoOut = *pInfo; + + (void)pPulseContext; /* Unused. */ +} + +static void ma_device_source_info_callback(ma_pa_context* pPulseContext, const ma_pa_source_info* pInfo, int endOfList, void* pUserData) { - size_t maxClientNameSize; - char clientName[256]; - ma_jack_status_t status; - ma_jack_client_t* pClient; + ma_pa_source_info* pInfoOut; - ma_assert(pContext != NULL); - ma_assert(ppClient != NULL); + if (endOfList > 0) { + return; + } - if (ppClient) { - *ppClient = NULL; + pInfoOut = (ma_pa_source_info*)pUserData; + MA_ASSERT(pInfoOut != NULL); + + *pInfoOut = *pInfo; + + (void)pPulseContext; /* Unused. */ +} + +static void ma_device_sink_name_callback(ma_pa_context* pPulseContext, const ma_pa_sink_info* pInfo, int endOfList, void* pUserData) +{ + ma_device* pDevice; + + if (endOfList > 0) { + return; } - maxClientNameSize = ((ma_jack_client_name_size_proc)pContext->jack.jack_client_name_size)(); /* Includes null terminator. */ - ma_strncpy_s(clientName, ma_min(sizeof(clientName), maxClientNameSize), (pContext->jack.pClientName != NULL) ? pContext->jack.pClientName : "miniaudio", (size_t)-1); + pDevice = (ma_device*)pUserData; + MA_ASSERT(pDevice != NULL); - pClient = ((ma_jack_client_open_proc)pContext->jack.jack_client_open)(clientName, (pContext->jack.tryStartServer) ? 0 : ma_JackNoStartServer, &status, NULL); - if (pClient == NULL) { - return MA_FAILED_TO_OPEN_BACKEND_DEVICE; + ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), pInfo->description, (size_t)-1); + + (void)pPulseContext; /* Unused. */ +} + +static void ma_device_source_name_callback(ma_pa_context* pPulseContext, const ma_pa_source_info* pInfo, int endOfList, void* pUserData) +{ + ma_device* pDevice; + + if (endOfList > 0) { + return; } - if (ppClient) { - *ppClient = pClient; + pDevice = (ma_device*)pUserData; + MA_ASSERT(pDevice != NULL); + + ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), pInfo->description, (size_t)-1); + + (void)pPulseContext; /* Unused. */ +} + +static void ma_device_uninit__pulse(ma_device* pDevice) +{ + ma_context* pContext; + + MA_ASSERT(pDevice != NULL); + + pContext = pDevice->pContext; + MA_ASSERT(pContext != NULL); + + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + ((ma_pa_stream_disconnect_proc)pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamCapture); + ((ma_pa_stream_unref_proc)pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamCapture); + } + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + ((ma_pa_stream_disconnect_proc)pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); + ((ma_pa_stream_unref_proc)pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); } - return MA_SUCCESS; + ((ma_pa_context_disconnect_proc)pContext->pulse.pa_context_disconnect)((ma_pa_context*)pDevice->pulse.pPulseContext); + ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)((ma_pa_context*)pDevice->pulse.pPulseContext); + ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)pDevice->pulse.pMainLoop); } -ma_bool32 ma_context_is_device_id_equal__jack(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) +static ma_pa_buffer_attr ma_device__pa_buffer_attr_new(ma_uint32 periodSizeInFrames, ma_uint32 periods, const ma_pa_sample_spec* ss) { - ma_assert(pContext != NULL); - ma_assert(pID0 != NULL); - ma_assert(pID1 != NULL); - (void)pContext; + ma_pa_buffer_attr attr; + attr.maxlength = periodSizeInFrames * periods * ma_get_bytes_per_frame(ma_format_from_pulse(ss->format), ss->channels); + attr.tlength = attr.maxlength / periods; + attr.prebuf = (ma_uint32)-1; + attr.minreq = (ma_uint32)-1; + attr.fragsize = attr.maxlength / periods; + + return attr; +} + +static ma_pa_stream* ma_device__pa_stream_new__pulse(ma_device* pDevice, const char* pStreamName, const ma_pa_sample_spec* ss, const ma_pa_channel_map* cmap) +{ + static int g_StreamCounter = 0; + char actualStreamName[256]; + + if (pStreamName != NULL) { + ma_strncpy_s(actualStreamName, sizeof(actualStreamName), pStreamName, (size_t)-1); + } else { + ma_strcpy_s(actualStreamName, sizeof(actualStreamName), "miniaudio:"); + ma_itoa_s(g_StreamCounter, actualStreamName + 8, sizeof(actualStreamName)-8, 10); /* 8 = strlen("miniaudio:") */ + } + g_StreamCounter += 1; - return pID0->jack == pID1->jack; + return ((ma_pa_stream_new_proc)pDevice->pContext->pulse.pa_stream_new)((ma_pa_context*)pDevice->pulse.pPulseContext, actualStreamName, ss, cmap); } -ma_result ma_context_enumerate_devices__jack(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +static ma_result ma_device_init__pulse(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) { - ma_bool32 cbResult = MA_TRUE; + ma_result result = MA_SUCCESS; + int error = 0; + const char* devPlayback = NULL; + const char* devCapture = NULL; + ma_uint32 periodSizeInMilliseconds; + ma_pa_sink_info sinkInfo; + ma_pa_source_info sourceInfo; + ma_pa_operation* pOP = NULL; + ma_pa_sample_spec ss; + ma_pa_channel_map cmap; + ma_pa_buffer_attr attr; + const ma_pa_sample_spec* pActualSS = NULL; + const ma_pa_channel_map* pActualCMap = NULL; + const ma_pa_buffer_attr* pActualAttr = NULL; + ma_uint32 iChannel; + ma_pa_stream_flags_t streamFlags; - ma_assert(pContext != NULL); - ma_assert(callback != NULL); + MA_ASSERT(pDevice != NULL); + MA_ZERO_OBJECT(&pDevice->pulse); - /* Playback. */ - if (cbResult) { - ma_device_info deviceInfo; - ma_zero_object(&deviceInfo); - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); - cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); + if (pConfig->deviceType == ma_device_type_loopback) { + return MA_DEVICE_TYPE_NOT_SUPPORTED; } - /* Capture. */ - if (cbResult) { - ma_device_info deviceInfo; - ma_zero_object(&deviceInfo); - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); - cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); + /* No exclusive mode with the PulseAudio backend. */ + if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) || + ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) { + return MA_SHARE_MODE_NOT_SUPPORTED; } - return MA_SUCCESS; -} - -ma_result ma_context_get_device_info__jack(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) -{ - ma_jack_client_t* pClient; - ma_result result; - const char** ppPorts; - - ma_assert(pContext != NULL); + if ((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.pDeviceID != NULL) { + devPlayback = pConfig->playback.pDeviceID->pulse; + } + if ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.pDeviceID != NULL) { + devCapture = pConfig->capture.pDeviceID->pulse; + } - /* No exclusive mode with the JACK backend. */ - if (shareMode == ma_share_mode_exclusive) { - return MA_SHARE_MODE_NOT_SUPPORTED; + periodSizeInMilliseconds = pConfig->periodSizeInMilliseconds; + if (periodSizeInMilliseconds == 0) { + periodSizeInMilliseconds = ma_calculate_buffer_size_in_milliseconds_from_frames(pConfig->periodSizeInFrames, pConfig->sampleRate); } - if (pDeviceID != NULL && pDeviceID->jack != 0) { - return MA_NO_DEVICE; /* Don't know the device. */ + pDevice->pulse.pMainLoop = ((ma_pa_mainloop_new_proc)pContext->pulse.pa_mainloop_new)(); + if (pDevice->pulse.pMainLoop == NULL) { + result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create main loop for device.", MA_FAILED_TO_INIT_BACKEND); + goto on_error0; } - /* Name / Description */ - if (deviceType == ma_device_type_playback) { - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); - } else { - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); + pDevice->pulse.pAPI = ((ma_pa_mainloop_get_api_proc)pContext->pulse.pa_mainloop_get_api)((ma_pa_mainloop*)pDevice->pulse.pMainLoop); + if (pDevice->pulse.pAPI == NULL) { + result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to retrieve PulseAudio main loop.", MA_FAILED_TO_INIT_BACKEND); + goto on_error1; } - /* Jack only supports f32 and has a specific channel count and sample rate. */ - pDeviceInfo->formatCount = 1; - pDeviceInfo->formats[0] = ma_format_f32; + pDevice->pulse.pPulseContext = ((ma_pa_context_new_proc)pContext->pulse.pa_context_new)((ma_pa_mainloop_api*)pDevice->pulse.pAPI, pContext->pulse.pApplicationName); + if (pDevice->pulse.pPulseContext == NULL) { + result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create PulseAudio context for device.", MA_FAILED_TO_INIT_BACKEND); + goto on_error1; + } - /* The channel count and sample rate can only be determined by opening the device. */ - result = ma_context_open_client__jack(pContext, &pClient); - if (result != MA_SUCCESS) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[JACK] Failed to open client.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + error = ((ma_pa_context_connect_proc)pContext->pulse.pa_context_connect)((ma_pa_context*)pDevice->pulse.pPulseContext, pContext->pulse.pServerName, (pContext->pulse.tryAutoSpawn) ? 0 : MA_PA_CONTEXT_NOAUTOSPAWN, NULL); + if (error != MA_PA_OK) { + result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to connect PulseAudio context.", ma_result_from_pulse(error)); + goto on_error2; } - pDeviceInfo->minSampleRate = ((ma_jack_get_sample_rate_proc)pContext->jack.jack_get_sample_rate)((ma_jack_client_t*)pClient); - pDeviceInfo->maxSampleRate = pDeviceInfo->minSampleRate; - pDeviceInfo->minChannels = 0; - pDeviceInfo->maxChannels = 0; + pDevice->pulse.pulseContextState = MA_PA_CONTEXT_UNCONNECTED; + ((ma_pa_context_set_state_callback_proc)pContext->pulse.pa_context_set_state_callback)((ma_pa_context*)pDevice->pulse.pPulseContext, ma_pulse_device_state_callback, pDevice); - ppPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pClient, NULL, NULL, ma_JackPortIsPhysical | ((deviceType == ma_device_type_playback) ? ma_JackPortIsInput : ma_JackPortIsOutput)); - if (ppPorts == NULL) { - ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pClient); - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[JACK] Failed to query physical ports.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } + /* Wait for PulseAudio to get itself ready before returning. */ + for (;;) { + if (pDevice->pulse.pulseContextState == MA_PA_CONTEXT_READY) { + break; + } - while (ppPorts[pDeviceInfo->minChannels] != NULL) { - pDeviceInfo->minChannels += 1; - pDeviceInfo->maxChannels += 1; - } + /* An error may have occurred. */ + if (pDevice->pulse.pulseContextState == MA_PA_CONTEXT_FAILED || pDevice->pulse.pulseContextState == MA_PA_CONTEXT_TERMINATED) { + result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] An error occurred while connecting the PulseAudio context.", MA_ERROR); + goto on_error3; + } - ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppPorts); - ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pClient); + error = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pDevice->pulse.pMainLoop, 1, NULL); + if (error < 0) { + result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] The PulseAudio main loop returned an error while connecting the PulseAudio context.", ma_result_from_pulse(error)); + goto on_error3; + } + } - (void)pContext; - return MA_SUCCESS; -} + if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { + pOP = ((ma_pa_context_get_source_info_by_name_proc)pContext->pulse.pa_context_get_source_info_by_name)((ma_pa_context*)pDevice->pulse.pPulseContext, devCapture, ma_device_source_info_callback, &sourceInfo); + if (pOP != NULL) { + ma_device__wait_for_operation__pulse(pDevice, pOP); + ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); + } else { + result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to retrieve source info for capture device.", ma_result_from_pulse(error)); + goto on_error3; + } + ss = sourceInfo.sample_spec; + cmap = sourceInfo.channel_map; -void ma_device_uninit__jack(ma_device* pDevice) -{ - ma_context* pContext; + pDevice->capture.internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(periodSizeInMilliseconds, ss.rate); + pDevice->capture.internalPeriods = pConfig->periods; - ma_assert(pDevice != NULL); + attr = ma_device__pa_buffer_attr_new(pDevice->capture.internalPeriodSizeInFrames, pConfig->periods, &ss); + #ifdef MA_DEBUG_OUTPUT + printf("[PulseAudio] Capture attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; internalPeriodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDevice->capture.internalPeriodSizeInFrames); + #endif - pContext = pDevice->pContext; - ma_assert(pContext != NULL); + pDevice->pulse.pStreamCapture = ma_device__pa_stream_new__pulse(pDevice, pConfig->pulse.pStreamNameCapture, &ss, &cmap); + if (pDevice->pulse.pStreamCapture == NULL) { + result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create PulseAudio capture stream.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + goto on_error3; + } - if (pDevice->jack.pClient != NULL) { - ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pDevice->jack.pClient); - } + streamFlags = MA_PA_STREAM_START_CORKED | MA_PA_STREAM_FIX_FORMAT | MA_PA_STREAM_FIX_RATE | MA_PA_STREAM_FIX_CHANNELS; + if (devCapture != NULL) { + streamFlags |= MA_PA_STREAM_DONT_MOVE; + } - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - ma_free(pDevice->jack.pIntermediaryBufferCapture); - } + error = ((ma_pa_stream_connect_record_proc)pContext->pulse.pa_stream_connect_record)((ma_pa_stream*)pDevice->pulse.pStreamCapture, devCapture, &attr, streamFlags); + if (error != MA_PA_OK) { + result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to connect PulseAudio capture stream.", ma_result_from_pulse(error)); + goto on_error4; + } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - ma_free(pDevice->jack.pIntermediaryBufferPlayback); - } + while (((ma_pa_stream_get_state_proc)pContext->pulse.pa_stream_get_state)((ma_pa_stream*)pDevice->pulse.pStreamCapture) != MA_PA_STREAM_READY) { + error = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pDevice->pulse.pMainLoop, 1, NULL); + if (error < 0) { + result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] The PulseAudio main loop returned an error while connecting the PulseAudio capture stream.", ma_result_from_pulse(error)); + goto on_error5; + } + } - if (pDevice->type == ma_device_type_duplex) { - ma_pcm_rb_uninit(&pDevice->jack.duplexRB); - } -} + /* Internal format. */ + pActualSS = ((ma_pa_stream_get_sample_spec_proc)pContext->pulse.pa_stream_get_sample_spec)((ma_pa_stream*)pDevice->pulse.pStreamCapture); + if (pActualSS != NULL) { + /* If anything has changed between the requested and the actual sample spec, we need to update the buffer. */ + if (ss.format != pActualSS->format || ss.channels != pActualSS->channels || ss.rate != pActualSS->rate) { + attr = ma_device__pa_buffer_attr_new(pDevice->capture.internalPeriodSizeInFrames, pConfig->periods, pActualSS); -void ma_device__jack_shutdown_callback(void* pUserData) -{ - /* JACK died. Stop the device. */ - ma_device* pDevice = (ma_device*)pUserData; - ma_assert(pDevice != NULL); + pOP = ((ma_pa_stream_set_buffer_attr_proc)pContext->pulse.pa_stream_set_buffer_attr)((ma_pa_stream*)pDevice->pulse.pStreamCapture, &attr, NULL, NULL); + if (pOP != NULL) { + ma_device__wait_for_operation__pulse(pDevice, pOP); + ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); + } + } - ma_device_stop(pDevice); -} + ss = *pActualSS; + } -int ma_device__jack_buffer_size_callback(ma_jack_nframes_t frameCount, void* pUserData) -{ - ma_device* pDevice = (ma_device*)pUserData; - ma_assert(pDevice != NULL); + pDevice->capture.internalFormat = ma_format_from_pulse(ss.format); + pDevice->capture.internalChannels = ss.channels; + pDevice->capture.internalSampleRate = ss.rate; - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - float* pNewBuffer = (float*)ma_realloc(pDevice->jack.pIntermediaryBufferCapture, frameCount * (pDevice->capture.internalChannels * ma_get_bytes_per_sample(pDevice->capture.internalFormat))); - if (pNewBuffer == NULL) { - return MA_OUT_OF_MEMORY; + /* Internal channel map. */ + pActualCMap = ((ma_pa_stream_get_channel_map_proc)pContext->pulse.pa_stream_get_channel_map)((ma_pa_stream*)pDevice->pulse.pStreamCapture); + if (pActualCMap != NULL) { + cmap = *pActualCMap; + } + for (iChannel = 0; iChannel < pDevice->capture.internalChannels; ++iChannel) { + pDevice->capture.internalChannelMap[iChannel] = ma_channel_position_from_pulse(cmap.map[iChannel]); } - pDevice->jack.pIntermediaryBufferCapture = pNewBuffer; - pDevice->playback.internalBufferSizeInFrames = frameCount * pDevice->capture.internalPeriods; - } - - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - float* pNewBuffer = (float*)ma_realloc(pDevice->jack.pIntermediaryBufferPlayback, frameCount * (pDevice->playback.internalChannels * ma_get_bytes_per_sample(pDevice->playback.internalFormat))); - if (pNewBuffer == NULL) { - return MA_OUT_OF_MEMORY; + /* Buffer. */ + pActualAttr = ((ma_pa_stream_get_buffer_attr_proc)pContext->pulse.pa_stream_get_buffer_attr)((ma_pa_stream*)pDevice->pulse.pStreamCapture); + if (pActualAttr != NULL) { + attr = *pActualAttr; } + pDevice->capture.internalPeriods = attr.maxlength / attr.fragsize; + pDevice->capture.internalPeriodSizeInFrames = attr.maxlength / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels) / pDevice->capture.internalPeriods; + #ifdef MA_DEBUG_OUTPUT + printf("[PulseAudio] Capture actual attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; internalPeriodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDevice->capture.internalPeriodSizeInFrames); + #endif - pDevice->jack.pIntermediaryBufferPlayback = pNewBuffer; - pDevice->playback.internalBufferSizeInFrames = frameCount * pDevice->playback.internalPeriods; + /* Name. */ + devCapture = ((ma_pa_stream_get_device_name_proc)pContext->pulse.pa_stream_get_device_name)((ma_pa_stream*)pDevice->pulse.pStreamCapture); + if (devCapture != NULL) { + ma_pa_operation* pOP = ((ma_pa_context_get_source_info_by_name_proc)pContext->pulse.pa_context_get_source_info_by_name)((ma_pa_context*)pDevice->pulse.pPulseContext, devCapture, ma_device_source_name_callback, pDevice); + if (pOP != NULL) { + ma_device__wait_for_operation__pulse(pDevice, pOP); + ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); + } + } } - return 0; -} - -int ma_device__jack_process_callback(ma_jack_nframes_t frameCount, void* pUserData) -{ - ma_device* pDevice; - ma_context* pContext; - ma_uint32 iChannel; - - pDevice = (ma_device*)pUserData; - ma_assert(pDevice != NULL); + if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { + pOP = ((ma_pa_context_get_sink_info_by_name_proc)pContext->pulse.pa_context_get_sink_info_by_name)((ma_pa_context*)pDevice->pulse.pPulseContext, devPlayback, ma_device_sink_info_callback, &sinkInfo); + if (pOP != NULL) { + ma_device__wait_for_operation__pulse(pDevice, pOP); + ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); + } else { + result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to retrieve sink info for playback device.", ma_result_from_pulse(error)); + goto on_error3; + } - pContext = pDevice->pContext; - ma_assert(pContext != NULL); + ss = sinkInfo.sample_spec; + cmap = sinkInfo.channel_map; - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - /* Channels need to be interleaved. */ - for (iChannel = 0; iChannel < pDevice->capture.internalChannels; ++iChannel) { - const float* pSrc = (const float*)((ma_jack_port_get_buffer_proc)pContext->jack.jack_port_get_buffer)((ma_jack_port_t*)pDevice->jack.pPortsCapture[iChannel], frameCount); - if (pSrc != NULL) { - float* pDst = pDevice->jack.pIntermediaryBufferCapture + iChannel; - ma_jack_nframes_t iFrame; - for (iFrame = 0; iFrame < frameCount; ++iFrame) { - *pDst = *pSrc; + pDevice->playback.internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(periodSizeInMilliseconds, ss.rate); + pDevice->playback.internalPeriods = pConfig->periods; - pDst += pDevice->capture.internalChannels; - pSrc += 1; - } - } + attr = ma_device__pa_buffer_attr_new(pDevice->playback.internalPeriodSizeInFrames, pConfig->periods, &ss); + #ifdef MA_DEBUG_OUTPUT + printf("[PulseAudio] Playback attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; internalPeriodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDevice->playback.internalPeriodSizeInFrames); + #endif + + pDevice->pulse.pStreamPlayback = ma_device__pa_stream_new__pulse(pDevice, pConfig->pulse.pStreamNamePlayback, &ss, &cmap); + if (pDevice->pulse.pStreamPlayback == NULL) { + result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create PulseAudio playback stream.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + goto on_error3; } - if (pDevice->type == ma_device_type_duplex) { - ma_device__handle_duplex_callback_capture(pDevice, frameCount, pDevice->jack.pIntermediaryBufferCapture, &pDevice->jack.duplexRB); - } else { - ma_device__send_frames_to_client(pDevice, frameCount, pDevice->jack.pIntermediaryBufferCapture); + streamFlags = MA_PA_STREAM_START_CORKED | MA_PA_STREAM_FIX_FORMAT | MA_PA_STREAM_FIX_RATE | MA_PA_STREAM_FIX_CHANNELS; + if (devPlayback != NULL) { + streamFlags |= MA_PA_STREAM_DONT_MOVE; } - } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - if (pDevice->type == ma_device_type_duplex) { - ma_device__handle_duplex_callback_playback(pDevice, frameCount, pDevice->jack.pIntermediaryBufferPlayback, &pDevice->jack.duplexRB); - } else { - ma_device__read_frames_from_client(pDevice, frameCount, pDevice->jack.pIntermediaryBufferPlayback); + error = ((ma_pa_stream_connect_playback_proc)pContext->pulse.pa_stream_connect_playback)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, devPlayback, &attr, streamFlags, NULL, NULL); + if (error != MA_PA_OK) { + result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to connect PulseAudio playback stream.", ma_result_from_pulse(error)); + goto on_error6; } - /* Channels need to be deinterleaved. */ - for (iChannel = 0; iChannel < pDevice->playback.internalChannels; ++iChannel) { - float* pDst = (float*)((ma_jack_port_get_buffer_proc)pContext->jack.jack_port_get_buffer)((ma_jack_port_t*)pDevice->jack.pPortsPlayback[iChannel], frameCount); - if (pDst != NULL) { - const float* pSrc = pDevice->jack.pIntermediaryBufferPlayback + iChannel; - ma_jack_nframes_t iFrame; - for (iFrame = 0; iFrame < frameCount; ++iFrame) { - *pDst = *pSrc; + while (((ma_pa_stream_get_state_proc)pContext->pulse.pa_stream_get_state)((ma_pa_stream*)pDevice->pulse.pStreamPlayback) != MA_PA_STREAM_READY) { + error = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pDevice->pulse.pMainLoop, 1, NULL); + if (error < 0) { + result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] The PulseAudio main loop returned an error while connecting the PulseAudio playback stream.", ma_result_from_pulse(error)); + goto on_error7; + } + } - pDst += 1; - pSrc += pDevice->playback.internalChannels; + /* Internal format. */ + pActualSS = ((ma_pa_stream_get_sample_spec_proc)pContext->pulse.pa_stream_get_sample_spec)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); + if (pActualSS != NULL) { + /* If anything has changed between the requested and the actual sample spec, we need to update the buffer. */ + if (ss.format != pActualSS->format || ss.channels != pActualSS->channels || ss.rate != pActualSS->rate) { + attr = ma_device__pa_buffer_attr_new(pDevice->playback.internalPeriodSizeInFrames, pConfig->periods, pActualSS); + + pOP = ((ma_pa_stream_set_buffer_attr_proc)pContext->pulse.pa_stream_set_buffer_attr)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, &attr, NULL, NULL); + if (pOP != NULL) { + ma_device__wait_for_operation__pulse(pDevice, pOP); + ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); } } + + ss = *pActualSS; } - } - return 0; -} + pDevice->playback.internalFormat = ma_format_from_pulse(ss.format); + pDevice->playback.internalChannels = ss.channels; + pDevice->playback.internalSampleRate = ss.rate; -ma_result ma_device_init__jack(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) -{ - ma_result result; - ma_uint32 periods; - ma_uint32 bufferSizeInFrames; + /* Internal channel map. */ + pActualCMap = ((ma_pa_stream_get_channel_map_proc)pContext->pulse.pa_stream_get_channel_map)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); + if (pActualCMap != NULL) { + cmap = *pActualCMap; + } + for (iChannel = 0; iChannel < pDevice->playback.internalChannels; ++iChannel) { + pDevice->playback.internalChannelMap[iChannel] = ma_channel_position_from_pulse(cmap.map[iChannel]); + } - ma_assert(pContext != NULL); - ma_assert(pConfig != NULL); - ma_assert(pDevice != NULL); + /* Buffer. */ + pActualAttr = ((ma_pa_stream_get_buffer_attr_proc)pContext->pulse.pa_stream_get_buffer_attr)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); + if (pActualAttr != NULL) { + attr = *pActualAttr; + } + pDevice->playback.internalPeriods = attr.maxlength / attr.tlength; + pDevice->playback.internalPeriodSizeInFrames = attr.maxlength / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels) / pDevice->playback.internalPeriods; + #ifdef MA_DEBUG_OUTPUT + printf("[PulseAudio] Playback actual attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; internalPeriodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDevice->playback.internalPeriodSizeInFrames); + #endif - if (pConfig->deviceType == ma_device_type_loopback) { - return MA_DEVICE_TYPE_NOT_SUPPORTED; + /* Name. */ + devPlayback = ((ma_pa_stream_get_device_name_proc)pContext->pulse.pa_stream_get_device_name)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); + if (devPlayback != NULL) { + ma_pa_operation* pOP = ((ma_pa_context_get_sink_info_by_name_proc)pContext->pulse.pa_context_get_sink_info_by_name)((ma_pa_context*)pDevice->pulse.pPulseContext, devPlayback, ma_device_sink_name_callback, pDevice); + if (pOP != NULL) { + ma_device__wait_for_operation__pulse(pDevice, pOP); + ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); + } + } } - /* Only supporting default devices with JACK. */ - if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.pDeviceID != NULL && pConfig->playback.pDeviceID->jack != 0) || - ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.pDeviceID != NULL && pConfig->capture.pDeviceID->jack != 0)) { - return MA_NO_DEVICE; - } + return MA_SUCCESS; - /* No exclusive mode with the JACK backend. */ - if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) || - ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) { - return MA_SHARE_MODE_NOT_SUPPORTED; - } - /* Open the client. */ - result = ma_context_open_client__jack(pContext, (ma_jack_client_t**)&pDevice->jack.pClient); - if (result != MA_SUCCESS) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to open client.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); +on_error7: + if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { + ((ma_pa_stream_disconnect_proc)pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); } - - /* Callbacks. */ - if (((ma_jack_set_process_callback_proc)pContext->jack.jack_set_process_callback)((ma_jack_client_t*)pDevice->jack.pClient, ma_device__jack_process_callback, pDevice) != 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to set process callback.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); +on_error6: + if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { + ((ma_pa_stream_unref_proc)pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); } - if (((ma_jack_set_buffer_size_callback_proc)pContext->jack.jack_set_buffer_size_callback)((ma_jack_client_t*)pDevice->jack.pClient, ma_device__jack_buffer_size_callback, pDevice) != 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to set buffer size callback.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); +on_error5: + if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { + ((ma_pa_stream_disconnect_proc)pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamCapture); } - - ((ma_jack_on_shutdown_proc)pContext->jack.jack_on_shutdown)((ma_jack_client_t*)pDevice->jack.pClient, ma_device__jack_shutdown_callback, pDevice); - - - /* The buffer size in frames can change. */ - periods = pConfig->periods; - bufferSizeInFrames = ((ma_jack_get_buffer_size_proc)pContext->jack.jack_get_buffer_size)((ma_jack_client_t*)pDevice->jack.pClient) * periods; - +on_error4: if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { - const char** ppPorts; - - pDevice->capture.internalFormat = ma_format_f32; - pDevice->capture.internalChannels = 0; - pDevice->capture.internalSampleRate = ((ma_jack_get_sample_rate_proc)pContext->jack.jack_get_sample_rate)((ma_jack_client_t*)pDevice->jack.pClient); - ma_get_standard_channel_map(ma_standard_channel_map_alsa, pDevice->capture.internalChannels, pDevice->capture.internalChannelMap); - - ppPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, NULL, ma_JackPortIsPhysical | ma_JackPortIsOutput); - if (ppPorts == NULL) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to query physical ports.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } + ((ma_pa_stream_unref_proc)pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamCapture); + } +on_error3: ((ma_pa_context_disconnect_proc)pContext->pulse.pa_context_disconnect)((ma_pa_context*)pDevice->pulse.pPulseContext); +on_error2: ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)((ma_pa_context*)pDevice->pulse.pPulseContext); +on_error1: ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)pDevice->pulse.pMainLoop); +on_error0: + return result; +} - while (ppPorts[pDevice->capture.internalChannels] != NULL) { - char name[64]; - ma_strcpy_s(name, sizeof(name), "capture"); - ma_itoa_s((int)pDevice->capture.internalChannels, name+7, sizeof(name)-7, 10); /* 7 = length of "capture" */ - pDevice->jack.pPortsCapture[pDevice->capture.internalChannels] = ((ma_jack_port_register_proc)pContext->jack.jack_port_register)((ma_jack_client_t*)pDevice->jack.pClient, name, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsInput, 0); - if (pDevice->jack.pPortsCapture[pDevice->capture.internalChannels] == NULL) { - ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppPorts); - ma_device_uninit__jack(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to register ports.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } +static void ma_pulse_operation_complete_callback(ma_pa_stream* pStream, int success, void* pUserData) +{ + ma_bool32* pIsSuccessful = (ma_bool32*)pUserData; + MA_ASSERT(pIsSuccessful != NULL); - pDevice->capture.internalChannels += 1; - } + *pIsSuccessful = (ma_bool32)success; - ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppPorts); + (void)pStream; /* Unused. */ +} - pDevice->capture.internalBufferSizeInFrames = bufferSizeInFrames; - pDevice->capture.internalPeriods = periods; +static ma_result ma_device__cork_stream__pulse(ma_device* pDevice, ma_device_type deviceType, int cork) +{ + ma_context* pContext = pDevice->pContext; + ma_bool32 wasSuccessful; + ma_pa_stream* pStream; + ma_pa_operation* pOP; + ma_result result; - pDevice->jack.pIntermediaryBufferCapture = (float*)ma_malloc((pDevice->capture.internalBufferSizeInFrames/pDevice->capture.internalPeriods) * (pDevice->capture.internalChannels * ma_get_bytes_per_sample(pDevice->capture.internalFormat))); - if (pDevice->jack.pIntermediaryBufferCapture == NULL) { - ma_device_uninit__jack(pDevice); - return MA_OUT_OF_MEMORY; - } + /* This should not be called with a duplex device type. */ + if (deviceType == ma_device_type_duplex) { + return MA_INVALID_ARGS; } - if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { - const char** ppPorts; + wasSuccessful = MA_FALSE; - pDevice->playback.internalFormat = ma_format_f32; - pDevice->playback.internalChannels = 0; - pDevice->playback.internalSampleRate = ((ma_jack_get_sample_rate_proc)pContext->jack.jack_get_sample_rate)((ma_jack_client_t*)pDevice->jack.pClient); - ma_get_standard_channel_map(ma_standard_channel_map_alsa, pDevice->playback.internalChannels, pDevice->playback.internalChannelMap); + pStream = (ma_pa_stream*)((deviceType == ma_device_type_capture) ? pDevice->pulse.pStreamCapture : pDevice->pulse.pStreamPlayback); + MA_ASSERT(pStream != NULL); - ppPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, NULL, ma_JackPortIsPhysical | ma_JackPortIsInput); - if (ppPorts == NULL) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to query physical ports.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } + pOP = ((ma_pa_stream_cork_proc)pContext->pulse.pa_stream_cork)(pStream, cork, ma_pulse_operation_complete_callback, &wasSuccessful); + if (pOP == NULL) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to cork PulseAudio stream.", (cork == 0) ? MA_FAILED_TO_START_BACKEND_DEVICE : MA_FAILED_TO_STOP_BACKEND_DEVICE); + } - while (ppPorts[pDevice->playback.internalChannels] != NULL) { - char name[64]; - ma_strcpy_s(name, sizeof(name), "playback"); - ma_itoa_s((int)pDevice->playback.internalChannels, name+8, sizeof(name)-8, 10); /* 8 = length of "playback" */ + result = ma_device__wait_for_operation__pulse(pDevice, pOP); + ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP); - pDevice->jack.pPortsPlayback[pDevice->playback.internalChannels] = ((ma_jack_port_register_proc)pContext->jack.jack_port_register)((ma_jack_client_t*)pDevice->jack.pClient, name, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsOutput, 0); - if (pDevice->jack.pPortsPlayback[pDevice->playback.internalChannels] == NULL) { - ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppPorts); - ma_device_uninit__jack(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to register ports.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } + if (result != MA_SUCCESS) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] An error occurred while waiting for the PulseAudio stream to cork.", result); + } - pDevice->playback.internalChannels += 1; + if (!wasSuccessful) { + if (cork) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to stop PulseAudio stream.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + } else { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to start PulseAudio stream.", MA_FAILED_TO_START_BACKEND_DEVICE); } + } - ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppPorts); + return MA_SUCCESS; +} - pDevice->playback.internalBufferSizeInFrames = bufferSizeInFrames; - pDevice->playback.internalPeriods = periods; +static ma_result ma_device_stop__pulse(ma_device* pDevice) +{ + ma_result result; + ma_bool32 wasSuccessful; + ma_pa_operation* pOP; - pDevice->jack.pIntermediaryBufferPlayback = (float*)ma_malloc((pDevice->playback.internalBufferSizeInFrames/pDevice->playback.internalPeriods) * (pDevice->playback.internalChannels * ma_get_bytes_per_sample(pDevice->playback.internalFormat))); - if (pDevice->jack.pIntermediaryBufferPlayback == NULL) { - ma_device_uninit__jack(pDevice); - return MA_OUT_OF_MEMORY; + MA_ASSERT(pDevice != NULL); + + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + result = ma_device__cork_stream__pulse(pDevice, ma_device_type_capture, 1); + if (result != MA_SUCCESS) { + return result; } } - if (pDevice->type == ma_device_type_duplex) { - ma_uint32 rbSizeInFrames = (ma_uint32)ma_calculate_frame_count_after_src(pDevice->sampleRate, pDevice->capture.internalSampleRate, pDevice->capture.internalBufferSizeInFrames); - result = ma_pcm_rb_init(pDevice->capture.format, pDevice->capture.channels, rbSizeInFrames, NULL, &pDevice->jack.duplexRB); - if (result != MA_SUCCESS) { - ma_device_uninit__jack(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to initialize ring buffer.", result); + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + /* The stream needs to be drained if it's a playback device. */ + pOP = ((ma_pa_stream_drain_proc)pDevice->pContext->pulse.pa_stream_drain)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, ma_pulse_operation_complete_callback, &wasSuccessful); + if (pOP != NULL) { + ma_device__wait_for_operation__pulse(pDevice, pOP); + ((ma_pa_operation_unref_proc)pDevice->pContext->pulse.pa_operation_unref)(pOP); } - /* We need a period to act as a buffer for cases where the playback and capture device's end up desyncing. */ - { - ma_uint32 marginSizeInFrames = rbSizeInFrames / pDevice->capture.internalPeriods; - void* pMarginData; - ma_pcm_rb_acquire_write(&pDevice->jack.duplexRB, &marginSizeInFrames, &pMarginData); - { - ma_zero_memory(pMarginData, marginSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels)); - } - ma_pcm_rb_commit_write(&pDevice->jack.duplexRB, marginSizeInFrames, pMarginData); + result = ma_device__cork_stream__pulse(pDevice, ma_device_type_playback, 1); + if (result != MA_SUCCESS) { + return result; } } return MA_SUCCESS; } - -ma_result ma_device_start__jack(ma_device* pDevice) +static ma_result ma_device_write__pulse(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) { - ma_context* pContext = pDevice->pContext; - int resultJACK; - size_t i; + ma_uint32 totalFramesWritten; - resultJACK = ((ma_jack_activate_proc)pContext->jack.jack_activate)((ma_jack_client_t*)pDevice->jack.pClient); - if (resultJACK != 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to activate the JACK client.", MA_FAILED_TO_START_BACKEND_DEVICE); + MA_ASSERT(pDevice != NULL); + MA_ASSERT(pPCMFrames != NULL); + MA_ASSERT(frameCount > 0); + + if (pFramesWritten != NULL) { + *pFramesWritten = 0; } - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - const char** ppServerPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, NULL, ma_JackPortIsPhysical | ma_JackPortIsOutput); - if (ppServerPorts == NULL) { - ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to retrieve physical ports.", MA_ERROR); + totalFramesWritten = 0; + while (totalFramesWritten < frameCount) { + if (ma_device__get_state(pDevice) != MA_STATE_STARTED) { + return MA_DEVICE_NOT_STARTED; } - for (i = 0; ppServerPorts[i] != NULL; ++i) { - const char* pServerPort = ppServerPorts[i]; - const char* pClientPort = ((ma_jack_port_name_proc)pContext->jack.jack_port_name)((ma_jack_port_t*)pDevice->jack.pPortsCapture[i]); + /* Place the data into the mapped buffer if we have one. */ + if (pDevice->pulse.pMappedBufferPlayback != NULL && pDevice->pulse.mappedBufferFramesRemainingPlayback > 0) { + ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + ma_uint32 mappedBufferFramesConsumed = pDevice->pulse.mappedBufferFramesCapacityPlayback - pDevice->pulse.mappedBufferFramesRemainingPlayback; - resultJACK = ((ma_jack_connect_proc)pContext->jack.jack_connect)((ma_jack_client_t*)pDevice->jack.pClient, pServerPort, pClientPort); - if (resultJACK != 0) { - ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts); - ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to connect ports.", MA_ERROR); + void* pDst = (ma_uint8*)pDevice->pulse.pMappedBufferPlayback + (mappedBufferFramesConsumed * bpf); + const void* pSrc = (const ma_uint8*)pPCMFrames + (totalFramesWritten * bpf); + ma_uint32 framesToCopy = ma_min(pDevice->pulse.mappedBufferFramesRemainingPlayback, (frameCount - totalFramesWritten)); + MA_COPY_MEMORY(pDst, pSrc, framesToCopy * bpf); + + pDevice->pulse.mappedBufferFramesRemainingPlayback -= framesToCopy; + totalFramesWritten += framesToCopy; + } + + /* + Getting here means we've run out of data in the currently mapped chunk. We need to write this to the device and then try + mapping another chunk. If this fails we need to wait for space to become available. + */ + if (pDevice->pulse.mappedBufferFramesCapacityPlayback > 0 && pDevice->pulse.mappedBufferFramesRemainingPlayback == 0) { + size_t nbytes = pDevice->pulse.mappedBufferFramesCapacityPlayback * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + + int error = ((ma_pa_stream_write_proc)pDevice->pContext->pulse.pa_stream_write)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, pDevice->pulse.pMappedBufferPlayback, nbytes, NULL, 0, MA_PA_SEEK_RELATIVE); + if (error < 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to write data to the PulseAudio stream.", ma_result_from_pulse(error)); } + + pDevice->pulse.pMappedBufferPlayback = NULL; + pDevice->pulse.mappedBufferFramesRemainingPlayback = 0; + pDevice->pulse.mappedBufferFramesCapacityPlayback = 0; } - ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts); - } - - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - const char** ppServerPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, NULL, ma_JackPortIsPhysical | ma_JackPortIsInput); - if (ppServerPorts == NULL) { - ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to retrieve physical ports.", MA_ERROR); + MA_ASSERT(totalFramesWritten <= frameCount); + if (totalFramesWritten == frameCount) { + break; } - for (i = 0; ppServerPorts[i] != NULL; ++i) { - const char* pServerPort = ppServerPorts[i]; - const char* pClientPort = ((ma_jack_port_name_proc)pContext->jack.jack_port_name)((ma_jack_port_t*)pDevice->jack.pPortsPlayback[i]); + /* Getting here means we need to map a new buffer. If we don't have enough space we need to wait for more. */ + for (;;) { + size_t writableSizeInBytes; - resultJACK = ((ma_jack_connect_proc)pContext->jack.jack_connect)((ma_jack_client_t*)pDevice->jack.pClient, pClientPort, pServerPort); - if (resultJACK != 0) { - ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts); - ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to connect ports.", MA_ERROR); + /* If the device has been corked, don't try to continue. */ + if (((ma_pa_stream_is_corked_proc)pDevice->pContext->pulse.pa_stream_is_corked)((ma_pa_stream*)pDevice->pulse.pStreamPlayback)) { + break; + } + + writableSizeInBytes = ((ma_pa_stream_writable_size_proc)pDevice->pContext->pulse.pa_stream_writable_size)((ma_pa_stream*)pDevice->pulse.pStreamPlayback); + if (writableSizeInBytes != (size_t)-1) { + if (writableSizeInBytes > 0) { + /* Data is avaialable. */ + size_t bytesToMap = writableSizeInBytes; + int error = ((ma_pa_stream_begin_write_proc)pDevice->pContext->pulse.pa_stream_begin_write)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, &pDevice->pulse.pMappedBufferPlayback, &bytesToMap); + if (error < 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to map write buffer.", ma_result_from_pulse(error)); + } + + pDevice->pulse.mappedBufferFramesCapacityPlayback = bytesToMap / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + pDevice->pulse.mappedBufferFramesRemainingPlayback = pDevice->pulse.mappedBufferFramesCapacityPlayback; + + break; + } else { + /* No data available. Need to wait for more. */ + int error = ((ma_pa_mainloop_iterate_proc)pDevice->pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pDevice->pulse.pMainLoop, 1, NULL); + if (error < 0) { + return ma_result_from_pulse(error); + } + + continue; + } + } else { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to query the stream's writable size.", MA_ERROR); } } + } - ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts); + if (pFramesWritten != NULL) { + *pFramesWritten = totalFramesWritten; } return MA_SUCCESS; } -ma_result ma_device_stop__jack(ma_device* pDevice) +static ma_result ma_device_read__pulse(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead) { - ma_context* pContext = pDevice->pContext; - ma_stop_proc onStop; + ma_uint32 totalFramesRead; - if (((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient) != 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] An error occurred when deactivating the JACK client.", MA_ERROR); - } - - onStop = pDevice->onStop; - if (onStop) { - onStop(pDevice); + MA_ASSERT(pDevice != NULL); + MA_ASSERT(pPCMFrames != NULL); + MA_ASSERT(frameCount > 0); + + if (pFramesRead != NULL) { + *pFramesRead = 0; } - return MA_SUCCESS; -} + totalFramesRead = 0; + while (totalFramesRead < frameCount) { + if (ma_device__get_state(pDevice) != MA_STATE_STARTED) { + return MA_DEVICE_NOT_STARTED; + } + + /* + If a buffer is mapped we need to read from that first. Once it's consumed we need to drop it. Note that pDevice->pulse.pMappedBufferCapture can be null in which + case it could be a hole. In this case we just write zeros into the output buffer. + */ + if (pDevice->pulse.mappedBufferFramesRemainingCapture > 0) { + ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + ma_uint32 mappedBufferFramesConsumed = pDevice->pulse.mappedBufferFramesCapacityCapture - pDevice->pulse.mappedBufferFramesRemainingCapture; + ma_uint32 framesToCopy = ma_min(pDevice->pulse.mappedBufferFramesRemainingCapture, (frameCount - totalFramesRead)); + void* pDst = (ma_uint8*)pPCMFrames + (totalFramesRead * bpf); -ma_result ma_context_uninit__jack(ma_context* pContext) -{ - ma_assert(pContext != NULL); - ma_assert(pContext->backend == ma_backend_jack); + /* + This little bit of logic here is specifically for PulseAudio and it's hole management. The buffer pointer will be set to NULL + when the current fragment is a hole. For a hole we just output silence. + */ + if (pDevice->pulse.pMappedBufferCapture != NULL) { + const void* pSrc = (const ma_uint8*)pDevice->pulse.pMappedBufferCapture + (mappedBufferFramesConsumed * bpf); + MA_COPY_MEMORY(pDst, pSrc, framesToCopy * bpf); + } else { + MA_ZERO_MEMORY(pDst, framesToCopy * bpf); + #if defined(MA_DEBUG_OUTPUT) + printf("[PulseAudio] ma_device_read__pulse: Filling hole with silence.\n"); + #endif + } - ma_free(pContext->jack.pClientName); - pContext->jack.pClientName = NULL; + pDevice->pulse.mappedBufferFramesRemainingCapture -= framesToCopy; + totalFramesRead += framesToCopy; + } -#ifndef MA_NO_RUNTIME_LINKING - ma_dlclose(pContext, pContext->jack.jackSO); -#endif + /* + Getting here means we've run out of data in the currently mapped chunk. We need to drop this from the device and then try + mapping another chunk. If this fails we need to wait for data to become available. + */ + if (pDevice->pulse.mappedBufferFramesCapacityCapture > 0 && pDevice->pulse.mappedBufferFramesRemainingCapture == 0) { + int error; - return MA_SUCCESS; -} + #if defined(MA_DEBUG_OUTPUT) + printf("[PulseAudio] ma_device_read__pulse: Call pa_stream_drop()\n"); + #endif -ma_result ma_context_init__jack(const ma_context_config* pConfig, ma_context* pContext) -{ -#ifndef MA_NO_RUNTIME_LINKING - const char* libjackNames[] = { -#ifdef MA_WIN32 - "libjack.dll" -#else - "libjack.so", - "libjack.so.0" -#endif - }; - size_t i; + error = ((ma_pa_stream_drop_proc)pDevice->pContext->pulse.pa_stream_drop)((ma_pa_stream*)pDevice->pulse.pStreamCapture); + if (error != 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to drop fragment.", ma_result_from_pulse(error)); + } - for (i = 0; i < ma_countof(libjackNames); ++i) { - pContext->jack.jackSO = ma_dlopen(pContext, libjackNames[i]); - if (pContext->jack.jackSO != NULL) { + pDevice->pulse.pMappedBufferCapture = NULL; + pDevice->pulse.mappedBufferFramesRemainingCapture = 0; + pDevice->pulse.mappedBufferFramesCapacityCapture = 0; + } + + MA_ASSERT(totalFramesRead <= frameCount); + if (totalFramesRead == frameCount) { break; } - } - if (pContext->jack.jackSO == NULL) { - return MA_NO_BACKEND; - } + /* Getting here means we need to map a new buffer. If we don't have enough data we wait for more. */ + for (;;) { + int error; + size_t bytesMapped; - pContext->jack.jack_client_open = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_client_open"); - pContext->jack.jack_client_close = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_client_close"); - pContext->jack.jack_client_name_size = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_client_name_size"); - pContext->jack.jack_set_process_callback = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_set_process_callback"); - pContext->jack.jack_set_buffer_size_callback = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_set_buffer_size_callback"); - pContext->jack.jack_on_shutdown = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_on_shutdown"); - pContext->jack.jack_get_sample_rate = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_get_sample_rate"); - pContext->jack.jack_get_buffer_size = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_get_buffer_size"); - pContext->jack.jack_get_ports = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_get_ports"); - pContext->jack.jack_activate = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_activate"); - pContext->jack.jack_deactivate = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_deactivate"); - pContext->jack.jack_connect = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_connect"); - pContext->jack.jack_port_register = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_port_register"); - pContext->jack.jack_port_name = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_port_name"); - pContext->jack.jack_port_get_buffer = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_port_get_buffer"); - pContext->jack.jack_free = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_free"); -#else - /* - This strange assignment system is here just to ensure type safety of miniaudio's function pointer - types. If anything differs slightly the compiler should throw a warning. - */ - ma_jack_client_open_proc _jack_client_open = jack_client_open; - ma_jack_client_close_proc _jack_client_close = jack_client_close; - ma_jack_client_name_size_proc _jack_client_name_size = jack_client_name_size; - ma_jack_set_process_callback_proc _jack_set_process_callback = jack_set_process_callback; - ma_jack_set_buffer_size_callback_proc _jack_set_buffer_size_callback = jack_set_buffer_size_callback; - ma_jack_on_shutdown_proc _jack_on_shutdown = jack_on_shutdown; - ma_jack_get_sample_rate_proc _jack_get_sample_rate = jack_get_sample_rate; - ma_jack_get_buffer_size_proc _jack_get_buffer_size = jack_get_buffer_size; - ma_jack_get_ports_proc _jack_get_ports = jack_get_ports; - ma_jack_activate_proc _jack_activate = jack_activate; - ma_jack_deactivate_proc _jack_deactivate = jack_deactivate; - ma_jack_connect_proc _jack_connect = jack_connect; - ma_jack_port_register_proc _jack_port_register = jack_port_register; - ma_jack_port_name_proc _jack_port_name = jack_port_name; - ma_jack_port_get_buffer_proc _jack_port_get_buffer = jack_port_get_buffer; - ma_jack_free_proc _jack_free = jack_free; + if (ma_device__get_state(pDevice) != MA_STATE_STARTED) { + break; + } - pContext->jack.jack_client_open = (ma_proc)_jack_client_open; - pContext->jack.jack_client_close = (ma_proc)_jack_client_close; - pContext->jack.jack_client_name_size = (ma_proc)_jack_client_name_size; - pContext->jack.jack_set_process_callback = (ma_proc)_jack_set_process_callback; - pContext->jack.jack_set_buffer_size_callback = (ma_proc)_jack_set_buffer_size_callback; - pContext->jack.jack_on_shutdown = (ma_proc)_jack_on_shutdown; - pContext->jack.jack_get_sample_rate = (ma_proc)_jack_get_sample_rate; - pContext->jack.jack_get_buffer_size = (ma_proc)_jack_get_buffer_size; - pContext->jack.jack_get_ports = (ma_proc)_jack_get_ports; - pContext->jack.jack_activate = (ma_proc)_jack_activate; - pContext->jack.jack_deactivate = (ma_proc)_jack_deactivate; - pContext->jack.jack_connect = (ma_proc)_jack_connect; - pContext->jack.jack_port_register = (ma_proc)_jack_port_register; - pContext->jack.jack_port_name = (ma_proc)_jack_port_name; - pContext->jack.jack_port_get_buffer = (ma_proc)_jack_port_get_buffer; - pContext->jack.jack_free = (ma_proc)_jack_free; -#endif + /* If the device has been corked, don't try to continue. */ + if (((ma_pa_stream_is_corked_proc)pDevice->pContext->pulse.pa_stream_is_corked)((ma_pa_stream*)pDevice->pulse.pStreamCapture)) { + #if defined(MA_DEBUG_OUTPUT) + printf("[PulseAudio] ma_device_read__pulse: Corked.\n"); + #endif + break; + } + + MA_ASSERT(pDevice->pulse.pMappedBufferCapture == NULL); /* <-- We're about to map a buffer which means we shouldn't have an existing mapping. */ + + error = ((ma_pa_stream_peek_proc)pDevice->pContext->pulse.pa_stream_peek)((ma_pa_stream*)pDevice->pulse.pStreamCapture, &pDevice->pulse.pMappedBufferCapture, &bytesMapped); + if (error < 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to peek capture buffer.", ma_result_from_pulse(error)); + } + + if (bytesMapped > 0) { + pDevice->pulse.mappedBufferFramesCapacityCapture = bytesMapped / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + pDevice->pulse.mappedBufferFramesRemainingCapture = pDevice->pulse.mappedBufferFramesCapacityCapture; - pContext->isBackendAsynchronous = MA_TRUE; + #if defined(MA_DEBUG_OUTPUT) + printf("[PulseAudio] ma_device_read__pulse: Mapped. mappedBufferFramesCapacityCapture=%d, mappedBufferFramesRemainingCapture=%d\n", pDevice->pulse.mappedBufferFramesCapacityCapture, pDevice->pulse.mappedBufferFramesRemainingCapture); + #endif - pContext->onUninit = ma_context_uninit__jack; - pContext->onDeviceIDEqual = ma_context_is_device_id_equal__jack; - pContext->onEnumDevices = ma_context_enumerate_devices__jack; - pContext->onGetDeviceInfo = ma_context_get_device_info__jack; - pContext->onDeviceInit = ma_device_init__jack; - pContext->onDeviceUninit = ma_device_uninit__jack; - pContext->onDeviceStart = ma_device_start__jack; - pContext->onDeviceStop = ma_device_stop__jack; + if (pDevice->pulse.pMappedBufferCapture == NULL) { + /* It's a hole. */ + #if defined(MA_DEBUG_OUTPUT) + printf("[PulseAudio] ma_device_read__pulse: Call pa_stream_peek(). Hole.\n"); + #endif + } - if (pConfig->jack.pClientName != NULL) { - pContext->jack.pClientName = ma_copy_string(pConfig->jack.pClientName); - } - pContext->jack.tryStartServer = pConfig->jack.tryStartServer; + break; + } else { + if (pDevice->pulse.pMappedBufferCapture == NULL) { + /* Nothing available yet. Need to wait for more. */ - /* - Getting here means the JACK library is installed, but it doesn't necessarily mean it's usable. We need to quickly test this by connecting - a temporary client. - */ - { - ma_jack_client_t* pDummyClient; - ma_result result = ma_context_open_client__jack(pContext, &pDummyClient); - if (result != MA_SUCCESS) { - ma_free(pContext->jack.pClientName); - #ifndef MA_NO_RUNTIME_LINKING - ma_dlclose(pContext, pContext->jack.jackSO); - #endif - return MA_NO_BACKEND; + /* + I have had reports of a deadlock in this part of the code. I have reproduced this when using the "Built-in Audio Analogue Stereo" device without + an actual microphone connected. I'm experimenting here by not blocking in pa_mainloop_iterate() and instead sleep for a bit when there are no + dispatches. + */ + error = ((ma_pa_mainloop_iterate_proc)pDevice->pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pDevice->pulse.pMainLoop, 0, NULL); + if (error < 0) { + return ma_result_from_pulse(error); + } + + /* Sleep for a bit if nothing was dispatched. */ + if (error == 0) { + ma_sleep(1); + } + + #if defined(MA_DEBUG_OUTPUT) + printf("[PulseAudio] ma_device_read__pulse: No data available. Waiting. mappedBufferFramesCapacityCapture=%d, mappedBufferFramesRemainingCapture=%d\n", pDevice->pulse.mappedBufferFramesCapacityCapture, pDevice->pulse.mappedBufferFramesRemainingCapture); + #endif + } else { + /* Getting here means we mapped 0 bytes, but have a non-NULL buffer. I don't think this should ever happen. */ + MA_ASSERT(MA_FALSE); + } + } } + } - ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pDummyClient); + if (pFramesRead != NULL) { + *pFramesRead = totalFramesRead; } return MA_SUCCESS; } -#endif /* JACK */ +static ma_result ma_device_main_loop__pulse(ma_device* pDevice) +{ + ma_result result = MA_SUCCESS; + ma_bool32 exitLoop = MA_FALSE; + MA_ASSERT(pDevice != NULL); -/****************************************************************************** + /* The stream needs to be uncorked first. We do this at the top for both capture and playback for PulseAudio. */ + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + result = ma_device__cork_stream__pulse(pDevice, ma_device_type_capture, 0); + if (result != MA_SUCCESS) { + return result; + } + } + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + result = ma_device__cork_stream__pulse(pDevice, ma_device_type_playback, 0); + if (result != MA_SUCCESS) { + return result; + } + } -Core Audio Backend -******************************************************************************/ -#ifdef MA_HAS_COREAUDIO -#include + while (ma_device__get_state(pDevice) == MA_STATE_STARTED && !exitLoop) { + switch (pDevice->type) + { + case ma_device_type_duplex: + { + /* The process is: device_read -> convert -> callback -> convert -> device_write */ + ma_uint32 totalCapturedDeviceFramesProcessed = 0; + ma_uint32 capturedDevicePeriodSizeInFrames = ma_min(pDevice->capture.internalPeriodSizeInFrames, pDevice->playback.internalPeriodSizeInFrames); + + while (totalCapturedDeviceFramesProcessed < capturedDevicePeriodSizeInFrames) { + ma_uint8 capturedDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint8 playbackDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 capturedDeviceDataCapInFrames = sizeof(capturedDeviceData) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + ma_uint32 playbackDeviceDataCapInFrames = sizeof(playbackDeviceData) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + ma_uint32 capturedDeviceFramesRemaining; + ma_uint32 capturedDeviceFramesProcessed; + ma_uint32 capturedDeviceFramesToProcess; + ma_uint32 capturedDeviceFramesToTryProcessing = capturedDevicePeriodSizeInFrames - totalCapturedDeviceFramesProcessed; + if (capturedDeviceFramesToTryProcessing > capturedDeviceDataCapInFrames) { + capturedDeviceFramesToTryProcessing = capturedDeviceDataCapInFrames; + } -#if defined(TARGET_OS_IPHONE) && TARGET_OS_IPHONE == 1 - #define MA_APPLE_MOBILE -#else - #define MA_APPLE_DESKTOP -#endif + result = ma_device_read__pulse(pDevice, capturedDeviceData, capturedDeviceFramesToTryProcessing, &capturedDeviceFramesToProcess); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } -#if defined(MA_APPLE_DESKTOP) -#include -#else -#include -#endif + capturedDeviceFramesRemaining = capturedDeviceFramesToProcess; + capturedDeviceFramesProcessed = 0; -#include + for (;;) { + ma_uint8 capturedClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint8 playbackClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 capturedClientDataCapInFrames = sizeof(capturedClientData) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); + ma_uint32 playbackClientDataCapInFrames = sizeof(playbackClientData) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); + ma_uint64 capturedClientFramesToProcessThisIteration = ma_min(capturedClientDataCapInFrames, playbackClientDataCapInFrames); + ma_uint64 capturedDeviceFramesToProcessThisIteration = capturedDeviceFramesRemaining; + ma_uint8* pRunningCapturedDeviceFrames = ma_offset_ptr(capturedDeviceData, capturedDeviceFramesProcessed * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); + + /* Convert capture data from device format to client format. */ + result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningCapturedDeviceFrames, &capturedDeviceFramesToProcessThisIteration, capturedClientData, &capturedClientFramesToProcessThisIteration); + if (result != MA_SUCCESS) { + break; + } -/* CoreFoundation */ -typedef Boolean (* ma_CFStringGetCString_proc)(CFStringRef theString, char* buffer, CFIndex bufferSize, CFStringEncoding encoding); -typedef void (* ma_CFRelease_proc)(CFTypeRef cf); + /* + If we weren't able to generate any output frames it must mean we've exhaused all of our input. The only time this would not be the case is if capturedClientData was too small + which should never be the case when it's of the size MA_DATA_CONVERTER_STACK_BUFFER_SIZE. + */ + if (capturedClientFramesToProcessThisIteration == 0) { + break; + } -/* CoreAudio */ -#if defined(MA_APPLE_DESKTOP) -typedef OSStatus (* ma_AudioObjectGetPropertyData_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, UInt32 inQualifierDataSize, const void* inQualifierData, UInt32* ioDataSize, void* outData); -typedef OSStatus (* ma_AudioObjectGetPropertyDataSize_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, UInt32 inQualifierDataSize, const void* inQualifierData, UInt32* outDataSize); -typedef OSStatus (* ma_AudioObjectSetPropertyData_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, UInt32 inQualifierDataSize, const void* inQualifierData, UInt32 inDataSize, const void* inData); -typedef OSStatus (* ma_AudioObjectAddPropertyListener_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, AudioObjectPropertyListenerProc inListener, void* inClientData); -typedef OSStatus (* ma_AudioObjectRemovePropertyListener_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, AudioObjectPropertyListenerProc inListener, void* inClientData); -#endif + ma_device__on_data(pDevice, playbackClientData, capturedClientData, (ma_uint32)capturedClientFramesToProcessThisIteration); /* Safe cast .*/ -/* AudioToolbox */ -typedef AudioComponent (* ma_AudioComponentFindNext_proc)(AudioComponent inComponent, const AudioComponentDescription* inDesc); -typedef OSStatus (* ma_AudioComponentInstanceDispose_proc)(AudioComponentInstance inInstance); -typedef OSStatus (* ma_AudioComponentInstanceNew_proc)(AudioComponent inComponent, AudioComponentInstance* outInstance); -typedef OSStatus (* ma_AudioOutputUnitStart_proc)(AudioUnit inUnit); -typedef OSStatus (* ma_AudioOutputUnitStop_proc)(AudioUnit inUnit); -typedef OSStatus (* ma_AudioUnitAddPropertyListener_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitPropertyListenerProc inProc, void* inProcUserData); -typedef OSStatus (* ma_AudioUnitGetPropertyInfo_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, UInt32* outDataSize, Boolean* outWriteable); -typedef OSStatus (* ma_AudioUnitGetProperty_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, void* outData, UInt32* ioDataSize); -typedef OSStatus (* ma_AudioUnitSetProperty_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, const void* inData, UInt32 inDataSize); -typedef OSStatus (* ma_AudioUnitInitialize_proc)(AudioUnit inUnit); -typedef OSStatus (* ma_AudioUnitRender_proc)(AudioUnit inUnit, AudioUnitRenderActionFlags* ioActionFlags, const AudioTimeStamp* inTimeStamp, UInt32 inOutputBusNumber, UInt32 inNumberFrames, AudioBufferList* ioData); + capturedDeviceFramesProcessed += (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */ + capturedDeviceFramesRemaining -= (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */ + /* At this point the playbackClientData buffer should be holding data that needs to be written to the device. */ + for (;;) { + ma_uint64 convertedClientFrameCount = capturedClientFramesToProcessThisIteration; + ma_uint64 convertedDeviceFrameCount = playbackDeviceDataCapInFrames; + result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, playbackClientData, &convertedClientFrameCount, playbackDeviceData, &convertedDeviceFrameCount); + if (result != MA_SUCCESS) { + break; + } -#define MA_COREAUDIO_OUTPUT_BUS 0 -#define MA_COREAUDIO_INPUT_BUS 1 + result = ma_device_write__pulse(pDevice, playbackDeviceData, (ma_uint32)convertedDeviceFrameCount, NULL); /* Safe cast. */ + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } -ma_result ma_device_reinit_internal__coreaudio(ma_device* pDevice, ma_device_type deviceType, ma_bool32 disposePreviousAudioUnit); + capturedClientFramesToProcessThisIteration -= (ma_uint32)convertedClientFrameCount; /* Safe cast. */ + if (capturedClientFramesToProcessThisIteration == 0) { + break; + } + } -/* -Core Audio + /* In case an error happened from ma_device_write__pulse()... */ + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } + } -So far, Core Audio has been the worst backend to work with due to being both unintuitive and having almost no documentation -apart from comments in the headers (which admittedly are quite good). For my own purposes, and for anybody out there whose -needing to figure out how this darn thing works, I'm going to outline a few things here. + totalCapturedDeviceFramesProcessed += capturedDeviceFramesProcessed; + } + } break; -Since miniaudio is a fairly low-level API, one of the things it needs is control over specific devices, and it needs to be -able to identify whether or not it can be used as playback and/or capture. The AudioObject API is the only one I've seen -that supports this level of detail. There was some public domain sample code I stumbled across that used the AudioComponent -and AudioUnit APIs, but I couldn't see anything that gave low-level control over device selection and capabilities (the -distinction between playback and capture in particular). Therefore, miniaudio is using the AudioObject API. + case ma_device_type_capture: + { + ma_uint8 intermediaryBuffer[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + ma_uint32 periodSizeInFrames = pDevice->capture.internalPeriodSizeInFrames; + ma_uint32 framesReadThisPeriod = 0; + while (framesReadThisPeriod < periodSizeInFrames) { + ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesReadThisPeriod; + ma_uint32 framesProcessed; + ma_uint32 framesToReadThisIteration = framesRemainingInPeriod; + if (framesToReadThisIteration > intermediaryBufferSizeInFrames) { + framesToReadThisIteration = intermediaryBufferSizeInFrames; + } -Most (all?) functions in the AudioObject API take a AudioObjectID as it's input. This is the device identifier. When -retrieving global information, such as the device list, you use kAudioObjectSystemObject. When retrieving device-specific -data, you pass in the ID for that device. In order to retrieve device-specific IDs you need to enumerate over each of the -devices. This is done using the AudioObjectGetPropertyDataSize() and AudioObjectGetPropertyData() APIs which seem to be -the central APIs for retrieving information about the system and specific devices. + result = ma_device_read__pulse(pDevice, intermediaryBuffer, framesToReadThisIteration, &framesProcessed); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } -To use the AudioObjectGetPropertyData() API you need to use the notion of a property address. A property address is a -structure with three variables and is used to identify which property you are getting or setting. The first is the "selector" -which is basically the specific property that you're wanting to retrieve or set. The second is the "scope", which is -typically set to kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyScopeInput for input-specific properties and -kAudioObjectPropertyScopeOutput for output-specific properties. The last is the "element" which is always set to -kAudioObjectPropertyElementMaster in miniaudio's case. I don't know of any cases where this would be set to anything different. + ma_device__send_frames_to_client(pDevice, framesProcessed, intermediaryBuffer); -Back to the earlier issue of device retrieval, you first use the AudioObjectGetPropertyDataSize() API to retrieve the size -of the raw data which is just a list of AudioDeviceID's. You use the kAudioObjectSystemObject AudioObjectID, and a property -address with the kAudioHardwarePropertyDevices selector and the kAudioObjectPropertyScopeGlobal scope. Once you have the -size, allocate a block of memory of that size and then call AudioObjectGetPropertyData(). The data is just a list of -AudioDeviceID's so just do "dataSize/sizeof(AudioDeviceID)" to know the device count. -*/ + framesReadThisPeriod += framesProcessed; + } + } break; -ma_result ma_result_from_OSStatus(OSStatus status) -{ - switch (status) - { - case noErr: return MA_SUCCESS; - #if defined(MA_APPLE_DESKTOP) - case kAudioHardwareNotRunningError: return MA_DEVICE_NOT_STARTED; - case kAudioHardwareUnspecifiedError: return MA_ERROR; - case kAudioHardwareUnknownPropertyError: return MA_INVALID_ARGS; - case kAudioHardwareBadPropertySizeError: return MA_INVALID_OPERATION; - case kAudioHardwareIllegalOperationError: return MA_INVALID_OPERATION; - case kAudioHardwareBadObjectError: return MA_INVALID_ARGS; - case kAudioHardwareBadDeviceError: return MA_INVALID_ARGS; - case kAudioHardwareBadStreamError: return MA_INVALID_ARGS; - case kAudioHardwareUnsupportedOperationError: return MA_INVALID_OPERATION; - case kAudioDeviceUnsupportedFormatError: return MA_FORMAT_NOT_SUPPORTED; - case kAudioDevicePermissionsError: return MA_ACCESS_DENIED; - #endif - default: return MA_ERROR; - } -} + case ma_device_type_playback: + { + ma_uint8 intermediaryBuffer[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + ma_uint32 periodSizeInFrames = pDevice->playback.internalPeriodSizeInFrames; + ma_uint32 framesWrittenThisPeriod = 0; + while (framesWrittenThisPeriod < periodSizeInFrames) { + ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesWrittenThisPeriod; + ma_uint32 framesProcessed; + ma_uint32 framesToWriteThisIteration = framesRemainingInPeriod; + if (framesToWriteThisIteration > intermediaryBufferSizeInFrames) { + framesToWriteThisIteration = intermediaryBufferSizeInFrames; + } -#if 0 -ma_channel ma_channel_from_AudioChannelBitmap(AudioChannelBitmap bit) -{ - switch (bit) - { - case kAudioChannelBit_Left: return MA_CHANNEL_LEFT; - case kAudioChannelBit_Right: return MA_CHANNEL_RIGHT; - case kAudioChannelBit_Center: return MA_CHANNEL_FRONT_CENTER; - case kAudioChannelBit_LFEScreen: return MA_CHANNEL_LFE; - case kAudioChannelBit_LeftSurround: return MA_CHANNEL_BACK_LEFT; - case kAudioChannelBit_RightSurround: return MA_CHANNEL_BACK_RIGHT; - case kAudioChannelBit_LeftCenter: return MA_CHANNEL_FRONT_LEFT_CENTER; - case kAudioChannelBit_RightCenter: return MA_CHANNEL_FRONT_RIGHT_CENTER; - case kAudioChannelBit_CenterSurround: return MA_CHANNEL_BACK_CENTER; - case kAudioChannelBit_LeftSurroundDirect: return MA_CHANNEL_SIDE_LEFT; - case kAudioChannelBit_RightSurroundDirect: return MA_CHANNEL_SIDE_RIGHT; - case kAudioChannelBit_TopCenterSurround: return MA_CHANNEL_TOP_CENTER; - case kAudioChannelBit_VerticalHeightLeft: return MA_CHANNEL_TOP_FRONT_LEFT; - case kAudioChannelBit_VerticalHeightCenter: return MA_CHANNEL_TOP_FRONT_CENTER; - case kAudioChannelBit_VerticalHeightRight: return MA_CHANNEL_TOP_FRONT_RIGHT; - case kAudioChannelBit_TopBackLeft: return MA_CHANNEL_TOP_BACK_LEFT; - case kAudioChannelBit_TopBackCenter: return MA_CHANNEL_TOP_BACK_CENTER; - case kAudioChannelBit_TopBackRight: return MA_CHANNEL_TOP_BACK_RIGHT; - default: return MA_CHANNEL_NONE; + ma_device__read_frames_from_client(pDevice, framesToWriteThisIteration, intermediaryBuffer); + + result = ma_device_write__pulse(pDevice, intermediaryBuffer, framesToWriteThisIteration, &framesProcessed); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } + + framesWrittenThisPeriod += framesProcessed; + } + } break; + + /* To silence a warning. Will never hit this. */ + case ma_device_type_loopback: + default: break; + } } + + /* Here is where the device needs to be stopped. */ + ma_device_stop__pulse(pDevice); + + return result; } + + +static ma_result ma_context_uninit__pulse(ma_context* pContext) +{ + MA_ASSERT(pContext != NULL); + MA_ASSERT(pContext->backend == ma_backend_pulseaudio); + + ma_free(pContext->pulse.pServerName, &pContext->allocationCallbacks); + pContext->pulse.pServerName = NULL; + + ma_free(pContext->pulse.pApplicationName, &pContext->allocationCallbacks); + pContext->pulse.pApplicationName = NULL; + +#ifndef MA_NO_RUNTIME_LINKING + ma_dlclose(pContext, pContext->pulse.pulseSO); #endif -ma_channel ma_channel_from_AudioChannelLabel(AudioChannelLabel label) + return MA_SUCCESS; +} + +static ma_result ma_context_init__pulse(const ma_context_config* pConfig, ma_context* pContext) { - switch (label) - { - case kAudioChannelLabel_Unknown: return MA_CHANNEL_NONE; - case kAudioChannelLabel_Unused: return MA_CHANNEL_NONE; - case kAudioChannelLabel_UseCoordinates: return MA_CHANNEL_NONE; - case kAudioChannelLabel_Left: return MA_CHANNEL_LEFT; - case kAudioChannelLabel_Right: return MA_CHANNEL_RIGHT; - case kAudioChannelLabel_Center: return MA_CHANNEL_FRONT_CENTER; - case kAudioChannelLabel_LFEScreen: return MA_CHANNEL_LFE; - case kAudioChannelLabel_LeftSurround: return MA_CHANNEL_BACK_LEFT; - case kAudioChannelLabel_RightSurround: return MA_CHANNEL_BACK_RIGHT; - case kAudioChannelLabel_LeftCenter: return MA_CHANNEL_FRONT_LEFT_CENTER; - case kAudioChannelLabel_RightCenter: return MA_CHANNEL_FRONT_RIGHT_CENTER; - case kAudioChannelLabel_CenterSurround: return MA_CHANNEL_BACK_CENTER; - case kAudioChannelLabel_LeftSurroundDirect: return MA_CHANNEL_SIDE_LEFT; - case kAudioChannelLabel_RightSurroundDirect: return MA_CHANNEL_SIDE_RIGHT; - case kAudioChannelLabel_TopCenterSurround: return MA_CHANNEL_TOP_CENTER; - case kAudioChannelLabel_VerticalHeightLeft: return MA_CHANNEL_TOP_FRONT_LEFT; - case kAudioChannelLabel_VerticalHeightCenter: return MA_CHANNEL_TOP_FRONT_CENTER; - case kAudioChannelLabel_VerticalHeightRight: return MA_CHANNEL_TOP_FRONT_RIGHT; - case kAudioChannelLabel_TopBackLeft: return MA_CHANNEL_TOP_BACK_LEFT; - case kAudioChannelLabel_TopBackCenter: return MA_CHANNEL_TOP_BACK_CENTER; - case kAudioChannelLabel_TopBackRight: return MA_CHANNEL_TOP_BACK_RIGHT; - case kAudioChannelLabel_RearSurroundLeft: return MA_CHANNEL_BACK_LEFT; - case kAudioChannelLabel_RearSurroundRight: return MA_CHANNEL_BACK_RIGHT; - case kAudioChannelLabel_LeftWide: return MA_CHANNEL_SIDE_LEFT; - case kAudioChannelLabel_RightWide: return MA_CHANNEL_SIDE_RIGHT; - case kAudioChannelLabel_LFE2: return MA_CHANNEL_LFE; - case kAudioChannelLabel_LeftTotal: return MA_CHANNEL_LEFT; - case kAudioChannelLabel_RightTotal: return MA_CHANNEL_RIGHT; - case kAudioChannelLabel_HearingImpaired: return MA_CHANNEL_NONE; - case kAudioChannelLabel_Narration: return MA_CHANNEL_MONO; - case kAudioChannelLabel_Mono: return MA_CHANNEL_MONO; - case kAudioChannelLabel_DialogCentricMix: return MA_CHANNEL_MONO; - case kAudioChannelLabel_CenterSurroundDirect: return MA_CHANNEL_BACK_CENTER; - case kAudioChannelLabel_Haptic: return MA_CHANNEL_NONE; - case kAudioChannelLabel_Ambisonic_W: return MA_CHANNEL_NONE; - case kAudioChannelLabel_Ambisonic_X: return MA_CHANNEL_NONE; - case kAudioChannelLabel_Ambisonic_Y: return MA_CHANNEL_NONE; - case kAudioChannelLabel_Ambisonic_Z: return MA_CHANNEL_NONE; - case kAudioChannelLabel_MS_Mid: return MA_CHANNEL_LEFT; - case kAudioChannelLabel_MS_Side: return MA_CHANNEL_RIGHT; - case kAudioChannelLabel_XY_X: return MA_CHANNEL_LEFT; - case kAudioChannelLabel_XY_Y: return MA_CHANNEL_RIGHT; - case kAudioChannelLabel_HeadphonesLeft: return MA_CHANNEL_LEFT; - case kAudioChannelLabel_HeadphonesRight: return MA_CHANNEL_RIGHT; - case kAudioChannelLabel_ClickTrack: return MA_CHANNEL_NONE; - case kAudioChannelLabel_ForeignLanguage: return MA_CHANNEL_NONE; - case kAudioChannelLabel_Discrete: return MA_CHANNEL_NONE; - case kAudioChannelLabel_Discrete_0: return MA_CHANNEL_AUX_0; - case kAudioChannelLabel_Discrete_1: return MA_CHANNEL_AUX_1; - case kAudioChannelLabel_Discrete_2: return MA_CHANNEL_AUX_2; - case kAudioChannelLabel_Discrete_3: return MA_CHANNEL_AUX_3; - case kAudioChannelLabel_Discrete_4: return MA_CHANNEL_AUX_4; - case kAudioChannelLabel_Discrete_5: return MA_CHANNEL_AUX_5; - case kAudioChannelLabel_Discrete_6: return MA_CHANNEL_AUX_6; - case kAudioChannelLabel_Discrete_7: return MA_CHANNEL_AUX_7; - case kAudioChannelLabel_Discrete_8: return MA_CHANNEL_AUX_8; - case kAudioChannelLabel_Discrete_9: return MA_CHANNEL_AUX_9; - case kAudioChannelLabel_Discrete_10: return MA_CHANNEL_AUX_10; - case kAudioChannelLabel_Discrete_11: return MA_CHANNEL_AUX_11; - case kAudioChannelLabel_Discrete_12: return MA_CHANNEL_AUX_12; - case kAudioChannelLabel_Discrete_13: return MA_CHANNEL_AUX_13; - case kAudioChannelLabel_Discrete_14: return MA_CHANNEL_AUX_14; - case kAudioChannelLabel_Discrete_15: return MA_CHANNEL_AUX_15; - case kAudioChannelLabel_Discrete_65535: return MA_CHANNEL_NONE; - - #if 0 /* Introduced in a later version of macOS. */ - case kAudioChannelLabel_HOA_ACN: return MA_CHANNEL_NONE; - case kAudioChannelLabel_HOA_ACN_0: return MA_CHANNEL_AUX_0; - case kAudioChannelLabel_HOA_ACN_1: return MA_CHANNEL_AUX_1; - case kAudioChannelLabel_HOA_ACN_2: return MA_CHANNEL_AUX_2; - case kAudioChannelLabel_HOA_ACN_3: return MA_CHANNEL_AUX_3; - case kAudioChannelLabel_HOA_ACN_4: return MA_CHANNEL_AUX_4; - case kAudioChannelLabel_HOA_ACN_5: return MA_CHANNEL_AUX_5; - case kAudioChannelLabel_HOA_ACN_6: return MA_CHANNEL_AUX_6; - case kAudioChannelLabel_HOA_ACN_7: return MA_CHANNEL_AUX_7; - case kAudioChannelLabel_HOA_ACN_8: return MA_CHANNEL_AUX_8; - case kAudioChannelLabel_HOA_ACN_9: return MA_CHANNEL_AUX_9; - case kAudioChannelLabel_HOA_ACN_10: return MA_CHANNEL_AUX_10; - case kAudioChannelLabel_HOA_ACN_11: return MA_CHANNEL_AUX_11; - case kAudioChannelLabel_HOA_ACN_12: return MA_CHANNEL_AUX_12; - case kAudioChannelLabel_HOA_ACN_13: return MA_CHANNEL_AUX_13; - case kAudioChannelLabel_HOA_ACN_14: return MA_CHANNEL_AUX_14; - case kAudioChannelLabel_HOA_ACN_15: return MA_CHANNEL_AUX_15; - case kAudioChannelLabel_HOA_ACN_65024: return MA_CHANNEL_NONE; - #endif - - default: return MA_CHANNEL_NONE; +#ifndef MA_NO_RUNTIME_LINKING + const char* libpulseNames[] = { + "libpulse.so", + "libpulse.so.0" + }; + size_t i; + + for (i = 0; i < ma_countof(libpulseNames); ++i) { + pContext->pulse.pulseSO = ma_dlopen(pContext, libpulseNames[i]); + if (pContext->pulse.pulseSO != NULL) { + break; + } } -} -ma_result ma_format_from_AudioStreamBasicDescription(const AudioStreamBasicDescription* pDescription, ma_format* pFormatOut) -{ - ma_assert(pDescription != NULL); - ma_assert(pFormatOut != NULL); - - *pFormatOut = ma_format_unknown; /* Safety. */ - - /* There's a few things miniaudio doesn't support. */ - if (pDescription->mFormatID != kAudioFormatLinearPCM) { - return MA_FORMAT_NOT_SUPPORTED; - } - - /* We don't support any non-packed formats that are aligned high. */ - if ((pDescription->mFormatFlags & kLinearPCMFormatFlagIsAlignedHigh) != 0) { - return MA_FORMAT_NOT_SUPPORTED; - } + if (pContext->pulse.pulseSO == NULL) { + return MA_NO_BACKEND; + } + + pContext->pulse.pa_mainloop_new = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_new"); + pContext->pulse.pa_mainloop_free = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_free"); + pContext->pulse.pa_mainloop_get_api = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_get_api"); + pContext->pulse.pa_mainloop_iterate = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_iterate"); + pContext->pulse.pa_mainloop_wakeup = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_wakeup"); + pContext->pulse.pa_context_new = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_new"); + pContext->pulse.pa_context_unref = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_unref"); + pContext->pulse.pa_context_connect = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_connect"); + pContext->pulse.pa_context_disconnect = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_disconnect"); + pContext->pulse.pa_context_set_state_callback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_set_state_callback"); + pContext->pulse.pa_context_get_state = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_state"); + pContext->pulse.pa_context_get_sink_info_list = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_sink_info_list"); + pContext->pulse.pa_context_get_source_info_list = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_source_info_list"); + pContext->pulse.pa_context_get_sink_info_by_name = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_sink_info_by_name"); + pContext->pulse.pa_context_get_source_info_by_name = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_source_info_by_name"); + pContext->pulse.pa_operation_unref = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_operation_unref"); + pContext->pulse.pa_operation_get_state = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_operation_get_state"); + pContext->pulse.pa_channel_map_init_extend = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_channel_map_init_extend"); + pContext->pulse.pa_channel_map_valid = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_channel_map_valid"); + pContext->pulse.pa_channel_map_compatible = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_channel_map_compatible"); + pContext->pulse.pa_stream_new = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_new"); + pContext->pulse.pa_stream_unref = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_unref"); + pContext->pulse.pa_stream_connect_playback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_connect_playback"); + pContext->pulse.pa_stream_connect_record = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_connect_record"); + pContext->pulse.pa_stream_disconnect = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_disconnect"); + pContext->pulse.pa_stream_get_state = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_state"); + pContext->pulse.pa_stream_get_sample_spec = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_sample_spec"); + pContext->pulse.pa_stream_get_channel_map = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_channel_map"); + pContext->pulse.pa_stream_get_buffer_attr = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_buffer_attr"); + pContext->pulse.pa_stream_set_buffer_attr = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_set_buffer_attr"); + pContext->pulse.pa_stream_get_device_name = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_device_name"); + pContext->pulse.pa_stream_set_write_callback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_set_write_callback"); + pContext->pulse.pa_stream_set_read_callback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_set_read_callback"); + pContext->pulse.pa_stream_flush = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_flush"); + pContext->pulse.pa_stream_drain = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_drain"); + pContext->pulse.pa_stream_is_corked = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_is_corked"); + pContext->pulse.pa_stream_cork = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_cork"); + pContext->pulse.pa_stream_trigger = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_trigger"); + pContext->pulse.pa_stream_begin_write = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_begin_write"); + pContext->pulse.pa_stream_write = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_write"); + pContext->pulse.pa_stream_peek = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_peek"); + pContext->pulse.pa_stream_drop = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_drop"); + pContext->pulse.pa_stream_writable_size = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_writable_size"); + pContext->pulse.pa_stream_readable_size = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_readable_size"); +#else + /* This strange assignment system is just for type safety. */ + ma_pa_mainloop_new_proc _pa_mainloop_new = pa_mainloop_new; + ma_pa_mainloop_free_proc _pa_mainloop_free = pa_mainloop_free; + ma_pa_mainloop_get_api_proc _pa_mainloop_get_api = pa_mainloop_get_api; + ma_pa_mainloop_iterate_proc _pa_mainloop_iterate = pa_mainloop_iterate; + ma_pa_mainloop_wakeup_proc _pa_mainloop_wakeup = pa_mainloop_wakeup; + ma_pa_context_new_proc _pa_context_new = pa_context_new; + ma_pa_context_unref_proc _pa_context_unref = pa_context_unref; + ma_pa_context_connect_proc _pa_context_connect = pa_context_connect; + ma_pa_context_disconnect_proc _pa_context_disconnect = pa_context_disconnect; + ma_pa_context_set_state_callback_proc _pa_context_set_state_callback = pa_context_set_state_callback; + ma_pa_context_get_state_proc _pa_context_get_state = pa_context_get_state; + ma_pa_context_get_sink_info_list_proc _pa_context_get_sink_info_list = pa_context_get_sink_info_list; + ma_pa_context_get_source_info_list_proc _pa_context_get_source_info_list = pa_context_get_source_info_list; + ma_pa_context_get_sink_info_by_name_proc _pa_context_get_sink_info_by_name = pa_context_get_sink_info_by_name; + ma_pa_context_get_source_info_by_name_proc _pa_context_get_source_info_by_name= pa_context_get_source_info_by_name; + ma_pa_operation_unref_proc _pa_operation_unref = pa_operation_unref; + ma_pa_operation_get_state_proc _pa_operation_get_state = pa_operation_get_state; + ma_pa_channel_map_init_extend_proc _pa_channel_map_init_extend = pa_channel_map_init_extend; + ma_pa_channel_map_valid_proc _pa_channel_map_valid = pa_channel_map_valid; + ma_pa_channel_map_compatible_proc _pa_channel_map_compatible = pa_channel_map_compatible; + ma_pa_stream_new_proc _pa_stream_new = pa_stream_new; + ma_pa_stream_unref_proc _pa_stream_unref = pa_stream_unref; + ma_pa_stream_connect_playback_proc _pa_stream_connect_playback = pa_stream_connect_playback; + ma_pa_stream_connect_record_proc _pa_stream_connect_record = pa_stream_connect_record; + ma_pa_stream_disconnect_proc _pa_stream_disconnect = pa_stream_disconnect; + ma_pa_stream_get_state_proc _pa_stream_get_state = pa_stream_get_state; + ma_pa_stream_get_sample_spec_proc _pa_stream_get_sample_spec = pa_stream_get_sample_spec; + ma_pa_stream_get_channel_map_proc _pa_stream_get_channel_map = pa_stream_get_channel_map; + ma_pa_stream_get_buffer_attr_proc _pa_stream_get_buffer_attr = pa_stream_get_buffer_attr; + ma_pa_stream_set_buffer_attr_proc _pa_stream_set_buffer_attr = pa_stream_set_buffer_attr; + ma_pa_stream_get_device_name_proc _pa_stream_get_device_name = pa_stream_get_device_name; + ma_pa_stream_set_write_callback_proc _pa_stream_set_write_callback = pa_stream_set_write_callback; + ma_pa_stream_set_read_callback_proc _pa_stream_set_read_callback = pa_stream_set_read_callback; + ma_pa_stream_flush_proc _pa_stream_flush = pa_stream_flush; + ma_pa_stream_drain_proc _pa_stream_drain = pa_stream_drain; + ma_pa_stream_is_corked_proc _pa_stream_is_corked = pa_stream_is_corked; + ma_pa_stream_cork_proc _pa_stream_cork = pa_stream_cork; + ma_pa_stream_trigger_proc _pa_stream_trigger = pa_stream_trigger; + ma_pa_stream_begin_write_proc _pa_stream_begin_write = pa_stream_begin_write; + ma_pa_stream_write_proc _pa_stream_write = pa_stream_write; + ma_pa_stream_peek_proc _pa_stream_peek = pa_stream_peek; + ma_pa_stream_drop_proc _pa_stream_drop = pa_stream_drop; + ma_pa_stream_writable_size_proc _pa_stream_writable_size = pa_stream_writable_size; + ma_pa_stream_readable_size_proc _pa_stream_readable_size = pa_stream_readable_size; + + pContext->pulse.pa_mainloop_new = (ma_proc)_pa_mainloop_new; + pContext->pulse.pa_mainloop_free = (ma_proc)_pa_mainloop_free; + pContext->pulse.pa_mainloop_get_api = (ma_proc)_pa_mainloop_get_api; + pContext->pulse.pa_mainloop_iterate = (ma_proc)_pa_mainloop_iterate; + pContext->pulse.pa_mainloop_wakeup = (ma_proc)_pa_mainloop_wakeup; + pContext->pulse.pa_context_new = (ma_proc)_pa_context_new; + pContext->pulse.pa_context_unref = (ma_proc)_pa_context_unref; + pContext->pulse.pa_context_connect = (ma_proc)_pa_context_connect; + pContext->pulse.pa_context_disconnect = (ma_proc)_pa_context_disconnect; + pContext->pulse.pa_context_set_state_callback = (ma_proc)_pa_context_set_state_callback; + pContext->pulse.pa_context_get_state = (ma_proc)_pa_context_get_state; + pContext->pulse.pa_context_get_sink_info_list = (ma_proc)_pa_context_get_sink_info_list; + pContext->pulse.pa_context_get_source_info_list = (ma_proc)_pa_context_get_source_info_list; + pContext->pulse.pa_context_get_sink_info_by_name = (ma_proc)_pa_context_get_sink_info_by_name; + pContext->pulse.pa_context_get_source_info_by_name = (ma_proc)_pa_context_get_source_info_by_name; + pContext->pulse.pa_operation_unref = (ma_proc)_pa_operation_unref; + pContext->pulse.pa_operation_get_state = (ma_proc)_pa_operation_get_state; + pContext->pulse.pa_channel_map_init_extend = (ma_proc)_pa_channel_map_init_extend; + pContext->pulse.pa_channel_map_valid = (ma_proc)_pa_channel_map_valid; + pContext->pulse.pa_channel_map_compatible = (ma_proc)_pa_channel_map_compatible; + pContext->pulse.pa_stream_new = (ma_proc)_pa_stream_new; + pContext->pulse.pa_stream_unref = (ma_proc)_pa_stream_unref; + pContext->pulse.pa_stream_connect_playback = (ma_proc)_pa_stream_connect_playback; + pContext->pulse.pa_stream_connect_record = (ma_proc)_pa_stream_connect_record; + pContext->pulse.pa_stream_disconnect = (ma_proc)_pa_stream_disconnect; + pContext->pulse.pa_stream_get_state = (ma_proc)_pa_stream_get_state; + pContext->pulse.pa_stream_get_sample_spec = (ma_proc)_pa_stream_get_sample_spec; + pContext->pulse.pa_stream_get_channel_map = (ma_proc)_pa_stream_get_channel_map; + pContext->pulse.pa_stream_get_buffer_attr = (ma_proc)_pa_stream_get_buffer_attr; + pContext->pulse.pa_stream_set_buffer_attr = (ma_proc)_pa_stream_set_buffer_attr; + pContext->pulse.pa_stream_get_device_name = (ma_proc)_pa_stream_get_device_name; + pContext->pulse.pa_stream_set_write_callback = (ma_proc)_pa_stream_set_write_callback; + pContext->pulse.pa_stream_set_read_callback = (ma_proc)_pa_stream_set_read_callback; + pContext->pulse.pa_stream_flush = (ma_proc)_pa_stream_flush; + pContext->pulse.pa_stream_drain = (ma_proc)_pa_stream_drain; + pContext->pulse.pa_stream_is_corked = (ma_proc)_pa_stream_is_corked; + pContext->pulse.pa_stream_cork = (ma_proc)_pa_stream_cork; + pContext->pulse.pa_stream_trigger = (ma_proc)_pa_stream_trigger; + pContext->pulse.pa_stream_begin_write = (ma_proc)_pa_stream_begin_write; + pContext->pulse.pa_stream_write = (ma_proc)_pa_stream_write; + pContext->pulse.pa_stream_peek = (ma_proc)_pa_stream_peek; + pContext->pulse.pa_stream_drop = (ma_proc)_pa_stream_drop; + pContext->pulse.pa_stream_writable_size = (ma_proc)_pa_stream_writable_size; + pContext->pulse.pa_stream_readable_size = (ma_proc)_pa_stream_readable_size; +#endif + + pContext->onUninit = ma_context_uninit__pulse; + pContext->onDeviceIDEqual = ma_context_is_device_id_equal__pulse; + pContext->onEnumDevices = ma_context_enumerate_devices__pulse; + pContext->onGetDeviceInfo = ma_context_get_device_info__pulse; + pContext->onDeviceInit = ma_device_init__pulse; + pContext->onDeviceUninit = ma_device_uninit__pulse; + pContext->onDeviceStart = NULL; + pContext->onDeviceStop = NULL; + pContext->onDeviceMainLoop = ma_device_main_loop__pulse; - /* Only supporting native-endian. */ - if ((ma_is_little_endian() && (pDescription->mFormatFlags & kAudioFormatFlagIsBigEndian) != 0) || (ma_is_big_endian() && (pDescription->mFormatFlags & kAudioFormatFlagIsBigEndian) == 0)) { - return MA_FORMAT_NOT_SUPPORTED; + if (pConfig->pulse.pApplicationName) { + pContext->pulse.pApplicationName = ma_copy_string(pConfig->pulse.pApplicationName, &pContext->allocationCallbacks); } + if (pConfig->pulse.pServerName) { + pContext->pulse.pServerName = ma_copy_string(pConfig->pulse.pServerName, &pContext->allocationCallbacks); + } + pContext->pulse.tryAutoSpawn = pConfig->pulse.tryAutoSpawn; - /* We are not currently supporting non-interleaved formats (this will be added in a future version of miniaudio). */ - /*if ((pDescription->mFormatFlags & kAudioFormatFlagIsNonInterleaved) != 0) { - return MA_FORMAT_NOT_SUPPORTED; - }*/ + /* + Although we have found the libpulse library, it doesn't necessarily mean PulseAudio is useable. We need to initialize + and connect a dummy PulseAudio context to test PulseAudio's usability. + */ + { + ma_pa_mainloop* pMainLoop; + ma_pa_mainloop_api* pAPI; + ma_pa_context* pPulseContext; + int error; - if ((pDescription->mFormatFlags & kLinearPCMFormatFlagIsFloat) != 0) { - if (pDescription->mBitsPerChannel == 32) { - *pFormatOut = ma_format_f32; - return MA_SUCCESS; - } - } else { - if ((pDescription->mFormatFlags & kLinearPCMFormatFlagIsSignedInteger) != 0) { - if (pDescription->mBitsPerChannel == 16) { - *pFormatOut = ma_format_s16; - return MA_SUCCESS; - } else if (pDescription->mBitsPerChannel == 24) { - if (pDescription->mBytesPerFrame == (pDescription->mBitsPerChannel/8 * pDescription->mChannelsPerFrame)) { - *pFormatOut = ma_format_s24; - return MA_SUCCESS; - } else { - if (pDescription->mBytesPerFrame/pDescription->mChannelsPerFrame == sizeof(ma_int32)) { - /* TODO: Implement ma_format_s24_32. */ - /**pFormatOut = ma_format_s24_32;*/ - /*return MA_SUCCESS;*/ - return MA_FORMAT_NOT_SUPPORTED; - } - } - } else if (pDescription->mBitsPerChannel == 32) { - *pFormatOut = ma_format_s32; - return MA_SUCCESS; - } - } else { - if (pDescription->mBitsPerChannel == 8) { - *pFormatOut = ma_format_u8; - return MA_SUCCESS; - } + pMainLoop = ((ma_pa_mainloop_new_proc)pContext->pulse.pa_mainloop_new)(); + if (pMainLoop == NULL) { + ma_free(pContext->pulse.pServerName, &pContext->allocationCallbacks); + ma_free(pContext->pulse.pApplicationName, &pContext->allocationCallbacks); + #ifndef MA_NO_RUNTIME_LINKING + ma_dlclose(pContext, pContext->pulse.pulseSO); + #endif + return MA_NO_BACKEND; } - } - - /* Getting here means the format is not supported. */ - return MA_FORMAT_NOT_SUPPORTED; -} -ma_result ma_get_channel_map_from_AudioChannelLayout(AudioChannelLayout* pChannelLayout, ma_channel channelMap[MA_MAX_CHANNELS]) -{ - ma_assert(pChannelLayout != NULL); - - if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelDescriptions) { - UInt32 iChannel; - for (iChannel = 0; iChannel < pChannelLayout->mNumberChannelDescriptions; ++iChannel) { - channelMap[iChannel] = ma_channel_from_AudioChannelLabel(pChannelLayout->mChannelDescriptions[iChannel].mChannelLabel); + pAPI = ((ma_pa_mainloop_get_api_proc)pContext->pulse.pa_mainloop_get_api)(pMainLoop); + if (pAPI == NULL) { + ma_free(pContext->pulse.pServerName, &pContext->allocationCallbacks); + ma_free(pContext->pulse.pApplicationName, &pContext->allocationCallbacks); + ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); + #ifndef MA_NO_RUNTIME_LINKING + ma_dlclose(pContext, pContext->pulse.pulseSO); + #endif + return MA_NO_BACKEND; } - } else -#if 0 - if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelBitmap) { - /* This is the same kind of system that's used by Windows audio APIs. */ - UInt32 iChannel = 0; - UInt32 iBit; - AudioChannelBitmap bitmap = pChannelLayout->mChannelBitmap; - for (iBit = 0; iBit < 32; ++iBit) { - AudioChannelBitmap bit = bitmap & (1 << iBit); - if (bit != 0) { - channelMap[iChannel++] = ma_channel_from_AudioChannelBit(bit); - } + + pPulseContext = ((ma_pa_context_new_proc)pContext->pulse.pa_context_new)(pAPI, pContext->pulse.pApplicationName); + if (pPulseContext == NULL) { + ma_free(pContext->pulse.pServerName, &pContext->allocationCallbacks); + ma_free(pContext->pulse.pApplicationName, &pContext->allocationCallbacks); + ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); + #ifndef MA_NO_RUNTIME_LINKING + ma_dlclose(pContext, pContext->pulse.pulseSO); + #endif + return MA_NO_BACKEND; } - } else -#endif - { - /* - Need to use the tag to determine the channel map. For now I'm just assuming a default channel map, but later on this should - be updated to determine the mapping based on the tag. - */ - UInt32 channelCount = AudioChannelLayoutTag_GetNumberOfChannels(pChannelLayout->mChannelLayoutTag); - switch (pChannelLayout->mChannelLayoutTag) - { - case kAudioChannelLayoutTag_Mono: - case kAudioChannelLayoutTag_Stereo: - case kAudioChannelLayoutTag_StereoHeadphones: - case kAudioChannelLayoutTag_MatrixStereo: - case kAudioChannelLayoutTag_MidSide: - case kAudioChannelLayoutTag_XY: - case kAudioChannelLayoutTag_Binaural: - case kAudioChannelLayoutTag_Ambisonic_B_Format: - { - ma_get_standard_channel_map(ma_standard_channel_map_default, channelCount, channelMap); - } break; - - case kAudioChannelLayoutTag_Octagonal: - { - channelMap[7] = MA_CHANNEL_SIDE_RIGHT; - channelMap[6] = MA_CHANNEL_SIDE_LEFT; - } /* Intentional fallthrough. */ - case kAudioChannelLayoutTag_Hexagonal: - { - channelMap[5] = MA_CHANNEL_BACK_CENTER; - } /* Intentional fallthrough. */ - case kAudioChannelLayoutTag_Pentagonal: - { - channelMap[4] = MA_CHANNEL_FRONT_CENTER; - } /* Intentional fallghrough. */ - case kAudioChannelLayoutTag_Quadraphonic: - { - channelMap[3] = MA_CHANNEL_BACK_RIGHT; - channelMap[2] = MA_CHANNEL_BACK_LEFT; - channelMap[1] = MA_CHANNEL_RIGHT; - channelMap[0] = MA_CHANNEL_LEFT; - } break; - - /* TODO: Add support for more tags here. */ - - default: - { - ma_get_standard_channel_map(ma_standard_channel_map_default, channelCount, channelMap); - } break; + + error = ((ma_pa_context_connect_proc)pContext->pulse.pa_context_connect)(pPulseContext, pContext->pulse.pServerName, 0, NULL); + if (error != MA_PA_OK) { + ma_free(pContext->pulse.pServerName, &pContext->allocationCallbacks); + ma_free(pContext->pulse.pApplicationName, &pContext->allocationCallbacks); + ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)(pPulseContext); + ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); + #ifndef MA_NO_RUNTIME_LINKING + ma_dlclose(pContext, pContext->pulse.pulseSO); + #endif + return MA_NO_BACKEND; } + + ((ma_pa_context_disconnect_proc)pContext->pulse.pa_context_disconnect)(pPulseContext); + ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)(pPulseContext); + ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)(pMainLoop); } - + return MA_SUCCESS; } +#endif + + +/****************************************************************************** + +JACK Backend + +******************************************************************************/ +#ifdef MA_HAS_JACK + +/* It is assumed jack.h is available when compile-time linking is being used. */ +#ifdef MA_NO_RUNTIME_LINKING +#include + +typedef jack_nframes_t ma_jack_nframes_t; +typedef jack_options_t ma_jack_options_t; +typedef jack_status_t ma_jack_status_t; +typedef jack_client_t ma_jack_client_t; +typedef jack_port_t ma_jack_port_t; +typedef JackProcessCallback ma_JackProcessCallback; +typedef JackBufferSizeCallback ma_JackBufferSizeCallback; +typedef JackShutdownCallback ma_JackShutdownCallback; +#define MA_JACK_DEFAULT_AUDIO_TYPE JACK_DEFAULT_AUDIO_TYPE +#define ma_JackNoStartServer JackNoStartServer +#define ma_JackPortIsInput JackPortIsInput +#define ma_JackPortIsOutput JackPortIsOutput +#define ma_JackPortIsPhysical JackPortIsPhysical +#else +typedef ma_uint32 ma_jack_nframes_t; +typedef int ma_jack_options_t; +typedef int ma_jack_status_t; +typedef struct ma_jack_client_t ma_jack_client_t; +typedef struct ma_jack_port_t ma_jack_port_t; +typedef int (* ma_JackProcessCallback) (ma_jack_nframes_t nframes, void* arg); +typedef int (* ma_JackBufferSizeCallback)(ma_jack_nframes_t nframes, void* arg); +typedef void (* ma_JackShutdownCallback) (void* arg); +#define MA_JACK_DEFAULT_AUDIO_TYPE "32 bit float mono audio" +#define ma_JackNoStartServer 1 +#define ma_JackPortIsInput 1 +#define ma_JackPortIsOutput 2 +#define ma_JackPortIsPhysical 4 +#endif +typedef ma_jack_client_t* (* ma_jack_client_open_proc) (const char* client_name, ma_jack_options_t options, ma_jack_status_t* status, ...); +typedef int (* ma_jack_client_close_proc) (ma_jack_client_t* client); +typedef int (* ma_jack_client_name_size_proc) (); +typedef int (* ma_jack_set_process_callback_proc) (ma_jack_client_t* client, ma_JackProcessCallback process_callback, void* arg); +typedef int (* ma_jack_set_buffer_size_callback_proc)(ma_jack_client_t* client, ma_JackBufferSizeCallback bufsize_callback, void* arg); +typedef void (* ma_jack_on_shutdown_proc) (ma_jack_client_t* client, ma_JackShutdownCallback function, void* arg); +typedef ma_jack_nframes_t (* ma_jack_get_sample_rate_proc) (ma_jack_client_t* client); +typedef ma_jack_nframes_t (* ma_jack_get_buffer_size_proc) (ma_jack_client_t* client); +typedef const char** (* ma_jack_get_ports_proc) (ma_jack_client_t* client, const char* port_name_pattern, const char* type_name_pattern, unsigned long flags); +typedef int (* ma_jack_activate_proc) (ma_jack_client_t* client); +typedef int (* ma_jack_deactivate_proc) (ma_jack_client_t* client); +typedef int (* ma_jack_connect_proc) (ma_jack_client_t* client, const char* source_port, const char* destination_port); +typedef ma_jack_port_t* (* ma_jack_port_register_proc) (ma_jack_client_t* client, const char* port_name, const char* port_type, unsigned long flags, unsigned long buffer_size); +typedef const char* (* ma_jack_port_name_proc) (const ma_jack_port_t* port); +typedef void* (* ma_jack_port_get_buffer_proc) (ma_jack_port_t* port, ma_jack_nframes_t nframes); +typedef void (* ma_jack_free_proc) (void* ptr); -#if defined(MA_APPLE_DESKTOP) -ma_result ma_get_device_object_ids__coreaudio(ma_context* pContext, UInt32* pDeviceCount, AudioObjectID** ppDeviceObjectIDs) /* NOTE: Free the returned buffer with ma_free(). */ +static ma_result ma_context_open_client__jack(ma_context* pContext, ma_jack_client_t** ppClient) { - AudioObjectPropertyAddress propAddressDevices; - UInt32 deviceObjectsDataSize; - OSStatus status; - AudioObjectID* pDeviceObjectIDs; - - ma_assert(pContext != NULL); - ma_assert(pDeviceCount != NULL); - ma_assert(ppDeviceObjectIDs != NULL); + size_t maxClientNameSize; + char clientName[256]; + ma_jack_status_t status; + ma_jack_client_t* pClient; - /* Safety. */ - *pDeviceCount = 0; - *ppDeviceObjectIDs = NULL; - - propAddressDevices.mSelector = kAudioHardwarePropertyDevices; - propAddressDevices.mScope = kAudioObjectPropertyScopeGlobal; - propAddressDevices.mElement = kAudioObjectPropertyElementMaster; + MA_ASSERT(pContext != NULL); + MA_ASSERT(ppClient != NULL); - status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(kAudioObjectSystemObject, &propAddressDevices, 0, NULL, &deviceObjectsDataSize); - if (status != noErr) { - return ma_result_from_OSStatus(status); + if (ppClient) { + *ppClient = NULL; } - - pDeviceObjectIDs = (AudioObjectID*)ma_malloc(deviceObjectsDataSize); - if (pDeviceObjectIDs == NULL) { - return MA_OUT_OF_MEMORY; + + maxClientNameSize = ((ma_jack_client_name_size_proc)pContext->jack.jack_client_name_size)(); /* Includes null terminator. */ + ma_strncpy_s(clientName, ma_min(sizeof(clientName), maxClientNameSize), (pContext->jack.pClientName != NULL) ? pContext->jack.pClientName : "miniaudio", (size_t)-1); + + pClient = ((ma_jack_client_open_proc)pContext->jack.jack_client_open)(clientName, (pContext->jack.tryStartServer) ? 0 : ma_JackNoStartServer, &status, NULL); + if (pClient == NULL) { + return MA_FAILED_TO_OPEN_BACKEND_DEVICE; } - - status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(kAudioObjectSystemObject, &propAddressDevices, 0, NULL, &deviceObjectsDataSize, pDeviceObjectIDs); - if (status != noErr) { - ma_free(pDeviceObjectIDs); - return ma_result_from_OSStatus(status); + + if (ppClient) { + *ppClient = pClient; } - - *pDeviceCount = deviceObjectsDataSize / sizeof(AudioObjectID); - *ppDeviceObjectIDs = pDeviceObjectIDs; - (void)pContext; /* Unused. */ return MA_SUCCESS; } -ma_result ma_get_AudioObject_uid_as_CFStringRef(ma_context* pContext, AudioObjectID objectID, CFStringRef* pUID) +static ma_bool32 ma_context_is_device_id_equal__jack(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) { - AudioObjectPropertyAddress propAddress; - UInt32 dataSize; - OSStatus status; + MA_ASSERT(pContext != NULL); + MA_ASSERT(pID0 != NULL); + MA_ASSERT(pID1 != NULL); + (void)pContext; + + return pID0->jack == pID1->jack; +} - ma_assert(pContext != NULL); +static ma_result ma_context_enumerate_devices__jack(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +{ + ma_bool32 cbResult = MA_TRUE; - propAddress.mSelector = kAudioDevicePropertyDeviceUID; - propAddress.mScope = kAudioObjectPropertyScopeGlobal; - propAddress.mElement = kAudioObjectPropertyElementMaster; + MA_ASSERT(pContext != NULL); + MA_ASSERT(callback != NULL); - dataSize = sizeof(*pUID); - status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(objectID, &propAddress, 0, NULL, &dataSize, pUID); - if (status != noErr) { - return ma_result_from_OSStatus(status); + /* Playback. */ + if (cbResult) { + ma_device_info deviceInfo; + MA_ZERO_OBJECT(&deviceInfo); + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); + cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); } - + + /* Capture. */ + if (cbResult) { + ma_device_info deviceInfo; + MA_ZERO_OBJECT(&deviceInfo); + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); + cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); + } + return MA_SUCCESS; } -ma_result ma_get_AudioObject_uid(ma_context* pContext, AudioObjectID objectID, size_t bufferSize, char* bufferOut) +static ma_result ma_context_get_device_info__jack(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) { - CFStringRef uid; + ma_jack_client_t* pClient; ma_result result; + const char** ppPorts; - ma_assert(pContext != NULL); + MA_ASSERT(pContext != NULL); - result = ma_get_AudioObject_uid_as_CFStringRef(pContext, objectID, &uid); - if (result != MA_SUCCESS) { - return result; + /* No exclusive mode with the JACK backend. */ + if (shareMode == ma_share_mode_exclusive) { + return MA_SHARE_MODE_NOT_SUPPORTED; } - - if (!((ma_CFStringGetCString_proc)pContext->coreaudio.CFStringGetCString)(uid, bufferOut, bufferSize, kCFStringEncodingUTF8)) { - return MA_ERROR; + + if (pDeviceID != NULL && pDeviceID->jack != 0) { + return MA_NO_DEVICE; /* Don't know the device. */ } - - ((ma_CFRelease_proc)pContext->coreaudio.CFRelease)(uid); - return MA_SUCCESS; -} -ma_result ma_get_AudioObject_name(ma_context* pContext, AudioObjectID objectID, size_t bufferSize, char* bufferOut) -{ - AudioObjectPropertyAddress propAddress; - CFStringRef deviceName = NULL; - UInt32 dataSize; - OSStatus status; + /* Name / Description */ + if (deviceType == ma_device_type_playback) { + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); + } else { + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); + } - ma_assert(pContext != NULL); + /* Jack only supports f32 and has a specific channel count and sample rate. */ + pDeviceInfo->formatCount = 1; + pDeviceInfo->formats[0] = ma_format_f32; - propAddress.mSelector = kAudioDevicePropertyDeviceNameCFString; - propAddress.mScope = kAudioObjectPropertyScopeGlobal; - propAddress.mElement = kAudioObjectPropertyElementMaster; + /* The channel count and sample rate can only be determined by opening the device. */ + result = ma_context_open_client__jack(pContext, &pClient); + if (result != MA_SUCCESS) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[JACK] Failed to open client.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - dataSize = sizeof(deviceName); - status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(objectID, &propAddress, 0, NULL, &dataSize, &deviceName); - if (status != noErr) { - return ma_result_from_OSStatus(status); + pDeviceInfo->minSampleRate = ((ma_jack_get_sample_rate_proc)pContext->jack.jack_get_sample_rate)((ma_jack_client_t*)pClient); + pDeviceInfo->maxSampleRate = pDeviceInfo->minSampleRate; + + pDeviceInfo->minChannels = 0; + pDeviceInfo->maxChannels = 0; + + ppPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ((deviceType == ma_device_type_playback) ? ma_JackPortIsInput : ma_JackPortIsOutput)); + if (ppPorts == NULL) { + ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pClient); + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[JACK] Failed to query physical ports.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); } - - if (!((ma_CFStringGetCString_proc)pContext->coreaudio.CFStringGetCString)(deviceName, bufferOut, bufferSize, kCFStringEncodingUTF8)) { - return MA_ERROR; + + while (ppPorts[pDeviceInfo->minChannels] != NULL) { + pDeviceInfo->minChannels += 1; + pDeviceInfo->maxChannels += 1; } - - ((ma_CFRelease_proc)pContext->coreaudio.CFRelease)(deviceName); + + ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppPorts); + ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pClient); + + (void)pContext; return MA_SUCCESS; } -ma_bool32 ma_does_AudioObject_support_scope(ma_context* pContext, AudioObjectID deviceObjectID, AudioObjectPropertyScope scope) + +static void ma_device_uninit__jack(ma_device* pDevice) { - AudioObjectPropertyAddress propAddress; - UInt32 dataSize; - OSStatus status; - AudioBufferList* pBufferList; - ma_bool32 isSupported; + ma_context* pContext; - ma_assert(pContext != NULL); + MA_ASSERT(pDevice != NULL); - /* To know whether or not a device is an input device we need ot look at the stream configuration. If it has an output channel it's a playback device. */ - propAddress.mSelector = kAudioDevicePropertyStreamConfiguration; - propAddress.mScope = scope; - propAddress.mElement = kAudioObjectPropertyElementMaster; - - status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize); - if (status != noErr) { - return MA_FALSE; + pContext = pDevice->pContext; + MA_ASSERT(pContext != NULL); + + if (pDevice->jack.pClient != NULL) { + ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pDevice->jack.pClient); } - - pBufferList = (AudioBufferList*)ma_malloc(dataSize); - if (pBufferList == NULL) { - return MA_FALSE; /* Out of memory. */ + + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + ma__free_from_callbacks(pDevice->jack.pIntermediaryBufferCapture, &pDevice->pContext->allocationCallbacks); } - - status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pBufferList); - if (status != noErr) { - ma_free(pBufferList); - return MA_FALSE; + + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + ma__free_from_callbacks(pDevice->jack.pIntermediaryBufferPlayback, &pDevice->pContext->allocationCallbacks); } - isSupported = MA_FALSE; - if (pBufferList->mNumberBuffers > 0) { - isSupported = MA_TRUE; + if (pDevice->type == ma_device_type_duplex) { + ma_pcm_rb_uninit(&pDevice->jack.duplexRB); } - - ma_free(pBufferList); - return isSupported; } -ma_bool32 ma_does_AudioObject_support_playback(ma_context* pContext, AudioObjectID deviceObjectID) +static void ma_device__jack_shutdown_callback(void* pUserData) { - return ma_does_AudioObject_support_scope(pContext, deviceObjectID, kAudioObjectPropertyScopeOutput); + /* JACK died. Stop the device. */ + ma_device* pDevice = (ma_device*)pUserData; + MA_ASSERT(pDevice != NULL); + + ma_device_stop(pDevice); } -ma_bool32 ma_does_AudioObject_support_capture(ma_context* pContext, AudioObjectID deviceObjectID) +static int ma_device__jack_buffer_size_callback(ma_jack_nframes_t frameCount, void* pUserData) { - return ma_does_AudioObject_support_scope(pContext, deviceObjectID, kAudioObjectPropertyScopeInput); -} + ma_device* pDevice = (ma_device*)pUserData; + MA_ASSERT(pDevice != NULL); + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + size_t newBufferSize = frameCount * (pDevice->capture.internalChannels * ma_get_bytes_per_sample(pDevice->capture.internalFormat)); + float* pNewBuffer = (float*)ma__calloc_from_callbacks(newBufferSize, &pDevice->pContext->allocationCallbacks); + if (pNewBuffer == NULL) { + return MA_OUT_OF_MEMORY; + } -ma_result ma_get_AudioObject_stream_descriptions(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, UInt32* pDescriptionCount, AudioStreamRangedDescription** ppDescriptions) /* NOTE: Free the returned pointer with ma_free(). */ -{ - AudioObjectPropertyAddress propAddress; - UInt32 dataSize; - OSStatus status; - AudioStreamRangedDescription* pDescriptions; + ma__free_from_callbacks(pDevice->jack.pIntermediaryBufferCapture, &pDevice->pContext->allocationCallbacks); - ma_assert(pContext != NULL); - ma_assert(pDescriptionCount != NULL); - ma_assert(ppDescriptions != NULL); - - /* - TODO: Experiment with kAudioStreamPropertyAvailablePhysicalFormats instead of (or in addition to) kAudioStreamPropertyAvailableVirtualFormats. My - MacBook Pro uses s24/32 format, however, which miniaudio does not currently support. - */ - propAddress.mSelector = kAudioStreamPropertyAvailableVirtualFormats; /*kAudioStreamPropertyAvailablePhysicalFormats;*/ - propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput; - propAddress.mElement = kAudioObjectPropertyElementMaster; - - status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize); - if (status != noErr) { - return ma_result_from_OSStatus(status); + pDevice->jack.pIntermediaryBufferCapture = pNewBuffer; + pDevice->playback.internalPeriodSizeInFrames = frameCount; } - - pDescriptions = (AudioStreamRangedDescription*)ma_malloc(dataSize); - if (pDescriptions == NULL) { - return MA_OUT_OF_MEMORY; + + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + size_t newBufferSize = frameCount * (pDevice->playback.internalChannels * ma_get_bytes_per_sample(pDevice->playback.internalFormat)); + float* pNewBuffer = (float*)ma__calloc_from_callbacks(newBufferSize, &pDevice->pContext->allocationCallbacks); + if (pNewBuffer == NULL) { + return MA_OUT_OF_MEMORY; + } + + ma__free_from_callbacks(pDevice->jack.pIntermediaryBufferPlayback, &pDevice->pContext->allocationCallbacks); + + pDevice->jack.pIntermediaryBufferPlayback = pNewBuffer; + pDevice->playback.internalPeriodSizeInFrames = frameCount; } - - status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pDescriptions); - if (status != noErr) { - ma_free(pDescriptions); - return ma_result_from_OSStatus(status); + + return 0; +} + +static int ma_device__jack_process_callback(ma_jack_nframes_t frameCount, void* pUserData) +{ + ma_device* pDevice; + ma_context* pContext; + ma_uint32 iChannel; + + pDevice = (ma_device*)pUserData; + MA_ASSERT(pDevice != NULL); + + pContext = pDevice->pContext; + MA_ASSERT(pContext != NULL); + + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + /* Channels need to be interleaved. */ + for (iChannel = 0; iChannel < pDevice->capture.internalChannels; ++iChannel) { + const float* pSrc = (const float*)((ma_jack_port_get_buffer_proc)pContext->jack.jack_port_get_buffer)((ma_jack_port_t*)pDevice->jack.pPortsCapture[iChannel], frameCount); + if (pSrc != NULL) { + float* pDst = pDevice->jack.pIntermediaryBufferCapture + iChannel; + ma_jack_nframes_t iFrame; + for (iFrame = 0; iFrame < frameCount; ++iFrame) { + *pDst = *pSrc; + + pDst += pDevice->capture.internalChannels; + pSrc += 1; + } + } + } + + if (pDevice->type == ma_device_type_duplex) { + ma_device__handle_duplex_callback_capture(pDevice, frameCount, pDevice->jack.pIntermediaryBufferCapture, &pDevice->jack.duplexRB); + } else { + ma_device__send_frames_to_client(pDevice, frameCount, pDevice->jack.pIntermediaryBufferCapture); + } } - - *pDescriptionCount = dataSize / sizeof(*pDescriptions); - *ppDescriptions = pDescriptions; - return MA_SUCCESS; -} + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + if (pDevice->type == ma_device_type_duplex) { + ma_device__handle_duplex_callback_playback(pDevice, frameCount, pDevice->jack.pIntermediaryBufferPlayback, &pDevice->jack.duplexRB); + } else { + ma_device__read_frames_from_client(pDevice, frameCount, pDevice->jack.pIntermediaryBufferPlayback); + } -ma_result ma_get_AudioObject_channel_layout(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, AudioChannelLayout** ppChannelLayout) /* NOTE: Free the returned pointer with ma_free(). */ -{ - AudioObjectPropertyAddress propAddress; - UInt32 dataSize; - OSStatus status; - AudioChannelLayout* pChannelLayout; + /* Channels need to be deinterleaved. */ + for (iChannel = 0; iChannel < pDevice->playback.internalChannels; ++iChannel) { + float* pDst = (float*)((ma_jack_port_get_buffer_proc)pContext->jack.jack_port_get_buffer)((ma_jack_port_t*)pDevice->jack.pPortsPlayback[iChannel], frameCount); + if (pDst != NULL) { + const float* pSrc = pDevice->jack.pIntermediaryBufferPlayback + iChannel; + ma_jack_nframes_t iFrame; + for (iFrame = 0; iFrame < frameCount; ++iFrame) { + *pDst = *pSrc; - ma_assert(pContext != NULL); - ma_assert(ppChannelLayout != NULL); - - *ppChannelLayout = NULL; /* Safety. */ - - propAddress.mSelector = kAudioDevicePropertyPreferredChannelLayout; - propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput; - propAddress.mElement = kAudioObjectPropertyElementMaster; - - status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize); - if (status != noErr) { - return ma_result_from_OSStatus(status); - } - - pChannelLayout = (AudioChannelLayout*)ma_malloc(dataSize); - if (pChannelLayout == NULL) { - return MA_OUT_OF_MEMORY; - } - - status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pChannelLayout); - if (status != noErr) { - ma_free(pChannelLayout); - return ma_result_from_OSStatus(status); + pDst += 1; + pSrc += pDevice->playback.internalChannels; + } + } + } } - - *ppChannelLayout = pChannelLayout; - return MA_SUCCESS; + + return 0; } -ma_result ma_get_AudioObject_channel_count(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32* pChannelCount) +static ma_result ma_device_init__jack(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) { - AudioChannelLayout* pChannelLayout; ma_result result; + ma_uint32 periods; + ma_uint32 periodSizeInFrames; - ma_assert(pContext != NULL); - ma_assert(pChannelCount != NULL); - - *pChannelCount = 0; /* Safety. */ + MA_ASSERT(pContext != NULL); + MA_ASSERT(pConfig != NULL); + MA_ASSERT(pDevice != NULL); - result = ma_get_AudioObject_channel_layout(pContext, deviceObjectID, deviceType, &pChannelLayout); - if (result != MA_SUCCESS) { - return result; - } - - if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelDescriptions) { - *pChannelCount = pChannelLayout->mNumberChannelDescriptions; - } else if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelBitmap) { - *pChannelCount = ma_count_set_bits(pChannelLayout->mChannelBitmap); - } else { - *pChannelCount = AudioChannelLayoutTag_GetNumberOfChannels(pChannelLayout->mChannelLayoutTag); + if (pConfig->deviceType == ma_device_type_loopback) { + return MA_DEVICE_TYPE_NOT_SUPPORTED; } - - ma_free(pChannelLayout); - return MA_SUCCESS; -} -ma_result ma_get_AudioObject_channel_map(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_channel channelMap[MA_MAX_CHANNELS]) -{ - AudioChannelLayout* pChannelLayout; - ma_result result; + /* Only supporting default devices with JACK. */ + if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.pDeviceID != NULL && pConfig->playback.pDeviceID->jack != 0) || + ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.pDeviceID != NULL && pConfig->capture.pDeviceID->jack != 0)) { + return MA_NO_DEVICE; + } - ma_assert(pContext != NULL); - - result = ma_get_AudioObject_channel_layout(pContext, deviceObjectID, deviceType, &pChannelLayout); - if (result != MA_SUCCESS) { - return result; /* Rather than always failing here, would it be more robust to simply assume a default? */ + /* No exclusive mode with the JACK backend. */ + if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) || + ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) { + return MA_SHARE_MODE_NOT_SUPPORTED; } - - result = ma_get_channel_map_from_AudioChannelLayout(pChannelLayout, channelMap); + + /* Open the client. */ + result = ma_context_open_client__jack(pContext, (ma_jack_client_t**)&pDevice->jack.pClient); if (result != MA_SUCCESS) { - ma_free(pChannelLayout); - return result; + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to open client.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); } - - ma_free(pChannelLayout); - return result; -} - -ma_result ma_get_AudioObject_sample_rates(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, UInt32* pSampleRateRangesCount, AudioValueRange** ppSampleRateRanges) /* NOTE: Free the returned pointer with ma_free(). */ -{ - AudioObjectPropertyAddress propAddress; - UInt32 dataSize; - OSStatus status; - AudioValueRange* pSampleRateRanges; - ma_assert(pContext != NULL); - ma_assert(pSampleRateRangesCount != NULL); - ma_assert(ppSampleRateRanges != NULL); - - /* Safety. */ - *pSampleRateRangesCount = 0; - *ppSampleRateRanges = NULL; - - propAddress.mSelector = kAudioDevicePropertyAvailableNominalSampleRates; - propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput; - propAddress.mElement = kAudioObjectPropertyElementMaster; - - status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize); - if (status != noErr) { - return ma_result_from_OSStatus(status); - } - - pSampleRateRanges = (AudioValueRange*)ma_malloc(dataSize); - if (pSampleRateRanges == NULL) { - return MA_OUT_OF_MEMORY; + /* Callbacks. */ + if (((ma_jack_set_process_callback_proc)pContext->jack.jack_set_process_callback)((ma_jack_client_t*)pDevice->jack.pClient, ma_device__jack_process_callback, pDevice) != 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to set process callback.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); } - - status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pSampleRateRanges); - if (status != noErr) { - ma_free(pSampleRateRanges); - return ma_result_from_OSStatus(status); + if (((ma_jack_set_buffer_size_callback_proc)pContext->jack.jack_set_buffer_size_callback)((ma_jack_client_t*)pDevice->jack.pClient, ma_device__jack_buffer_size_callback, pDevice) != 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to set buffer size callback.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); } - - *pSampleRateRangesCount = dataSize / sizeof(*pSampleRateRanges); - *ppSampleRateRanges = pSampleRateRanges; - return MA_SUCCESS; -} -ma_result ma_get_AudioObject_get_closest_sample_rate(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32 sampleRateIn, ma_uint32* pSampleRateOut) -{ - UInt32 sampleRateRangeCount; - AudioValueRange* pSampleRateRanges; - ma_result result; + ((ma_jack_on_shutdown_proc)pContext->jack.jack_on_shutdown)((ma_jack_client_t*)pDevice->jack.pClient, ma_device__jack_shutdown_callback, pDevice); - ma_assert(pContext != NULL); - ma_assert(pSampleRateOut != NULL); - - *pSampleRateOut = 0; /* Safety. */ - - result = ma_get_AudioObject_sample_rates(pContext, deviceObjectID, deviceType, &sampleRateRangeCount, &pSampleRateRanges); - if (result != MA_SUCCESS) { - return result; - } - - if (sampleRateRangeCount == 0) { - ma_free(pSampleRateRanges); - return MA_ERROR; /* Should never hit this case should we? */ - } + + /* The buffer size in frames can change. */ + periods = pConfig->periods; + periodSizeInFrames = ((ma_jack_get_buffer_size_proc)pContext->jack.jack_get_buffer_size)((ma_jack_client_t*)pDevice->jack.pClient); - if (sampleRateIn == 0) { - /* Search in order of miniaudio's preferred priority. */ - UInt32 iMALSampleRate; - for (iMALSampleRate = 0; iMALSampleRate < ma_countof(g_maStandardSampleRatePriorities); ++iMALSampleRate) { - ma_uint32 malSampleRate = g_maStandardSampleRatePriorities[iMALSampleRate]; - UInt32 iCASampleRate; - for (iCASampleRate = 0; iCASampleRate < sampleRateRangeCount; ++iCASampleRate) { - AudioValueRange caSampleRate = pSampleRateRanges[iCASampleRate]; - if (caSampleRate.mMinimum <= malSampleRate && caSampleRate.mMaximum >= malSampleRate) { - *pSampleRateOut = malSampleRate; - ma_free(pSampleRateRanges); - return MA_SUCCESS; - } + if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { + const char** ppPorts; + + pDevice->capture.internalFormat = ma_format_f32; + pDevice->capture.internalChannels = 0; + pDevice->capture.internalSampleRate = ((ma_jack_get_sample_rate_proc)pContext->jack.jack_get_sample_rate)((ma_jack_client_t*)pDevice->jack.pClient); + ma_get_standard_channel_map(ma_standard_channel_map_alsa, pDevice->capture.internalChannels, pDevice->capture.internalChannelMap); + + ppPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsOutput); + if (ppPorts == NULL) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to query physical ports.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } + + while (ppPorts[pDevice->capture.internalChannels] != NULL) { + char name[64]; + ma_strcpy_s(name, sizeof(name), "capture"); + ma_itoa_s((int)pDevice->capture.internalChannels, name+7, sizeof(name)-7, 10); /* 7 = length of "capture" */ + + pDevice->jack.pPortsCapture[pDevice->capture.internalChannels] = ((ma_jack_port_register_proc)pContext->jack.jack_port_register)((ma_jack_client_t*)pDevice->jack.pClient, name, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsInput, 0); + if (pDevice->jack.pPortsCapture[pDevice->capture.internalChannels] == NULL) { + ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppPorts); + ma_device_uninit__jack(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to register ports.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); } + + pDevice->capture.internalChannels += 1; } - - /* - If we get here it means none of miniaudio's standard sample rates matched any of the supported sample rates from the device. In this - case we just fall back to the first one reported by Core Audio. - */ - ma_assert(sampleRateRangeCount > 0); - - *pSampleRateOut = pSampleRateRanges[0].mMinimum; - ma_free(pSampleRateRanges); - return MA_SUCCESS; - } else { - /* Find the closest match to this sample rate. */ - UInt32 currentAbsoluteDifference = INT32_MAX; - UInt32 iCurrentClosestRange = (UInt32)-1; - UInt32 iRange; - for (iRange = 0; iRange < sampleRateRangeCount; ++iRange) { - if (pSampleRateRanges[iRange].mMinimum <= sampleRateIn && pSampleRateRanges[iRange].mMaximum >= sampleRateIn) { - *pSampleRateOut = sampleRateIn; - ma_free(pSampleRateRanges); - return MA_SUCCESS; - } else { - UInt32 absoluteDifference; - if (pSampleRateRanges[iRange].mMinimum > sampleRateIn) { - absoluteDifference = pSampleRateRanges[iRange].mMinimum - sampleRateIn; - } else { - absoluteDifference = sampleRateIn - pSampleRateRanges[iRange].mMaximum; - } - - if (currentAbsoluteDifference > absoluteDifference) { - currentAbsoluteDifference = absoluteDifference; - iCurrentClosestRange = iRange; - } + + ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppPorts); + + pDevice->capture.internalPeriodSizeInFrames = periodSizeInFrames; + pDevice->capture.internalPeriods = periods; + + pDevice->jack.pIntermediaryBufferCapture = (float*)ma__calloc_from_callbacks(pDevice->capture.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels), &pContext->allocationCallbacks); + if (pDevice->jack.pIntermediaryBufferCapture == NULL) { + ma_device_uninit__jack(pDevice); + return MA_OUT_OF_MEMORY; + } + } + + if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { + const char** ppPorts; + + pDevice->playback.internalFormat = ma_format_f32; + pDevice->playback.internalChannels = 0; + pDevice->playback.internalSampleRate = ((ma_jack_get_sample_rate_proc)pContext->jack.jack_get_sample_rate)((ma_jack_client_t*)pDevice->jack.pClient); + ma_get_standard_channel_map(ma_standard_channel_map_alsa, pDevice->playback.internalChannels, pDevice->playback.internalChannelMap); + + ppPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsInput); + if (ppPorts == NULL) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to query physical ports.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } + + while (ppPorts[pDevice->playback.internalChannels] != NULL) { + char name[64]; + ma_strcpy_s(name, sizeof(name), "playback"); + ma_itoa_s((int)pDevice->playback.internalChannels, name+8, sizeof(name)-8, 10); /* 8 = length of "playback" */ + + pDevice->jack.pPortsPlayback[pDevice->playback.internalChannels] = ((ma_jack_port_register_proc)pContext->jack.jack_port_register)((ma_jack_client_t*)pDevice->jack.pClient, name, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsOutput, 0); + if (pDevice->jack.pPortsPlayback[pDevice->playback.internalChannels] == NULL) { + ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppPorts); + ma_device_uninit__jack(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to register ports.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } + + pDevice->playback.internalChannels += 1; + } + + ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppPorts); + + pDevice->playback.internalPeriodSizeInFrames = periodSizeInFrames; + pDevice->playback.internalPeriods = periods; + + pDevice->jack.pIntermediaryBufferPlayback = (float*)ma__calloc_from_callbacks(pDevice->playback.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels), &pContext->allocationCallbacks); + if (pDevice->jack.pIntermediaryBufferPlayback == NULL) { + ma_device_uninit__jack(pDevice); + return MA_OUT_OF_MEMORY; + } + } + + if (pDevice->type == ma_device_type_duplex) { + ma_uint32 rbSizeInFrames = (ma_uint32)ma_calculate_frame_count_after_resampling(pDevice->sampleRate, pDevice->capture.internalSampleRate, pDevice->capture.internalPeriodSizeInFrames * pDevice->capture.internalPeriods); + result = ma_pcm_rb_init(pDevice->capture.format, pDevice->capture.channels, rbSizeInFrames, NULL, &pDevice->pContext->allocationCallbacks, &pDevice->jack.duplexRB); + if (result != MA_SUCCESS) { + ma_device_uninit__jack(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to initialize ring buffer.", result); + } + + /* We need a period to act as a buffer for cases where the playback and capture device's end up desyncing. */ + { + ma_uint32 marginSizeInFrames = rbSizeInFrames / pDevice->capture.internalPeriods; + void* pMarginData; + ma_pcm_rb_acquire_write(&pDevice->jack.duplexRB, &marginSizeInFrames, &pMarginData); + { + MA_ZERO_MEMORY(pMarginData, marginSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels)); } + ma_pcm_rb_commit_write(&pDevice->jack.duplexRB, marginSizeInFrames, pMarginData); } - - ma_assert(iCurrentClosestRange != (UInt32)-1); - - *pSampleRateOut = pSampleRateRanges[iCurrentClosestRange].mMinimum; - ma_free(pSampleRateRanges); - return MA_SUCCESS; } - - /* Should never get here, but it would mean we weren't able to find any suitable sample rates. */ - /*ma_free(pSampleRateRanges);*/ - /*return MA_ERROR;*/ + + return MA_SUCCESS; } -ma_result ma_get_AudioObject_closest_buffer_size_in_frames(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32 bufferSizeInFramesIn, ma_uint32* pBufferSizeInFramesOut) +static ma_result ma_device_start__jack(ma_device* pDevice) { - AudioObjectPropertyAddress propAddress; - AudioValueRange bufferSizeRange; - UInt32 dataSize; - OSStatus status; + ma_context* pContext = pDevice->pContext; + int resultJACK; + size_t i; - ma_assert(pContext != NULL); - ma_assert(pBufferSizeInFramesOut != NULL); - - *pBufferSizeInFramesOut = 0; /* Safety. */ - - propAddress.mSelector = kAudioDevicePropertyBufferFrameSizeRange; - propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput; - propAddress.mElement = kAudioObjectPropertyElementMaster; + resultJACK = ((ma_jack_activate_proc)pContext->jack.jack_activate)((ma_jack_client_t*)pDevice->jack.pClient); + if (resultJACK != 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to activate the JACK client.", MA_FAILED_TO_START_BACKEND_DEVICE); + } - dataSize = sizeof(bufferSizeRange); - status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, &bufferSizeRange); - if (status != noErr) { - return ma_result_from_OSStatus(status); + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + const char** ppServerPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsOutput); + if (ppServerPorts == NULL) { + ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to retrieve physical ports.", MA_ERROR); + } + + for (i = 0; ppServerPorts[i] != NULL; ++i) { + const char* pServerPort = ppServerPorts[i]; + const char* pClientPort = ((ma_jack_port_name_proc)pContext->jack.jack_port_name)((ma_jack_port_t*)pDevice->jack.pPortsCapture[i]); + + resultJACK = ((ma_jack_connect_proc)pContext->jack.jack_connect)((ma_jack_client_t*)pDevice->jack.pClient, pServerPort, pClientPort); + if (resultJACK != 0) { + ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts); + ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to connect ports.", MA_ERROR); + } + } + + ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts); } - /* This is just a clamp. */ - if (bufferSizeInFramesIn < bufferSizeRange.mMinimum) { - *pBufferSizeInFramesOut = (ma_uint32)bufferSizeRange.mMinimum; - } else if (bufferSizeInFramesIn > bufferSizeRange.mMaximum) { - *pBufferSizeInFramesOut = (ma_uint32)bufferSizeRange.mMaximum; - } else { - *pBufferSizeInFramesOut = bufferSizeInFramesIn; + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + const char** ppServerPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsInput); + if (ppServerPorts == NULL) { + ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to retrieve physical ports.", MA_ERROR); + } + + for (i = 0; ppServerPorts[i] != NULL; ++i) { + const char* pServerPort = ppServerPorts[i]; + const char* pClientPort = ((ma_jack_port_name_proc)pContext->jack.jack_port_name)((ma_jack_port_t*)pDevice->jack.pPortsPlayback[i]); + + resultJACK = ((ma_jack_connect_proc)pContext->jack.jack_connect)((ma_jack_client_t*)pDevice->jack.pClient, pClientPort, pServerPort); + if (resultJACK != 0) { + ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts); + ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to connect ports.", MA_ERROR); + } + } + + ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts); } return MA_SUCCESS; } -ma_result ma_set_AudioObject_buffer_size_in_frames(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32* pBufferSizeInOut) +static ma_result ma_device_stop__jack(ma_device* pDevice) { - ma_result result; - ma_uint32 chosenBufferSizeInFrames; - AudioObjectPropertyAddress propAddress; - UInt32 dataSize; - OSStatus status; - - ma_assert(pContext != NULL); + ma_context* pContext = pDevice->pContext; + ma_stop_proc onStop; - result = ma_get_AudioObject_closest_buffer_size_in_frames(pContext, deviceObjectID, deviceType, *pBufferSizeInOut, &chosenBufferSizeInFrames); - if (result != MA_SUCCESS) { - return result; + if (((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient) != 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] An error occurred when deactivating the JACK client.", MA_ERROR); } - - /* Try setting the size of the buffer... If this fails we just use whatever is currently set. */ - propAddress.mSelector = kAudioDevicePropertyBufferFrameSize; - propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput; - propAddress.mElement = kAudioObjectPropertyElementMaster; - - ((ma_AudioObjectSetPropertyData_proc)pContext->coreaudio.AudioObjectSetPropertyData)(deviceObjectID, &propAddress, 0, NULL, sizeof(chosenBufferSizeInFrames), &chosenBufferSizeInFrames); - /* Get the actual size of the buffer. */ - dataSize = sizeof(*pBufferSizeInOut); - status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, &chosenBufferSizeInFrames); - if (status != noErr) { - return ma_result_from_OSStatus(status); + onStop = pDevice->onStop; + if (onStop) { + onStop(pDevice); } - - *pBufferSizeInOut = chosenBufferSizeInFrames; + return MA_SUCCESS; } -ma_result ma_find_AudioObjectID(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, AudioObjectID* pDeviceObjectID) +static ma_result ma_context_uninit__jack(ma_context* pContext) { - ma_assert(pContext != NULL); - ma_assert(pDeviceObjectID != NULL); + MA_ASSERT(pContext != NULL); + MA_ASSERT(pContext->backend == ma_backend_jack); - /* Safety. */ - *pDeviceObjectID = 0; - - if (pDeviceID == NULL) { - /* Default device. */ - AudioObjectPropertyAddress propAddressDefaultDevice; - UInt32 defaultDeviceObjectIDSize = sizeof(AudioObjectID); - AudioObjectID defaultDeviceObjectID; - OSStatus status; + ma_free(pContext->jack.pClientName, &pContext->allocationCallbacks); + pContext->jack.pClientName = NULL; - propAddressDefaultDevice.mScope = kAudioObjectPropertyScopeGlobal; - propAddressDefaultDevice.mElement = kAudioObjectPropertyElementMaster; - if (deviceType == ma_device_type_playback) { - propAddressDefaultDevice.mSelector = kAudioHardwarePropertyDefaultOutputDevice; - } else { - propAddressDefaultDevice.mSelector = kAudioHardwarePropertyDefaultInputDevice; - } - - defaultDeviceObjectIDSize = sizeof(AudioObjectID); - status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(kAudioObjectSystemObject, &propAddressDefaultDevice, 0, NULL, &defaultDeviceObjectIDSize, &defaultDeviceObjectID); - if (status == noErr) { - *pDeviceObjectID = defaultDeviceObjectID; - return MA_SUCCESS; - } - } else { - /* Explicit device. */ - UInt32 deviceCount; - AudioObjectID* pDeviceObjectIDs; - ma_result result; - UInt32 iDevice; +#ifndef MA_NO_RUNTIME_LINKING + ma_dlclose(pContext, pContext->jack.jackSO); +#endif - result = ma_get_device_object_ids__coreaudio(pContext, &deviceCount, &pDeviceObjectIDs); - if (result != MA_SUCCESS) { - return result; - } - - for (iDevice = 0; iDevice < deviceCount; ++iDevice) { - AudioObjectID deviceObjectID = pDeviceObjectIDs[iDevice]; - - char uid[256]; - if (ma_get_AudioObject_uid(pContext, deviceObjectID, sizeof(uid), uid) != MA_SUCCESS) { - continue; - } - - if (deviceType == ma_device_type_playback) { - if (ma_does_AudioObject_support_playback(pContext, deviceObjectID)) { - if (strcmp(uid, pDeviceID->coreaudio) == 0) { - *pDeviceObjectID = deviceObjectID; - ma_free(pDeviceObjectIDs); - return MA_SUCCESS; - } - } - } else { - if (ma_does_AudioObject_support_capture(pContext, deviceObjectID)) { - if (strcmp(uid, pDeviceID->coreaudio) == 0) { - *pDeviceObjectID = deviceObjectID; - ma_free(pDeviceObjectIDs); - return MA_SUCCESS; - } - } - } + return MA_SUCCESS; +} + +static ma_result ma_context_init__jack(const ma_context_config* pConfig, ma_context* pContext) +{ +#ifndef MA_NO_RUNTIME_LINKING + const char* libjackNames[] = { +#ifdef MA_WIN32 + "libjack.dll" +#else + "libjack.so", + "libjack.so.0" +#endif + }; + size_t i; + + for (i = 0; i < ma_countof(libjackNames); ++i) { + pContext->jack.jackSO = ma_dlopen(pContext, libjackNames[i]); + if (pContext->jack.jackSO != NULL) { + break; } + } - ma_free(pDeviceObjectIDs); + if (pContext->jack.jackSO == NULL) { + return MA_NO_BACKEND; } - - /* If we get here it means we couldn't find the device. */ - return MA_NO_DEVICE; -} + pContext->jack.jack_client_open = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_client_open"); + pContext->jack.jack_client_close = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_client_close"); + pContext->jack.jack_client_name_size = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_client_name_size"); + pContext->jack.jack_set_process_callback = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_set_process_callback"); + pContext->jack.jack_set_buffer_size_callback = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_set_buffer_size_callback"); + pContext->jack.jack_on_shutdown = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_on_shutdown"); + pContext->jack.jack_get_sample_rate = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_get_sample_rate"); + pContext->jack.jack_get_buffer_size = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_get_buffer_size"); + pContext->jack.jack_get_ports = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_get_ports"); + pContext->jack.jack_activate = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_activate"); + pContext->jack.jack_deactivate = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_deactivate"); + pContext->jack.jack_connect = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_connect"); + pContext->jack.jack_port_register = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_port_register"); + pContext->jack.jack_port_name = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_port_name"); + pContext->jack.jack_port_get_buffer = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_port_get_buffer"); + pContext->jack.jack_free = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_free"); +#else + /* + This strange assignment system is here just to ensure type safety of miniaudio's function pointer + types. If anything differs slightly the compiler should throw a warning. + */ + ma_jack_client_open_proc _jack_client_open = jack_client_open; + ma_jack_client_close_proc _jack_client_close = jack_client_close; + ma_jack_client_name_size_proc _jack_client_name_size = jack_client_name_size; + ma_jack_set_process_callback_proc _jack_set_process_callback = jack_set_process_callback; + ma_jack_set_buffer_size_callback_proc _jack_set_buffer_size_callback = jack_set_buffer_size_callback; + ma_jack_on_shutdown_proc _jack_on_shutdown = jack_on_shutdown; + ma_jack_get_sample_rate_proc _jack_get_sample_rate = jack_get_sample_rate; + ma_jack_get_buffer_size_proc _jack_get_buffer_size = jack_get_buffer_size; + ma_jack_get_ports_proc _jack_get_ports = jack_get_ports; + ma_jack_activate_proc _jack_activate = jack_activate; + ma_jack_deactivate_proc _jack_deactivate = jack_deactivate; + ma_jack_connect_proc _jack_connect = jack_connect; + ma_jack_port_register_proc _jack_port_register = jack_port_register; + ma_jack_port_name_proc _jack_port_name = jack_port_name; + ma_jack_port_get_buffer_proc _jack_port_get_buffer = jack_port_get_buffer; + ma_jack_free_proc _jack_free = jack_free; + + pContext->jack.jack_client_open = (ma_proc)_jack_client_open; + pContext->jack.jack_client_close = (ma_proc)_jack_client_close; + pContext->jack.jack_client_name_size = (ma_proc)_jack_client_name_size; + pContext->jack.jack_set_process_callback = (ma_proc)_jack_set_process_callback; + pContext->jack.jack_set_buffer_size_callback = (ma_proc)_jack_set_buffer_size_callback; + pContext->jack.jack_on_shutdown = (ma_proc)_jack_on_shutdown; + pContext->jack.jack_get_sample_rate = (ma_proc)_jack_get_sample_rate; + pContext->jack.jack_get_buffer_size = (ma_proc)_jack_get_buffer_size; + pContext->jack.jack_get_ports = (ma_proc)_jack_get_ports; + pContext->jack.jack_activate = (ma_proc)_jack_activate; + pContext->jack.jack_deactivate = (ma_proc)_jack_deactivate; + pContext->jack.jack_connect = (ma_proc)_jack_connect; + pContext->jack.jack_port_register = (ma_proc)_jack_port_register; + pContext->jack.jack_port_name = (ma_proc)_jack_port_name; + pContext->jack.jack_port_get_buffer = (ma_proc)_jack_port_get_buffer; + pContext->jack.jack_free = (ma_proc)_jack_free; +#endif + + pContext->isBackendAsynchronous = MA_TRUE; -ma_result ma_find_best_format__coreaudio(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_bool32 usingDefaultFormat, ma_bool32 usingDefaultChannels, ma_bool32 usingDefaultSampleRate, AudioStreamBasicDescription* pFormat) -{ - UInt32 deviceFormatDescriptionCount; - AudioStreamRangedDescription* pDeviceFormatDescriptions; - ma_result result; - ma_uint32 desiredSampleRate; - ma_uint32 desiredChannelCount; - ma_format desiredFormat; - AudioStreamBasicDescription bestDeviceFormatSoFar; - ma_bool32 hasSupportedFormat; - UInt32 iFormat; + pContext->onUninit = ma_context_uninit__jack; + pContext->onDeviceIDEqual = ma_context_is_device_id_equal__jack; + pContext->onEnumDevices = ma_context_enumerate_devices__jack; + pContext->onGetDeviceInfo = ma_context_get_device_info__jack; + pContext->onDeviceInit = ma_device_init__jack; + pContext->onDeviceUninit = ma_device_uninit__jack; + pContext->onDeviceStart = ma_device_start__jack; + pContext->onDeviceStop = ma_device_stop__jack; - result = ma_get_AudioObject_stream_descriptions(pContext, deviceObjectID, deviceType, &deviceFormatDescriptionCount, &pDeviceFormatDescriptions); - if (result != MA_SUCCESS) { - return result; + if (pConfig->jack.pClientName != NULL) { + pContext->jack.pClientName = ma_copy_string(pConfig->jack.pClientName, &pContext->allocationCallbacks); } - - desiredSampleRate = sampleRate; - if (usingDefaultSampleRate) { - /* - When using the device's default sample rate, we get the highest priority standard rate supported by the device. Otherwise - we just use the pre-set rate. - */ - ma_uint32 iStandardRate; - for (iStandardRate = 0; iStandardRate < ma_countof(g_maStandardSampleRatePriorities); ++iStandardRate) { - ma_uint32 standardRate = g_maStandardSampleRatePriorities[iStandardRate]; - ma_bool32 foundRate = MA_FALSE; - UInt32 iDeviceRate; + pContext->jack.tryStartServer = pConfig->jack.tryStartServer; - for (iDeviceRate = 0; iDeviceRate < deviceFormatDescriptionCount; ++iDeviceRate) { - ma_uint32 deviceRate = (ma_uint32)pDeviceFormatDescriptions[iDeviceRate].mFormat.mSampleRate; - - if (deviceRate == standardRate) { - desiredSampleRate = standardRate; - foundRate = MA_TRUE; - break; - } - } - - if (foundRate) { - break; - } - } - } - - desiredChannelCount = channels; - if (usingDefaultChannels) { - ma_get_AudioObject_channel_count(pContext, deviceObjectID, deviceType, &desiredChannelCount); /* <-- Not critical if this fails. */ - } - - desiredFormat = format; - if (usingDefaultFormat) { - desiredFormat = g_maFormatPriorities[0]; - } - /* - If we get here it means we don't have an exact match to what the client is asking for. We'll need to find the closest one. The next - loop will check for formats that have the same sample rate to what we're asking for. If there is, we prefer that one in all cases. + Getting here means the JACK library is installed, but it doesn't necessarily mean it's usable. We need to quickly test this by connecting + a temporary client. */ - ma_zero_object(&bestDeviceFormatSoFar); - - hasSupportedFormat = MA_FALSE; - for (iFormat = 0; iFormat < deviceFormatDescriptionCount; ++iFormat) { - ma_format format; - ma_result formatResult = ma_format_from_AudioStreamBasicDescription(&pDeviceFormatDescriptions[iFormat].mFormat, &format); - if (formatResult == MA_SUCCESS && format != ma_format_unknown) { - hasSupportedFormat = MA_TRUE; - bestDeviceFormatSoFar = pDeviceFormatDescriptions[iFormat].mFormat; - break; + { + ma_jack_client_t* pDummyClient; + ma_result result = ma_context_open_client__jack(pContext, &pDummyClient); + if (result != MA_SUCCESS) { + ma_free(pContext->jack.pClientName, &pContext->allocationCallbacks); + #ifndef MA_NO_RUNTIME_LINKING + ma_dlclose(pContext, pContext->jack.jackSO); + #endif + return MA_NO_BACKEND; } - } - - if (!hasSupportedFormat) { - ma_free(pDeviceFormatDescriptions); - return MA_FORMAT_NOT_SUPPORTED; - } - - - for (iFormat = 0; iFormat < deviceFormatDescriptionCount; ++iFormat) { - AudioStreamBasicDescription thisDeviceFormat = pDeviceFormatDescriptions[iFormat].mFormat; - ma_format thisSampleFormat; - ma_result formatResult; - ma_format bestSampleFormatSoFar; - /* If the format is not supported by miniaudio we need to skip this one entirely. */ - formatResult = ma_format_from_AudioStreamBasicDescription(&pDeviceFormatDescriptions[iFormat].mFormat, &thisSampleFormat); - if (formatResult != MA_SUCCESS || thisSampleFormat == ma_format_unknown) { - continue; /* The format is not supported by miniaudio. Skip. */ - } - - ma_format_from_AudioStreamBasicDescription(&bestDeviceFormatSoFar, &bestSampleFormatSoFar); - - /* Getting here means the format is supported by miniaudio which makes this format a candidate. */ - if (thisDeviceFormat.mSampleRate != desiredSampleRate) { - /* - The sample rate does not match, but this format could still be usable, although it's a very low priority. If the best format - so far has an equal sample rate we can just ignore this one. - */ - if (bestDeviceFormatSoFar.mSampleRate == desiredSampleRate) { - continue; /* The best sample rate so far has the same sample rate as what we requested which means it's still the best so far. Skip this format. */ - } else { - /* In this case, neither the best format so far nor this one have the same sample rate. Check the channel count next. */ - if (thisDeviceFormat.mChannelsPerFrame != desiredChannelCount) { - /* This format has a different sample rate _and_ a different channel count. */ - if (bestDeviceFormatSoFar.mChannelsPerFrame == desiredChannelCount) { - continue; /* No change to the best format. */ - } else { - /* - Both this format and the best so far have different sample rates and different channel counts. Whichever has the - best format is the new best. - */ - if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) { - bestDeviceFormatSoFar = thisDeviceFormat; - continue; - } else { - continue; /* No change to the best format. */ - } - } - } else { - /* This format has a different sample rate but the desired channel count. */ - if (bestDeviceFormatSoFar.mChannelsPerFrame == desiredChannelCount) { - /* Both this format and the best so far have the desired channel count. Whichever has the best format is the new best. */ - if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) { - bestDeviceFormatSoFar = thisDeviceFormat; - continue; - } else { - continue; /* No change to the best format for now. */ - } - } else { - /* This format has the desired channel count, but the best so far does not. We have a new best. */ - bestDeviceFormatSoFar = thisDeviceFormat; - continue; - } - } - } - } else { - /* - The sample rates match which makes this format a very high priority contender. If the best format so far has a different - sample rate it needs to be replaced with this one. - */ - if (bestDeviceFormatSoFar.mSampleRate != desiredSampleRate) { - bestDeviceFormatSoFar = thisDeviceFormat; - continue; - } else { - /* In this case both this format and the best format so far have the same sample rate. Check the channel count next. */ - if (thisDeviceFormat.mChannelsPerFrame == desiredChannelCount) { - /* - In this case this format has the same channel count as what the client is requesting. If the best format so far has - a different count, this one becomes the new best. - */ - if (bestDeviceFormatSoFar.mChannelsPerFrame != desiredChannelCount) { - bestDeviceFormatSoFar = thisDeviceFormat; - continue; - } else { - /* In this case both this format and the best so far have the ideal sample rate and channel count. Check the format. */ - if (thisSampleFormat == desiredFormat) { - bestDeviceFormatSoFar = thisDeviceFormat; - break; /* Found the exact match. */ - } else { - /* The formats are different. The new best format is the one with the highest priority format according to miniaudio. */ - if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) { - bestDeviceFormatSoFar = thisDeviceFormat; - continue; - } else { - continue; /* No change to the best format for now. */ - } - } - } - } else { - /* - In this case the channel count is different to what the client has requested. If the best so far has the same channel - count as the requested count then it remains the best. - */ - if (bestDeviceFormatSoFar.mChannelsPerFrame == desiredChannelCount) { - continue; - } else { - /* - This is the case where both have the same sample rate (good) but different channel counts. Right now both have about - the same priority, but we need to compare the format now. - */ - if (thisSampleFormat == bestSampleFormatSoFar) { - if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) { - bestDeviceFormatSoFar = thisDeviceFormat; - continue; - } else { - continue; /* No change to the best format for now. */ - } - } - } - } - } - } + ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pDummyClient); } - - *pFormat = bestDeviceFormatSoFar; - ma_free(pDeviceFormatDescriptions); return MA_SUCCESS; } +#endif /* JACK */ + + + +/****************************************************************************** + +Core Audio Backend + +******************************************************************************/ +#ifdef MA_HAS_COREAUDIO +#include + +#if defined(TARGET_OS_IPHONE) && TARGET_OS_IPHONE == 1 + #define MA_APPLE_MOBILE + #if defined(TARGET_OS_TV) && TARGET_OS_TV == 1 + #define MA_APPLE_TV + #endif + #if defined(TARGET_OS_WATCH) && TARGET_OS_WATCH == 1 + #define MA_APPLE_WATCH + #endif +#else + #define MA_APPLE_DESKTOP +#endif + +#if defined(MA_APPLE_DESKTOP) +#include +#else +#include +#endif + +#include + +/* CoreFoundation */ +typedef Boolean (* ma_CFStringGetCString_proc)(CFStringRef theString, char* buffer, CFIndex bufferSize, CFStringEncoding encoding); +typedef void (* ma_CFRelease_proc)(CFTypeRef cf); + +/* CoreAudio */ +#if defined(MA_APPLE_DESKTOP) +typedef OSStatus (* ma_AudioObjectGetPropertyData_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, UInt32 inQualifierDataSize, const void* inQualifierData, UInt32* ioDataSize, void* outData); +typedef OSStatus (* ma_AudioObjectGetPropertyDataSize_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, UInt32 inQualifierDataSize, const void* inQualifierData, UInt32* outDataSize); +typedef OSStatus (* ma_AudioObjectSetPropertyData_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, UInt32 inQualifierDataSize, const void* inQualifierData, UInt32 inDataSize, const void* inData); +typedef OSStatus (* ma_AudioObjectAddPropertyListener_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, AudioObjectPropertyListenerProc inListener, void* inClientData); +typedef OSStatus (* ma_AudioObjectRemovePropertyListener_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, AudioObjectPropertyListenerProc inListener, void* inClientData); #endif -ma_result ma_get_AudioUnit_channel_map(ma_context* pContext, AudioUnit audioUnit, ma_device_type deviceType, ma_channel channelMap[MA_MAX_CHANNELS]) -{ - AudioUnitScope deviceScope; - AudioUnitElement deviceBus; - UInt32 channelLayoutSize; - OSStatus status; - AudioChannelLayout* pChannelLayout; - ma_result result; +/* AudioToolbox */ +typedef AudioComponent (* ma_AudioComponentFindNext_proc)(AudioComponent inComponent, const AudioComponentDescription* inDesc); +typedef OSStatus (* ma_AudioComponentInstanceDispose_proc)(AudioComponentInstance inInstance); +typedef OSStatus (* ma_AudioComponentInstanceNew_proc)(AudioComponent inComponent, AudioComponentInstance* outInstance); +typedef OSStatus (* ma_AudioOutputUnitStart_proc)(AudioUnit inUnit); +typedef OSStatus (* ma_AudioOutputUnitStop_proc)(AudioUnit inUnit); +typedef OSStatus (* ma_AudioUnitAddPropertyListener_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitPropertyListenerProc inProc, void* inProcUserData); +typedef OSStatus (* ma_AudioUnitGetPropertyInfo_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, UInt32* outDataSize, Boolean* outWriteable); +typedef OSStatus (* ma_AudioUnitGetProperty_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, void* outData, UInt32* ioDataSize); +typedef OSStatus (* ma_AudioUnitSetProperty_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, const void* inData, UInt32 inDataSize); +typedef OSStatus (* ma_AudioUnitInitialize_proc)(AudioUnit inUnit); +typedef OSStatus (* ma_AudioUnitRender_proc)(AudioUnit inUnit, AudioUnitRenderActionFlags* ioActionFlags, const AudioTimeStamp* inTimeStamp, UInt32 inOutputBusNumber, UInt32 inNumberFrames, AudioBufferList* ioData); - ma_assert(pContext != NULL); - - if (deviceType == ma_device_type_playback) { - deviceScope = kAudioUnitScope_Output; - deviceBus = MA_COREAUDIO_OUTPUT_BUS; - } else { - deviceScope = kAudioUnitScope_Input; - deviceBus = MA_COREAUDIO_INPUT_BUS; - } - - status = ((ma_AudioUnitGetPropertyInfo_proc)pContext->coreaudio.AudioUnitGetPropertyInfo)(audioUnit, kAudioUnitProperty_AudioChannelLayout, deviceScope, deviceBus, &channelLayoutSize, NULL); - if (status != noErr) { - return ma_result_from_OSStatus(status); - } - - pChannelLayout = (AudioChannelLayout*)ma_malloc(channelLayoutSize); - if (pChannelLayout == NULL) { - return MA_OUT_OF_MEMORY; - } - - status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(audioUnit, kAudioUnitProperty_AudioChannelLayout, deviceScope, deviceBus, pChannelLayout, &channelLayoutSize); - if (status != noErr) { - ma_free(pChannelLayout); - return ma_result_from_OSStatus(status); - } - - result = ma_get_channel_map_from_AudioChannelLayout(pChannelLayout, channelMap); - if (result != MA_SUCCESS) { - ma_free(pChannelLayout); - return result; - } - ma_free(pChannelLayout); - return MA_SUCCESS; -} +#define MA_COREAUDIO_OUTPUT_BUS 0 +#define MA_COREAUDIO_INPUT_BUS 1 -ma_bool32 ma_context_is_device_id_equal__coreaudio(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) -{ - ma_assert(pContext != NULL); - ma_assert(pID0 != NULL); - ma_assert(pID1 != NULL); - (void)pContext; +static ma_result ma_device_reinit_internal__coreaudio(ma_device* pDevice, ma_device_type deviceType, ma_bool32 disposePreviousAudioUnit); - return strcmp(pID0->coreaudio, pID1->coreaudio) == 0; -} +/* +Core Audio -ma_result ma_context_enumerate_devices__coreaudio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) -{ -#if defined(MA_APPLE_DESKTOP) - UInt32 deviceCount; - AudioObjectID* pDeviceObjectIDs; - ma_result result; - UInt32 iDevice; +So far, Core Audio has been the worst backend to work with due to being both unintuitive and having almost no documentation +apart from comments in the headers (which admittedly are quite good). For my own purposes, and for anybody out there whose +needing to figure out how this darn thing works, I'm going to outline a few things here. - result = ma_get_device_object_ids__coreaudio(pContext, &deviceCount, &pDeviceObjectIDs); - if (result != MA_SUCCESS) { - return result; +Since miniaudio is a fairly low-level API, one of the things it needs is control over specific devices, and it needs to be +able to identify whether or not it can be used as playback and/or capture. The AudioObject API is the only one I've seen +that supports this level of detail. There was some public domain sample code I stumbled across that used the AudioComponent +and AudioUnit APIs, but I couldn't see anything that gave low-level control over device selection and capabilities (the +distinction between playback and capture in particular). Therefore, miniaudio is using the AudioObject API. + +Most (all?) functions in the AudioObject API take a AudioObjectID as it's input. This is the device identifier. When +retrieving global information, such as the device list, you use kAudioObjectSystemObject. When retrieving device-specific +data, you pass in the ID for that device. In order to retrieve device-specific IDs you need to enumerate over each of the +devices. This is done using the AudioObjectGetPropertyDataSize() and AudioObjectGetPropertyData() APIs which seem to be +the central APIs for retrieving information about the system and specific devices. + +To use the AudioObjectGetPropertyData() API you need to use the notion of a property address. A property address is a +structure with three variables and is used to identify which property you are getting or setting. The first is the "selector" +which is basically the specific property that you're wanting to retrieve or set. The second is the "scope", which is +typically set to kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyScopeInput for input-specific properties and +kAudioObjectPropertyScopeOutput for output-specific properties. The last is the "element" which is always set to +kAudioObjectPropertyElementMaster in miniaudio's case. I don't know of any cases where this would be set to anything different. + +Back to the earlier issue of device retrieval, you first use the AudioObjectGetPropertyDataSize() API to retrieve the size +of the raw data which is just a list of AudioDeviceID's. You use the kAudioObjectSystemObject AudioObjectID, and a property +address with the kAudioHardwarePropertyDevices selector and the kAudioObjectPropertyScopeGlobal scope. Once you have the +size, allocate a block of memory of that size and then call AudioObjectGetPropertyData(). The data is just a list of +AudioDeviceID's so just do "dataSize/sizeof(AudioDeviceID)" to know the device count. +*/ + +static ma_result ma_result_from_OSStatus(OSStatus status) +{ + switch (status) + { + case noErr: return MA_SUCCESS; + #if defined(MA_APPLE_DESKTOP) + case kAudioHardwareNotRunningError: return MA_DEVICE_NOT_STARTED; + case kAudioHardwareUnspecifiedError: return MA_ERROR; + case kAudioHardwareUnknownPropertyError: return MA_INVALID_ARGS; + case kAudioHardwareBadPropertySizeError: return MA_INVALID_OPERATION; + case kAudioHardwareIllegalOperationError: return MA_INVALID_OPERATION; + case kAudioHardwareBadObjectError: return MA_INVALID_ARGS; + case kAudioHardwareBadDeviceError: return MA_INVALID_ARGS; + case kAudioHardwareBadStreamError: return MA_INVALID_ARGS; + case kAudioHardwareUnsupportedOperationError: return MA_INVALID_OPERATION; + case kAudioDeviceUnsupportedFormatError: return MA_FORMAT_NOT_SUPPORTED; + case kAudioDevicePermissionsError: return MA_ACCESS_DENIED; + #endif + default: return MA_ERROR; } - - for (iDevice = 0; iDevice < deviceCount; ++iDevice) { - AudioObjectID deviceObjectID = pDeviceObjectIDs[iDevice]; - ma_device_info info; +} - ma_zero_object(&info); - if (ma_get_AudioObject_uid(pContext, deviceObjectID, sizeof(info.id.coreaudio), info.id.coreaudio) != MA_SUCCESS) { - continue; - } - if (ma_get_AudioObject_name(pContext, deviceObjectID, sizeof(info.name), info.name) != MA_SUCCESS) { - continue; - } +#if 0 +static ma_channel ma_channel_from_AudioChannelBitmap(AudioChannelBitmap bit) +{ + switch (bit) + { + case kAudioChannelBit_Left: return MA_CHANNEL_LEFT; + case kAudioChannelBit_Right: return MA_CHANNEL_RIGHT; + case kAudioChannelBit_Center: return MA_CHANNEL_FRONT_CENTER; + case kAudioChannelBit_LFEScreen: return MA_CHANNEL_LFE; + case kAudioChannelBit_LeftSurround: return MA_CHANNEL_BACK_LEFT; + case kAudioChannelBit_RightSurround: return MA_CHANNEL_BACK_RIGHT; + case kAudioChannelBit_LeftCenter: return MA_CHANNEL_FRONT_LEFT_CENTER; + case kAudioChannelBit_RightCenter: return MA_CHANNEL_FRONT_RIGHT_CENTER; + case kAudioChannelBit_CenterSurround: return MA_CHANNEL_BACK_CENTER; + case kAudioChannelBit_LeftSurroundDirect: return MA_CHANNEL_SIDE_LEFT; + case kAudioChannelBit_RightSurroundDirect: return MA_CHANNEL_SIDE_RIGHT; + case kAudioChannelBit_TopCenterSurround: return MA_CHANNEL_TOP_CENTER; + case kAudioChannelBit_VerticalHeightLeft: return MA_CHANNEL_TOP_FRONT_LEFT; + case kAudioChannelBit_VerticalHeightCenter: return MA_CHANNEL_TOP_FRONT_CENTER; + case kAudioChannelBit_VerticalHeightRight: return MA_CHANNEL_TOP_FRONT_RIGHT; + case kAudioChannelBit_TopBackLeft: return MA_CHANNEL_TOP_BACK_LEFT; + case kAudioChannelBit_TopBackCenter: return MA_CHANNEL_TOP_BACK_CENTER; + case kAudioChannelBit_TopBackRight: return MA_CHANNEL_TOP_BACK_RIGHT; + default: return MA_CHANNEL_NONE; + } +} +#endif - if (ma_does_AudioObject_support_playback(pContext, deviceObjectID)) { - if (!callback(pContext, ma_device_type_playback, &info, pUserData)) { - break; - } - } - if (ma_does_AudioObject_support_capture(pContext, deviceObjectID)) { - if (!callback(pContext, ma_device_type_capture, &info, pUserData)) { - break; - } - } +static ma_channel ma_channel_from_AudioChannelLabel(AudioChannelLabel label) +{ + switch (label) + { + case kAudioChannelLabel_Unknown: return MA_CHANNEL_NONE; + case kAudioChannelLabel_Unused: return MA_CHANNEL_NONE; + case kAudioChannelLabel_UseCoordinates: return MA_CHANNEL_NONE; + case kAudioChannelLabel_Left: return MA_CHANNEL_LEFT; + case kAudioChannelLabel_Right: return MA_CHANNEL_RIGHT; + case kAudioChannelLabel_Center: return MA_CHANNEL_FRONT_CENTER; + case kAudioChannelLabel_LFEScreen: return MA_CHANNEL_LFE; + case kAudioChannelLabel_LeftSurround: return MA_CHANNEL_BACK_LEFT; + case kAudioChannelLabel_RightSurround: return MA_CHANNEL_BACK_RIGHT; + case kAudioChannelLabel_LeftCenter: return MA_CHANNEL_FRONT_LEFT_CENTER; + case kAudioChannelLabel_RightCenter: return MA_CHANNEL_FRONT_RIGHT_CENTER; + case kAudioChannelLabel_CenterSurround: return MA_CHANNEL_BACK_CENTER; + case kAudioChannelLabel_LeftSurroundDirect: return MA_CHANNEL_SIDE_LEFT; + case kAudioChannelLabel_RightSurroundDirect: return MA_CHANNEL_SIDE_RIGHT; + case kAudioChannelLabel_TopCenterSurround: return MA_CHANNEL_TOP_CENTER; + case kAudioChannelLabel_VerticalHeightLeft: return MA_CHANNEL_TOP_FRONT_LEFT; + case kAudioChannelLabel_VerticalHeightCenter: return MA_CHANNEL_TOP_FRONT_CENTER; + case kAudioChannelLabel_VerticalHeightRight: return MA_CHANNEL_TOP_FRONT_RIGHT; + case kAudioChannelLabel_TopBackLeft: return MA_CHANNEL_TOP_BACK_LEFT; + case kAudioChannelLabel_TopBackCenter: return MA_CHANNEL_TOP_BACK_CENTER; + case kAudioChannelLabel_TopBackRight: return MA_CHANNEL_TOP_BACK_RIGHT; + case kAudioChannelLabel_RearSurroundLeft: return MA_CHANNEL_BACK_LEFT; + case kAudioChannelLabel_RearSurroundRight: return MA_CHANNEL_BACK_RIGHT; + case kAudioChannelLabel_LeftWide: return MA_CHANNEL_SIDE_LEFT; + case kAudioChannelLabel_RightWide: return MA_CHANNEL_SIDE_RIGHT; + case kAudioChannelLabel_LFE2: return MA_CHANNEL_LFE; + case kAudioChannelLabel_LeftTotal: return MA_CHANNEL_LEFT; + case kAudioChannelLabel_RightTotal: return MA_CHANNEL_RIGHT; + case kAudioChannelLabel_HearingImpaired: return MA_CHANNEL_NONE; + case kAudioChannelLabel_Narration: return MA_CHANNEL_MONO; + case kAudioChannelLabel_Mono: return MA_CHANNEL_MONO; + case kAudioChannelLabel_DialogCentricMix: return MA_CHANNEL_MONO; + case kAudioChannelLabel_CenterSurroundDirect: return MA_CHANNEL_BACK_CENTER; + case kAudioChannelLabel_Haptic: return MA_CHANNEL_NONE; + case kAudioChannelLabel_Ambisonic_W: return MA_CHANNEL_NONE; + case kAudioChannelLabel_Ambisonic_X: return MA_CHANNEL_NONE; + case kAudioChannelLabel_Ambisonic_Y: return MA_CHANNEL_NONE; + case kAudioChannelLabel_Ambisonic_Z: return MA_CHANNEL_NONE; + case kAudioChannelLabel_MS_Mid: return MA_CHANNEL_LEFT; + case kAudioChannelLabel_MS_Side: return MA_CHANNEL_RIGHT; + case kAudioChannelLabel_XY_X: return MA_CHANNEL_LEFT; + case kAudioChannelLabel_XY_Y: return MA_CHANNEL_RIGHT; + case kAudioChannelLabel_HeadphonesLeft: return MA_CHANNEL_LEFT; + case kAudioChannelLabel_HeadphonesRight: return MA_CHANNEL_RIGHT; + case kAudioChannelLabel_ClickTrack: return MA_CHANNEL_NONE; + case kAudioChannelLabel_ForeignLanguage: return MA_CHANNEL_NONE; + case kAudioChannelLabel_Discrete: return MA_CHANNEL_NONE; + case kAudioChannelLabel_Discrete_0: return MA_CHANNEL_AUX_0; + case kAudioChannelLabel_Discrete_1: return MA_CHANNEL_AUX_1; + case kAudioChannelLabel_Discrete_2: return MA_CHANNEL_AUX_2; + case kAudioChannelLabel_Discrete_3: return MA_CHANNEL_AUX_3; + case kAudioChannelLabel_Discrete_4: return MA_CHANNEL_AUX_4; + case kAudioChannelLabel_Discrete_5: return MA_CHANNEL_AUX_5; + case kAudioChannelLabel_Discrete_6: return MA_CHANNEL_AUX_6; + case kAudioChannelLabel_Discrete_7: return MA_CHANNEL_AUX_7; + case kAudioChannelLabel_Discrete_8: return MA_CHANNEL_AUX_8; + case kAudioChannelLabel_Discrete_9: return MA_CHANNEL_AUX_9; + case kAudioChannelLabel_Discrete_10: return MA_CHANNEL_AUX_10; + case kAudioChannelLabel_Discrete_11: return MA_CHANNEL_AUX_11; + case kAudioChannelLabel_Discrete_12: return MA_CHANNEL_AUX_12; + case kAudioChannelLabel_Discrete_13: return MA_CHANNEL_AUX_13; + case kAudioChannelLabel_Discrete_14: return MA_CHANNEL_AUX_14; + case kAudioChannelLabel_Discrete_15: return MA_CHANNEL_AUX_15; + case kAudioChannelLabel_Discrete_65535: return MA_CHANNEL_NONE; + + #if 0 /* Introduced in a later version of macOS. */ + case kAudioChannelLabel_HOA_ACN: return MA_CHANNEL_NONE; + case kAudioChannelLabel_HOA_ACN_0: return MA_CHANNEL_AUX_0; + case kAudioChannelLabel_HOA_ACN_1: return MA_CHANNEL_AUX_1; + case kAudioChannelLabel_HOA_ACN_2: return MA_CHANNEL_AUX_2; + case kAudioChannelLabel_HOA_ACN_3: return MA_CHANNEL_AUX_3; + case kAudioChannelLabel_HOA_ACN_4: return MA_CHANNEL_AUX_4; + case kAudioChannelLabel_HOA_ACN_5: return MA_CHANNEL_AUX_5; + case kAudioChannelLabel_HOA_ACN_6: return MA_CHANNEL_AUX_6; + case kAudioChannelLabel_HOA_ACN_7: return MA_CHANNEL_AUX_7; + case kAudioChannelLabel_HOA_ACN_8: return MA_CHANNEL_AUX_8; + case kAudioChannelLabel_HOA_ACN_9: return MA_CHANNEL_AUX_9; + case kAudioChannelLabel_HOA_ACN_10: return MA_CHANNEL_AUX_10; + case kAudioChannelLabel_HOA_ACN_11: return MA_CHANNEL_AUX_11; + case kAudioChannelLabel_HOA_ACN_12: return MA_CHANNEL_AUX_12; + case kAudioChannelLabel_HOA_ACN_13: return MA_CHANNEL_AUX_13; + case kAudioChannelLabel_HOA_ACN_14: return MA_CHANNEL_AUX_14; + case kAudioChannelLabel_HOA_ACN_15: return MA_CHANNEL_AUX_15; + case kAudioChannelLabel_HOA_ACN_65024: return MA_CHANNEL_NONE; + #endif + + default: return MA_CHANNEL_NONE; } +} + +static ma_result ma_format_from_AudioStreamBasicDescription(const AudioStreamBasicDescription* pDescription, ma_format* pFormatOut) +{ + MA_ASSERT(pDescription != NULL); + MA_ASSERT(pFormatOut != NULL); - ma_free(pDeviceObjectIDs); -#else - /* Only supporting default devices on non-Desktop platforms. */ - ma_device_info info; + *pFormatOut = ma_format_unknown; /* Safety. */ - ma_zero_object(&info); - ma_strncpy_s(info.name, sizeof(info.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); - if (!callback(pContext, ma_device_type_playback, &info, pUserData)) { - return MA_SUCCESS; + /* There's a few things miniaudio doesn't support. */ + if (pDescription->mFormatID != kAudioFormatLinearPCM) { + return MA_FORMAT_NOT_SUPPORTED; } - ma_zero_object(&info); - ma_strncpy_s(info.name, sizeof(info.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); - if (!callback(pContext, ma_device_type_capture, &info, pUserData)) { - return MA_SUCCESS; + /* We don't support any non-packed formats that are aligned high. */ + if ((pDescription->mFormatFlags & kLinearPCMFormatFlagIsAlignedHigh) != 0) { + return MA_FORMAT_NOT_SUPPORTED; } -#endif - - return MA_SUCCESS; -} - -ma_result ma_context_get_device_info__coreaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) -{ - ma_result result; - - ma_assert(pContext != NULL); - /* No exclusive mode with the Core Audio backend for now. */ - if (shareMode == ma_share_mode_exclusive) { - return MA_SHARE_MODE_NOT_SUPPORTED; + /* Only supporting native-endian. */ + if ((ma_is_little_endian() && (pDescription->mFormatFlags & kAudioFormatFlagIsBigEndian) != 0) || (ma_is_big_endian() && (pDescription->mFormatFlags & kAudioFormatFlagIsBigEndian) == 0)) { + return MA_FORMAT_NOT_SUPPORTED; } -#if defined(MA_APPLE_DESKTOP) - /* Desktop */ - { - AudioObjectID deviceObjectID; - UInt32 streamDescriptionCount; - AudioStreamRangedDescription* pStreamDescriptions; - UInt32 iStreamDescription; - UInt32 sampleRateRangeCount; - AudioValueRange* pSampleRateRanges; + /* We are not currently supporting non-interleaved formats (this will be added in a future version of miniaudio). */ + /*if ((pDescription->mFormatFlags & kAudioFormatFlagIsNonInterleaved) != 0) { + return MA_FORMAT_NOT_SUPPORTED; + }*/ - result = ma_find_AudioObjectID(pContext, deviceType, pDeviceID, &deviceObjectID); - if (result != MA_SUCCESS) { - return result; - } - - result = ma_get_AudioObject_uid(pContext, deviceObjectID, sizeof(pDeviceInfo->id.coreaudio), pDeviceInfo->id.coreaudio); - if (result != MA_SUCCESS) { - return result; - } - - result = ma_get_AudioObject_name(pContext, deviceObjectID, sizeof(pDeviceInfo->name), pDeviceInfo->name); - if (result != MA_SUCCESS) { - return result; - } - - /* Formats. */ - result = ma_get_AudioObject_stream_descriptions(pContext, deviceObjectID, deviceType, &streamDescriptionCount, &pStreamDescriptions); - if (result != MA_SUCCESS) { - return result; + if ((pDescription->mFormatFlags & kLinearPCMFormatFlagIsFloat) != 0) { + if (pDescription->mBitsPerChannel == 32) { + *pFormatOut = ma_format_f32; + return MA_SUCCESS; } - - for (iStreamDescription = 0; iStreamDescription < streamDescriptionCount; ++iStreamDescription) { - ma_format format; - ma_bool32 formatExists = MA_FALSE; - ma_uint32 iOutputFormat; - - result = ma_format_from_AudioStreamBasicDescription(&pStreamDescriptions[iStreamDescription].mFormat, &format); - if (result != MA_SUCCESS) { - continue; - } - - ma_assert(format != ma_format_unknown); - - /* Make sure the format isn't already in the output list. */ - for (iOutputFormat = 0; iOutputFormat < pDeviceInfo->formatCount; ++iOutputFormat) { - if (pDeviceInfo->formats[iOutputFormat] == format) { - formatExists = MA_TRUE; - break; + } else { + if ((pDescription->mFormatFlags & kLinearPCMFormatFlagIsSignedInteger) != 0) { + if (pDescription->mBitsPerChannel == 16) { + *pFormatOut = ma_format_s16; + return MA_SUCCESS; + } else if (pDescription->mBitsPerChannel == 24) { + if (pDescription->mBytesPerFrame == (pDescription->mBitsPerChannel/8 * pDescription->mChannelsPerFrame)) { + *pFormatOut = ma_format_s24; + return MA_SUCCESS; + } else { + if (pDescription->mBytesPerFrame/pDescription->mChannelsPerFrame == sizeof(ma_int32)) { + /* TODO: Implement ma_format_s24_32. */ + /**pFormatOut = ma_format_s24_32;*/ + /*return MA_SUCCESS;*/ + return MA_FORMAT_NOT_SUPPORTED; + } } + } else if (pDescription->mBitsPerChannel == 32) { + *pFormatOut = ma_format_s32; + return MA_SUCCESS; } - - if (!formatExists) { - pDeviceInfo->formats[pDeviceInfo->formatCount++] = format; + } else { + if (pDescription->mBitsPerChannel == 8) { + *pFormatOut = ma_format_u8; + return MA_SUCCESS; } } + } - ma_free(pStreamDescriptions); - - - /* Channels. */ - result = ma_get_AudioObject_channel_count(pContext, deviceObjectID, deviceType, &pDeviceInfo->minChannels); - if (result != MA_SUCCESS) { - return result; - } - pDeviceInfo->maxChannels = pDeviceInfo->minChannels; - + /* Getting here means the format is not supported. */ + return MA_FORMAT_NOT_SUPPORTED; +} + +static ma_result ma_get_channel_map_from_AudioChannelLayout(AudioChannelLayout* pChannelLayout, ma_channel channelMap[MA_MAX_CHANNELS]) +{ + MA_ASSERT(pChannelLayout != NULL); - /* Sample rates. */ - result = ma_get_AudioObject_sample_rates(pContext, deviceObjectID, deviceType, &sampleRateRangeCount, &pSampleRateRanges); - if (result != MA_SUCCESS) { - return result; + if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelDescriptions) { + UInt32 iChannel; + for (iChannel = 0; iChannel < pChannelLayout->mNumberChannelDescriptions; ++iChannel) { + channelMap[iChannel] = ma_channel_from_AudioChannelLabel(pChannelLayout->mChannelDescriptions[iChannel].mChannelLabel); } - - if (sampleRateRangeCount > 0) { - UInt32 iSampleRate; - pDeviceInfo->minSampleRate = UINT32_MAX; - pDeviceInfo->maxSampleRate = 0; - for (iSampleRate = 0; iSampleRate < sampleRateRangeCount; ++iSampleRate) { - if (pDeviceInfo->minSampleRate > pSampleRateRanges[iSampleRate].mMinimum) { - pDeviceInfo->minSampleRate = pSampleRateRanges[iSampleRate].mMinimum; - } - if (pDeviceInfo->maxSampleRate < pSampleRateRanges[iSampleRate].mMaximum) { - pDeviceInfo->maxSampleRate = pSampleRateRanges[iSampleRate].mMaximum; - } + } else +#if 0 + if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelBitmap) { + /* This is the same kind of system that's used by Windows audio APIs. */ + UInt32 iChannel = 0; + UInt32 iBit; + AudioChannelBitmap bitmap = pChannelLayout->mChannelBitmap; + for (iBit = 0; iBit < 32; ++iBit) { + AudioChannelBitmap bit = bitmap & (1 << iBit); + if (bit != 0) { + channelMap[iChannel++] = ma_channel_from_AudioChannelBit(bit); } } - } -#else - /* Mobile */ + } else +#endif { - AudioComponentDescription desc; - AudioComponent component; - AudioUnit audioUnit; - OSStatus status; - AudioUnitScope formatScope; - AudioUnitElement formatElement; - AudioStreamBasicDescription bestFormat; - UInt32 propSize; - - if (deviceType == ma_device_type_playback) { - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); - } else { - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); - } - - /* - Retrieving device information is more annoying on mobile than desktop. For simplicity I'm locking this down to whatever format is - reported on a temporary I/O unit. The problem, however, is that this doesn't return a value for the sample rate which we need to - retrieve from the AVAudioSession shared instance. - */ - desc.componentType = kAudioUnitType_Output; - desc.componentSubType = kAudioUnitSubType_RemoteIO; - desc.componentManufacturer = kAudioUnitManufacturer_Apple; - desc.componentFlags = 0; - desc.componentFlagsMask = 0; - - component = ((ma_AudioComponentFindNext_proc)pContext->coreaudio.AudioComponentFindNext)(NULL, &desc); - if (component == NULL) { - return MA_FAILED_TO_INIT_BACKEND; - } - - status = ((ma_AudioComponentInstanceNew_proc)pContext->coreaudio.AudioComponentInstanceNew)(component, &audioUnit); - if (status != noErr) { - return ma_result_from_OSStatus(status); - } - - formatScope = (deviceType == ma_device_type_playback) ? kAudioUnitScope_Input : kAudioUnitScope_Output; - formatElement = (deviceType == ma_device_type_playback) ? MA_COREAUDIO_OUTPUT_BUS : MA_COREAUDIO_INPUT_BUS; - - propSize = sizeof(bestFormat); - status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, &propSize); - if (status != noErr) { - ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(audioUnit); - return ma_result_from_OSStatus(status); - } - - ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(audioUnit); - audioUnit = NULL; - - - pDeviceInfo->minChannels = bestFormat.mChannelsPerFrame; - pDeviceInfo->maxChannels = bestFormat.mChannelsPerFrame; - - pDeviceInfo->formatCount = 1; - result = ma_format_from_AudioStreamBasicDescription(&bestFormat, &pDeviceInfo->formats[0]); - if (result != MA_SUCCESS) { - return result; - } - /* - It looks like Apple are wanting to push the whole AVAudioSession thing. Thus, we need to use that to determine device settings. To do - this we just get the shared instance and inspect. + Need to use the tag to determine the channel map. For now I'm just assuming a default channel map, but later on this should + be updated to determine the mapping based on the tag. */ - @autoreleasepool { - AVAudioSession* pAudioSession = [AVAudioSession sharedInstance]; - ma_assert(pAudioSession != NULL); - - pDeviceInfo->minSampleRate = (ma_uint32)pAudioSession.sampleRate; - pDeviceInfo->maxSampleRate = pDeviceInfo->minSampleRate; + UInt32 channelCount = AudioChannelLayoutTag_GetNumberOfChannels(pChannelLayout->mChannelLayoutTag); + switch (pChannelLayout->mChannelLayoutTag) + { + case kAudioChannelLayoutTag_Mono: + case kAudioChannelLayoutTag_Stereo: + case kAudioChannelLayoutTag_StereoHeadphones: + case kAudioChannelLayoutTag_MatrixStereo: + case kAudioChannelLayoutTag_MidSide: + case kAudioChannelLayoutTag_XY: + case kAudioChannelLayoutTag_Binaural: + case kAudioChannelLayoutTag_Ambisonic_B_Format: + { + ma_get_standard_channel_map(ma_standard_channel_map_default, channelCount, channelMap); + } break; + + case kAudioChannelLayoutTag_Octagonal: + { + channelMap[7] = MA_CHANNEL_SIDE_RIGHT; + channelMap[6] = MA_CHANNEL_SIDE_LEFT; + } /* Intentional fallthrough. */ + case kAudioChannelLayoutTag_Hexagonal: + { + channelMap[5] = MA_CHANNEL_BACK_CENTER; + } /* Intentional fallthrough. */ + case kAudioChannelLayoutTag_Pentagonal: + { + channelMap[4] = MA_CHANNEL_FRONT_CENTER; + } /* Intentional fallghrough. */ + case kAudioChannelLayoutTag_Quadraphonic: + { + channelMap[3] = MA_CHANNEL_BACK_RIGHT; + channelMap[2] = MA_CHANNEL_BACK_LEFT; + channelMap[1] = MA_CHANNEL_RIGHT; + channelMap[0] = MA_CHANNEL_LEFT; + } break; + + /* TODO: Add support for more tags here. */ + + default: + { + ma_get_standard_channel_map(ma_standard_channel_map_default, channelCount, channelMap); + } break; } } -#endif - (void)pDeviceInfo; /* Unused. */ return MA_SUCCESS; } -OSStatus ma_on_output__coreaudio(void* pUserData, AudioUnitRenderActionFlags* pActionFlags, const AudioTimeStamp* pTimeStamp, UInt32 busNumber, UInt32 frameCount, AudioBufferList* pBufferList) +#if defined(MA_APPLE_DESKTOP) +static ma_result ma_get_device_object_ids__coreaudio(ma_context* pContext, UInt32* pDeviceCount, AudioObjectID** ppDeviceObjectIDs) /* NOTE: Free the returned buffer with ma_free(). */ { - ma_device* pDevice = (ma_device*)pUserData; - ma_stream_layout layout; - - ma_assert(pDevice != NULL); + AudioObjectPropertyAddress propAddressDevices; + UInt32 deviceObjectsDataSize; + OSStatus status; + AudioObjectID* pDeviceObjectIDs; -#if defined(MA_DEBUG_OUTPUT) - printf("INFO: Output Callback: busNumber=%d, frameCount=%d, mNumberBuffers=%d\n", busNumber, frameCount, pBufferList->mNumberBuffers); -#endif + MA_ASSERT(pContext != NULL); + MA_ASSERT(pDeviceCount != NULL); + MA_ASSERT(ppDeviceObjectIDs != NULL); - /* We need to check whether or not we are outputting interleaved or non-interleaved samples. The way we do this is slightly different for each type. */ - layout = ma_stream_layout_interleaved; - if (pBufferList->mBuffers[0].mNumberChannels != pDevice->playback.internalChannels) { - layout = ma_stream_layout_deinterleaved; - } + /* Safety. */ + *pDeviceCount = 0; + *ppDeviceObjectIDs = NULL; - if (layout == ma_stream_layout_interleaved) { - /* For now we can assume everything is interleaved. */ - UInt32 iBuffer; - for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; ++iBuffer) { - if (pBufferList->mBuffers[iBuffer].mNumberChannels == pDevice->playback.internalChannels) { - ma_uint32 frameCountForThisBuffer = pBufferList->mBuffers[iBuffer].mDataByteSize / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - if (frameCountForThisBuffer > 0) { - if (pDevice->type == ma_device_type_duplex) { - ma_device__handle_duplex_callback_playback(pDevice, frameCountForThisBuffer, pBufferList->mBuffers[iBuffer].mData, &pDevice->coreaudio.duplexRB); - } else { - ma_device__read_frames_from_client(pDevice, frameCountForThisBuffer, pBufferList->mBuffers[iBuffer].mData); - } - } - - #if defined(MA_DEBUG_OUTPUT) - printf(" frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", frameCount, pBufferList->mBuffers[iBuffer].mNumberChannels, pBufferList->mBuffers[iBuffer].mDataByteSize); - #endif - } else { - /* - This case is where the number of channels in the output buffer do not match our internal channels. It could mean that it's - not interleaved, in which case we can't handle right now since miniaudio does not yet support non-interleaved streams. We just - output silence here. - */ - ma_zero_memory(pBufferList->mBuffers[iBuffer].mData, pBufferList->mBuffers[iBuffer].mDataByteSize); - - #if defined(MA_DEBUG_OUTPUT) - printf(" WARNING: Outputting silence. frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", frameCount, pBufferList->mBuffers[iBuffer].mNumberChannels, pBufferList->mBuffers[iBuffer].mDataByteSize); - #endif - } - } - } else { - /* This is the deinterleaved case. We need to update each buffer in groups of internalChannels. This assumes each buffer is the same size. */ - ma_uint8 tempBuffer[4096]; - UInt32 iBuffer; - for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; iBuffer += pDevice->playback.internalChannels) { - ma_uint32 frameCountPerBuffer = pBufferList->mBuffers[iBuffer].mDataByteSize / ma_get_bytes_per_sample(pDevice->playback.internalFormat); - ma_uint32 framesRemaining = frameCountPerBuffer; + propAddressDevices.mSelector = kAudioHardwarePropertyDevices; + propAddressDevices.mScope = kAudioObjectPropertyScopeGlobal; + propAddressDevices.mElement = kAudioObjectPropertyElementMaster; - while (framesRemaining > 0) { - void* ppDeinterleavedBuffers[MA_MAX_CHANNELS]; - ma_uint32 iChannel; - ma_uint32 framesToRead = sizeof(tempBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - if (framesToRead > framesRemaining) { - framesToRead = framesRemaining; - } - - if (pDevice->type == ma_device_type_duplex) { - ma_device__handle_duplex_callback_playback(pDevice, framesToRead, tempBuffer, &pDevice->coreaudio.duplexRB); - } else { - ma_device__read_frames_from_client(pDevice, framesToRead, tempBuffer); - } - - for (iChannel = 0; iChannel < pDevice->playback.internalChannels; ++iChannel) { - ppDeinterleavedBuffers[iChannel] = (void*)ma_offset_ptr(pBufferList->mBuffers[iBuffer].mData, (frameCountPerBuffer - framesRemaining) * ma_get_bytes_per_sample(pDevice->playback.internalFormat)); - } - - ma_deinterleave_pcm_frames(pDevice->playback.internalFormat, pDevice->playback.internalChannels, framesToRead, tempBuffer, ppDeinterleavedBuffers); - - framesRemaining -= framesToRead; - } - } + status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(kAudioObjectSystemObject, &propAddressDevices, 0, NULL, &deviceObjectsDataSize); + if (status != noErr) { + return ma_result_from_OSStatus(status); } - (void)pActionFlags; - (void)pTimeStamp; - (void)busNumber; - - return noErr; -} - -OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFlags* pActionFlags, const AudioTimeStamp* pTimeStamp, UInt32 busNumber, UInt32 frameCount, AudioBufferList* pUnusedBufferList) -{ - ma_device* pDevice = (ma_device*)pUserData; - AudioBufferList* pRenderedBufferList; - ma_stream_layout layout; - OSStatus status; - - ma_assert(pDevice != NULL); - - pRenderedBufferList = (AudioBufferList*)pDevice->coreaudio.pAudioBufferList; - ma_assert(pRenderedBufferList); - - /* We need to check whether or not we are outputting interleaved or non-interleaved samples. The way we do this is slightly different for each type. */ - layout = ma_stream_layout_interleaved; - if (pRenderedBufferList->mBuffers[0].mNumberChannels != pDevice->capture.internalChannels) { - layout = ma_stream_layout_deinterleaved; + pDeviceObjectIDs = (AudioObjectID*)ma_malloc(deviceObjectsDataSize, &pContext->allocationCallbacks); + if (pDeviceObjectIDs == NULL) { + return MA_OUT_OF_MEMORY; } -#if defined(MA_DEBUG_OUTPUT) - printf("INFO: Input Callback: busNumber=%d, frameCount=%d, mNumberBuffers=%d\n", busNumber, frameCount, pRenderedBufferList->mNumberBuffers); -#endif - - status = ((ma_AudioUnitRender_proc)pDevice->pContext->coreaudio.AudioUnitRender)((AudioUnit)pDevice->coreaudio.audioUnitCapture, pActionFlags, pTimeStamp, busNumber, frameCount, pRenderedBufferList); + status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(kAudioObjectSystemObject, &propAddressDevices, 0, NULL, &deviceObjectsDataSize, pDeviceObjectIDs); if (status != noErr) { - #if defined(MA_DEBUG_OUTPUT) - printf(" ERROR: AudioUnitRender() failed with %d\n", status); - #endif - return status; + ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks); + return ma_result_from_OSStatus(status); } - if (layout == ma_stream_layout_interleaved) { - UInt32 iBuffer; - for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; ++iBuffer) { - if (pRenderedBufferList->mBuffers[iBuffer].mNumberChannels == pDevice->capture.internalChannels) { - if (pDevice->type == ma_device_type_duplex) { - ma_device__handle_duplex_callback_capture(pDevice, frameCount, pRenderedBufferList->mBuffers[iBuffer].mData, &pDevice->coreaudio.duplexRB); - } else { - ma_device__send_frames_to_client(pDevice, frameCount, pRenderedBufferList->mBuffers[iBuffer].mData); - } - #if defined(MA_DEBUG_OUTPUT) - printf(" mDataByteSize=%d\n", pRenderedBufferList->mBuffers[iBuffer].mDataByteSize); - #endif - } else { - /* - This case is where the number of channels in the output buffer do not match our internal channels. It could mean that it's - not interleaved, in which case we can't handle right now since miniaudio does not yet support non-interleaved streams. - */ - ma_uint8 silentBuffer[4096]; - ma_uint32 framesRemaining; - - ma_zero_memory(silentBuffer, sizeof(silentBuffer)); - - framesRemaining = frameCount; - while (framesRemaining > 0) { - ma_uint32 framesToSend = sizeof(silentBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - if (framesToSend > framesRemaining) { - framesToSend = framesRemaining; - } - - if (pDevice->type == ma_device_type_duplex) { - ma_device__handle_duplex_callback_capture(pDevice, framesToSend, silentBuffer, &pDevice->coreaudio.duplexRB); - } else { - ma_device__send_frames_to_client(pDevice, framesToSend, silentBuffer); - } - - framesRemaining -= framesToSend; - } - - #if defined(MA_DEBUG_OUTPUT) - printf(" WARNING: Outputting silence. frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", frameCount, pRenderedBufferList->mBuffers[iBuffer].mNumberChannels, pRenderedBufferList->mBuffers[iBuffer].mDataByteSize); - #endif - } - } - } else { - /* This is the deinterleaved case. We need to interleave the audio data before sending it to the client. This assumes each buffer is the same size. */ - ma_uint8 tempBuffer[4096]; - UInt32 iBuffer; - for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; iBuffer += pDevice->capture.internalChannels) { - ma_uint32 framesRemaining = frameCount; - while (framesRemaining > 0) { - void* ppDeinterleavedBuffers[MA_MAX_CHANNELS]; - ma_uint32 iChannel; - ma_uint32 framesToSend = sizeof(tempBuffer) / ma_get_bytes_per_sample(pDevice->capture.internalFormat); - if (framesToSend > framesRemaining) { - framesToSend = framesRemaining; - } - - for (iChannel = 0; iChannel < pDevice->capture.internalChannels; ++iChannel) { - ppDeinterleavedBuffers[iChannel] = (void*)ma_offset_ptr(pRenderedBufferList->mBuffers[iBuffer].mData, (frameCount - framesRemaining) * ma_get_bytes_per_sample(pDevice->capture.internalFormat)); - } - - ma_interleave_pcm_frames(pDevice->capture.internalFormat, pDevice->capture.internalChannels, framesToSend, (const void**)ppDeinterleavedBuffers, tempBuffer); - - if (pDevice->type == ma_device_type_duplex) { - ma_device__handle_duplex_callback_capture(pDevice, framesToSend, tempBuffer, &pDevice->coreaudio.duplexRB); - } else { - ma_device__send_frames_to_client(pDevice, framesToSend, tempBuffer); - } - - framesRemaining -= framesToSend; - } - } - } - - (void)pActionFlags; - (void)pTimeStamp; - (void)busNumber; - (void)frameCount; - (void)pUnusedBufferList; + *pDeviceCount = deviceObjectsDataSize / sizeof(AudioObjectID); + *ppDeviceObjectIDs = pDeviceObjectIDs; - return noErr; + return MA_SUCCESS; } -void on_start_stop__coreaudio(void* pUserData, AudioUnit audioUnit, AudioUnitPropertyID propertyID, AudioUnitScope scope, AudioUnitElement element) +static ma_result ma_get_AudioObject_uid_as_CFStringRef(ma_context* pContext, AudioObjectID objectID, CFStringRef* pUID) { - ma_device* pDevice = (ma_device*)pUserData; - ma_assert(pDevice != NULL); - - /* - There's been a report of a deadlock here when triggered by ma_device_uninit(). It looks like - AudioUnitGetProprty (called below) and AudioComponentInstanceDispose (called in ma_device_uninit) - can try waiting on the same lock. I'm going to try working around this by not calling any Core - Audio APIs in the callback when the device has been stopped or uninitialized. - */ - if (ma_device__get_state(pDevice) == MA_STATE_UNINITIALIZED || ma_device__get_state(pDevice) == MA_STATE_STOPPING || ma_device__get_state(pDevice) == MA_STATE_STOPPED) { - ma_stop_proc onStop = pDevice->onStop; - if (onStop) { - onStop(pDevice); - } - - ma_event_signal(&pDevice->coreaudio.stopEvent); - } else { - UInt32 isRunning; - UInt32 isRunningSize = sizeof(isRunning); - OSStatus status = ((ma_AudioUnitGetProperty_proc)pDevice->pContext->coreaudio.AudioUnitGetProperty)(audioUnit, kAudioOutputUnitProperty_IsRunning, scope, element, &isRunning, &isRunningSize); - if (status != noErr) { - return; /* Don't really know what to do in this case... just ignore it, I suppose... */ - } - - if (!isRunning) { - ma_stop_proc onStop; + AudioObjectPropertyAddress propAddress; + UInt32 dataSize; + OSStatus status; - /* - The stop event is a bit annoying in Core Audio because it will be called when we automatically switch the default device. Some scenarios to consider: - - 1) When the device is unplugged, this will be called _before_ the default device change notification. - 2) When the device is changed via the default device change notification, this will be called _after_ the switch. - - For case #1, we just check if there's a new default device available. If so, we just ignore the stop event. For case #2 we check a flag. - */ - if (((audioUnit == pDevice->coreaudio.audioUnitPlayback) && pDevice->coreaudio.isDefaultPlaybackDevice) || - ((audioUnit == pDevice->coreaudio.audioUnitCapture) && pDevice->coreaudio.isDefaultCaptureDevice)) { - /* - It looks like the device is switching through an external event, such as the user unplugging the device or changing the default device - via the operating system's sound settings. If we're re-initializing the device, we just terminate because we want the stopping of the - device to be seamless to the client (we don't want them receiving the onStop event and thinking that the device has stopped when it - hasn't!). - */ - if (((audioUnit == pDevice->coreaudio.audioUnitPlayback) && pDevice->coreaudio.isSwitchingPlaybackDevice) || - ((audioUnit == pDevice->coreaudio.audioUnitCapture) && pDevice->coreaudio.isSwitchingCaptureDevice)) { - return; - } - - /* - Getting here means the device is not reinitializing which means it may have been unplugged. From what I can see, it looks like Core Audio - will try switching to the new default device seamlessly. We need to somehow find a way to determine whether or not Core Audio will most - likely be successful in switching to the new device. - - TODO: Try to predict if Core Audio will switch devices. If not, the onStop callback needs to be posted. - */ - return; - } - - /* Getting here means we need to stop the device. */ - onStop = pDevice->onStop; - if (onStop) { - onStop(pDevice); - } - } + MA_ASSERT(pContext != NULL); + + propAddress.mSelector = kAudioDevicePropertyDeviceUID; + propAddress.mScope = kAudioObjectPropertyScopeGlobal; + propAddress.mElement = kAudioObjectPropertyElementMaster; + + dataSize = sizeof(*pUID); + status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(objectID, &propAddress, 0, NULL, &dataSize, pUID); + if (status != noErr) { + return ma_result_from_OSStatus(status); } + + return MA_SUCCESS; +} - (void)propertyID; /* Unused. */ +static ma_result ma_get_AudioObject_uid(ma_context* pContext, AudioObjectID objectID, size_t bufferSize, char* bufferOut) +{ + CFStringRef uid; + ma_result result; + + MA_ASSERT(pContext != NULL); + + result = ma_get_AudioObject_uid_as_CFStringRef(pContext, objectID, &uid); + if (result != MA_SUCCESS) { + return result; + } + + if (!((ma_CFStringGetCString_proc)pContext->coreaudio.CFStringGetCString)(uid, bufferOut, bufferSize, kCFStringEncodingUTF8)) { + return MA_ERROR; + } + + ((ma_CFRelease_proc)pContext->coreaudio.CFRelease)(uid); + return MA_SUCCESS; } -#if defined(MA_APPLE_DESKTOP) -static ma_uint32 g_DeviceTrackingInitCounter_CoreAudio = 0; -static ma_mutex g_DeviceTrackingMutex_CoreAudio; -static ma_device** g_ppTrackedDevices_CoreAudio = NULL; -static ma_uint32 g_TrackedDeviceCap_CoreAudio = 0; -static ma_uint32 g_TrackedDeviceCount_CoreAudio = 0; +static ma_result ma_get_AudioObject_name(ma_context* pContext, AudioObjectID objectID, size_t bufferSize, char* bufferOut) +{ + AudioObjectPropertyAddress propAddress; + CFStringRef deviceName = NULL; + UInt32 dataSize; + OSStatus status; + + MA_ASSERT(pContext != NULL); + + propAddress.mSelector = kAudioDevicePropertyDeviceNameCFString; + propAddress.mScope = kAudioObjectPropertyScopeGlobal; + propAddress.mElement = kAudioObjectPropertyElementMaster; + + dataSize = sizeof(deviceName); + status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(objectID, &propAddress, 0, NULL, &dataSize, &deviceName); + if (status != noErr) { + return ma_result_from_OSStatus(status); + } + + if (!((ma_CFStringGetCString_proc)pContext->coreaudio.CFStringGetCString)(deviceName, bufferOut, bufferSize, kCFStringEncodingUTF8)) { + return MA_ERROR; + } + + ((ma_CFRelease_proc)pContext->coreaudio.CFRelease)(deviceName); + return MA_SUCCESS; +} -OSStatus ma_default_device_changed__coreaudio(AudioObjectID objectID, UInt32 addressCount, const AudioObjectPropertyAddress* pAddresses, void* pUserData) +static ma_bool32 ma_does_AudioObject_support_scope(ma_context* pContext, AudioObjectID deviceObjectID, AudioObjectPropertyScope scope) { - ma_device_type deviceType; + AudioObjectPropertyAddress propAddress; + UInt32 dataSize; + OSStatus status; + AudioBufferList* pBufferList; + ma_bool32 isSupported; + + MA_ASSERT(pContext != NULL); + + /* To know whether or not a device is an input device we need ot look at the stream configuration. If it has an output channel it's a playback device. */ + propAddress.mSelector = kAudioDevicePropertyStreamConfiguration; + propAddress.mScope = scope; + propAddress.mElement = kAudioObjectPropertyElementMaster; - /* Not sure if I really need to check this, but it makes me feel better. */ - if (addressCount == 0) { - return noErr; + status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize); + if (status != noErr) { + return MA_FALSE; } - if (pAddresses[0].mSelector == kAudioHardwarePropertyDefaultOutputDevice) { - deviceType = ma_device_type_playback; - } else if (pAddresses[0].mSelector == kAudioHardwarePropertyDefaultInputDevice) { - deviceType = ma_device_type_capture; - } else { - return noErr; /* Should never hit this. */ + pBufferList = (AudioBufferList*)ma__malloc_from_callbacks(dataSize, &pContext->allocationCallbacks); + if (pBufferList == NULL) { + return MA_FALSE; /* Out of memory. */ } - ma_mutex_lock(&g_DeviceTrackingMutex_CoreAudio); - { - ma_uint32 iDevice; - for (iDevice = 0; iDevice < g_TrackedDeviceCount_CoreAudio; iDevice += 1) { - ma_result reinitResult; - ma_device* pDevice; - - pDevice = g_ppTrackedDevices_CoreAudio[iDevice]; - if (pDevice->type == deviceType || pDevice->type == ma_device_type_duplex) { - if (deviceType == ma_device_type_playback) { - pDevice->coreaudio.isSwitchingPlaybackDevice = MA_TRUE; - reinitResult = ma_device_reinit_internal__coreaudio(pDevice, deviceType, MA_TRUE); - pDevice->coreaudio.isSwitchingPlaybackDevice = MA_FALSE; - } else { - pDevice->coreaudio.isSwitchingCaptureDevice = MA_TRUE; - reinitResult = ma_device_reinit_internal__coreaudio(pDevice, deviceType, MA_TRUE); - pDevice->coreaudio.isSwitchingCaptureDevice = MA_FALSE; - } - - if (reinitResult == MA_SUCCESS) { - ma_device__post_init_setup(pDevice, deviceType); - - /* Restart the device if required. If this fails we need to stop the device entirely. */ - if (ma_device__get_state(pDevice) == MA_STATE_STARTED) { - OSStatus status; - if (deviceType == ma_device_type_playback) { - status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); - if (status != noErr) { - if (pDevice->type == ma_device_type_duplex) { - ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture); - } - ma_device__set_state(pDevice, MA_STATE_STOPPED); - } - } else if (deviceType == ma_device_type_capture) { - status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitCapture); - if (status != noErr) { - if (pDevice->type == ma_device_type_duplex) { - ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); - } - ma_device__set_state(pDevice, MA_STATE_STOPPED); - } - } - } - } - } - } + status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pBufferList); + if (status != noErr) { + ma__free_from_callbacks(pBufferList, &pContext->allocationCallbacks); + return MA_FALSE; + } + + isSupported = MA_FALSE; + if (pBufferList->mNumberBuffers > 0) { + isSupported = MA_TRUE; } - ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio); - (void)objectID; /* Unused. */ - return noErr; + ma__free_from_callbacks(pBufferList, &pContext->allocationCallbacks); + return isSupported; } -static ma_result ma_context__init_device_tracking__coreaudio(ma_context* pContext) +static ma_bool32 ma_does_AudioObject_support_playback(ma_context* pContext, AudioObjectID deviceObjectID) +{ + return ma_does_AudioObject_support_scope(pContext, deviceObjectID, kAudioObjectPropertyScopeOutput); +} + +static ma_bool32 ma_does_AudioObject_support_capture(ma_context* pContext, AudioObjectID deviceObjectID) +{ + return ma_does_AudioObject_support_scope(pContext, deviceObjectID, kAudioObjectPropertyScopeInput); +} + + +static ma_result ma_get_AudioObject_stream_descriptions(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, UInt32* pDescriptionCount, AudioStreamRangedDescription** ppDescriptions) /* NOTE: Free the returned pointer with ma_free(). */ { - ma_assert(pContext != NULL); + AudioObjectPropertyAddress propAddress; + UInt32 dataSize; + OSStatus status; + AudioStreamRangedDescription* pDescriptions; + + MA_ASSERT(pContext != NULL); + MA_ASSERT(pDescriptionCount != NULL); + MA_ASSERT(ppDescriptions != NULL); - if (ma_atomic_increment_32(&g_DeviceTrackingInitCounter_CoreAudio) == 1) { - AudioObjectPropertyAddress propAddress; - propAddress.mScope = kAudioObjectPropertyScopeGlobal; - propAddress.mElement = kAudioObjectPropertyElementMaster; - - ma_mutex_init(pContext, &g_DeviceTrackingMutex_CoreAudio); - - propAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice; - ((ma_AudioObjectAddPropertyListener_proc)pContext->coreaudio.AudioObjectAddPropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL); - - propAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice; - ((ma_AudioObjectAddPropertyListener_proc)pContext->coreaudio.AudioObjectAddPropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL); + /* + TODO: Experiment with kAudioStreamPropertyAvailablePhysicalFormats instead of (or in addition to) kAudioStreamPropertyAvailableVirtualFormats. My + MacBook Pro uses s24/32 format, however, which miniaudio does not currently support. + */ + propAddress.mSelector = kAudioStreamPropertyAvailableVirtualFormats; /*kAudioStreamPropertyAvailablePhysicalFormats;*/ + propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput; + propAddress.mElement = kAudioObjectPropertyElementMaster; + + status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize); + if (status != noErr) { + return ma_result_from_OSStatus(status); } + pDescriptions = (AudioStreamRangedDescription*)ma_malloc(dataSize, &pContext->allocationCallbacks); + if (pDescriptions == NULL) { + return MA_OUT_OF_MEMORY; + } + + status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pDescriptions); + if (status != noErr) { + ma_free(pDescriptions, &pContext->allocationCallbacks); + return ma_result_from_OSStatus(status); + } + + *pDescriptionCount = dataSize / sizeof(*pDescriptions); + *ppDescriptions = pDescriptions; return MA_SUCCESS; } -static ma_result ma_context__uninit_device_tracking__coreaudio(ma_context* pContext) + +static ma_result ma_get_AudioObject_channel_layout(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, AudioChannelLayout** ppChannelLayout) /* NOTE: Free the returned pointer with ma_free(). */ { - ma_assert(pContext != NULL); + AudioObjectPropertyAddress propAddress; + UInt32 dataSize; + OSStatus status; + AudioChannelLayout* pChannelLayout; + + MA_ASSERT(pContext != NULL); + MA_ASSERT(ppChannelLayout != NULL); - if (ma_atomic_decrement_32(&g_DeviceTrackingInitCounter_CoreAudio) == 0) { - AudioObjectPropertyAddress propAddress; - propAddress.mScope = kAudioObjectPropertyScopeGlobal; - propAddress.mElement = kAudioObjectPropertyElementMaster; - - propAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice; - ((ma_AudioObjectRemovePropertyListener_proc)pContext->coreaudio.AudioObjectRemovePropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL); - - propAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice; - ((ma_AudioObjectRemovePropertyListener_proc)pContext->coreaudio.AudioObjectRemovePropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL); - - /* At this point there should be no tracked devices. If so there's an error somewhere. */ - ma_assert(g_ppTrackedDevices_CoreAudio == NULL); - ma_assert(g_TrackedDeviceCount_CoreAudio == 0); - - ma_mutex_uninit(&g_DeviceTrackingMutex_CoreAudio); + *ppChannelLayout = NULL; /* Safety. */ + + propAddress.mSelector = kAudioDevicePropertyPreferredChannelLayout; + propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput; + propAddress.mElement = kAudioObjectPropertyElementMaster; + + status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize); + if (status != noErr) { + return ma_result_from_OSStatus(status); + } + + pChannelLayout = (AudioChannelLayout*)ma_malloc(dataSize, &pContext->allocationCallbacks); + if (pChannelLayout == NULL) { + return MA_OUT_OF_MEMORY; + } + + status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pChannelLayout); + if (status != noErr) { + ma_free(pChannelLayout, &pContext->allocationCallbacks); + return ma_result_from_OSStatus(status); } + *ppChannelLayout = pChannelLayout; return MA_SUCCESS; } -static ma_result ma_device__track__coreaudio(ma_device* pDevice) +static ma_result ma_get_AudioObject_channel_count(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32* pChannelCount) { + AudioChannelLayout* pChannelLayout; ma_result result; - ma_assert(pDevice != NULL); + MA_ASSERT(pContext != NULL); + MA_ASSERT(pChannelCount != NULL); - result = ma_context__init_device_tracking__coreaudio(pDevice->pContext); + *pChannelCount = 0; /* Safety. */ + + result = ma_get_AudioObject_channel_layout(pContext, deviceObjectID, deviceType, &pChannelLayout); if (result != MA_SUCCESS) { return result; } - ma_mutex_lock(&g_DeviceTrackingMutex_CoreAudio); - { - /* Allocate memory if required. */ - if (g_TrackedDeviceCap_CoreAudio <= g_TrackedDeviceCount_CoreAudio) { - ma_uint32 newCap; - ma_device** ppNewDevices; - - newCap = g_TrackedDeviceCap_CoreAudio * 2; - if (newCap == 0) { - newCap = 1; - } - - ppNewDevices = (ma_device**)ma_realloc(g_ppTrackedDevices_CoreAudio, sizeof(*g_ppTrackedDevices_CoreAudio) * newCap); - if (ppNewDevices == NULL) { - ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio); - return MA_OUT_OF_MEMORY; - } - - g_ppTrackedDevices_CoreAudio = ppNewDevices; - g_TrackedDeviceCap_CoreAudio = newCap; - } - - g_ppTrackedDevices_CoreAudio[g_TrackedDeviceCount_CoreAudio] = pDevice; - g_TrackedDeviceCount_CoreAudio += 1; + if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelDescriptions) { + *pChannelCount = pChannelLayout->mNumberChannelDescriptions; + } else if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelBitmap) { + *pChannelCount = ma_count_set_bits(pChannelLayout->mChannelBitmap); + } else { + *pChannelCount = AudioChannelLayoutTag_GetNumberOfChannels(pChannelLayout->mChannelLayoutTag); } - ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio); + ma_free(pChannelLayout, &pContext->allocationCallbacks); return MA_SUCCESS; } -static ma_result ma_device__untrack__coreaudio(ma_device* pDevice) +#if 0 +static ma_result ma_get_AudioObject_channel_map(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_channel channelMap[MA_MAX_CHANNELS]) { + AudioChannelLayout* pChannelLayout; ma_result result; + + MA_ASSERT(pContext != NULL); - ma_assert(pDevice != NULL); - - ma_mutex_lock(&g_DeviceTrackingMutex_CoreAudio); - { - ma_uint32 iDevice; - for (iDevice = 0; iDevice < g_TrackedDeviceCount_CoreAudio; iDevice += 1) { - if (g_ppTrackedDevices_CoreAudio[iDevice] == pDevice) { - /* We've found the device. We now need to remove it from the list. */ - ma_uint32 jDevice; - for (jDevice = iDevice; jDevice < g_TrackedDeviceCount_CoreAudio-1; jDevice += 1) { - g_ppTrackedDevices_CoreAudio[jDevice] = g_ppTrackedDevices_CoreAudio[jDevice+1]; - } - - g_TrackedDeviceCount_CoreAudio -= 1; - - /* If there's nothing else in the list we need to free memory. */ - if (g_TrackedDeviceCount_CoreAudio == 0) { - ma_free(g_ppTrackedDevices_CoreAudio); - g_ppTrackedDevices_CoreAudio = NULL; - g_TrackedDeviceCap_CoreAudio = 0; - } - - break; - } - } + result = ma_get_AudioObject_channel_layout(pContext, deviceObjectID, deviceType, &pChannelLayout); + if (result != MA_SUCCESS) { + return result; /* Rather than always failing here, would it be more robust to simply assume a default? */ } - ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio); - - result = ma_context__uninit_device_tracking__coreaudio(pDevice->pContext); + + result = ma_get_channel_map_from_AudioChannelLayout(pChannelLayout, channelMap); if (result != MA_SUCCESS) { + ma_free(pChannelLayout, &pContext->allocationCallbacks); return result; } - return MA_SUCCESS; + ma_free(pChannelLayout, &pContext->allocationCallbacks); + return result; } #endif -void ma_device_uninit__coreaudio(ma_device* pDevice) +static ma_result ma_get_AudioObject_sample_rates(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, UInt32* pSampleRateRangesCount, AudioValueRange** ppSampleRateRanges) /* NOTE: Free the returned pointer with ma_free(). */ { - ma_assert(pDevice != NULL); - ma_assert(ma_device__get_state(pDevice) == MA_STATE_UNINITIALIZED); + AudioObjectPropertyAddress propAddress; + UInt32 dataSize; + OSStatus status; + AudioValueRange* pSampleRateRanges; + + MA_ASSERT(pContext != NULL); + MA_ASSERT(pSampleRateRangesCount != NULL); + MA_ASSERT(ppSampleRateRanges != NULL); + + /* Safety. */ + *pSampleRateRangesCount = 0; + *ppSampleRateRanges = NULL; -#if defined(MA_APPLE_DESKTOP) - /* - Make sure we're no longer tracking the device. It doesn't matter if we call this for a non-default device because it'll - just gracefully ignore it. - */ - ma_device__untrack__coreaudio(pDevice); -#endif + propAddress.mSelector = kAudioDevicePropertyAvailableNominalSampleRates; + propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput; + propAddress.mElement = kAudioObjectPropertyElementMaster; - if (pDevice->coreaudio.audioUnitCapture != NULL) { - ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitCapture); - } - if (pDevice->coreaudio.audioUnitPlayback != NULL) { - ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); + status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize); + if (status != noErr) { + return ma_result_from_OSStatus(status); } - if (pDevice->coreaudio.pAudioBufferList) { - ma_free(pDevice->coreaudio.pAudioBufferList); + pSampleRateRanges = (AudioValueRange*)ma_malloc(dataSize, &pContext->allocationCallbacks); + if (pSampleRateRanges == NULL) { + return MA_OUT_OF_MEMORY; } - - if (pDevice->type == ma_device_type_duplex) { - ma_pcm_rb_uninit(&pDevice->coreaudio.duplexRB); + + status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pSampleRateRanges); + if (status != noErr) { + ma_free(pSampleRateRanges, &pContext->allocationCallbacks); + return ma_result_from_OSStatus(status); } + + *pSampleRateRangesCount = dataSize / sizeof(*pSampleRateRanges); + *ppSampleRateRanges = pSampleRateRanges; + return MA_SUCCESS; } -typedef struct -{ - /* Input. */ - ma_format formatIn; - ma_uint32 channelsIn; - ma_uint32 sampleRateIn; - ma_channel channelMapIn[MA_MAX_CHANNELS]; - ma_uint32 bufferSizeInFramesIn; - ma_uint32 bufferSizeInMillisecondsIn; - ma_uint32 periodsIn; - ma_bool32 usingDefaultFormat; - ma_bool32 usingDefaultChannels; - ma_bool32 usingDefaultSampleRate; - ma_bool32 usingDefaultChannelMap; - ma_share_mode shareMode; - ma_bool32 registerStopEvent; - - /* Output. */ -#if defined(MA_APPLE_DESKTOP) - AudioObjectID deviceObjectID; -#endif - AudioComponent component; - AudioUnit audioUnit; - AudioBufferList* pAudioBufferList; /* Only used for input devices. */ - ma_format formatOut; - ma_uint32 channelsOut; - ma_uint32 sampleRateOut; - ma_channel channelMapOut[MA_MAX_CHANNELS]; - ma_uint32 bufferSizeInFramesOut; - ma_uint32 periodsOut; - char deviceName[256]; -} ma_device_init_internal_data__coreaudio; - -ma_result ma_device_init_internal__coreaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_init_internal_data__coreaudio* pData, void* pDevice_DoNotReference) /* <-- pDevice is typed as void* intentionally so as to avoid accidentally referencing it. */ +#if 0 +static ma_result ma_get_AudioObject_get_closest_sample_rate(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32 sampleRateIn, ma_uint32* pSampleRateOut) { + UInt32 sampleRateRangeCount; + AudioValueRange* pSampleRateRanges; ma_result result; - OSStatus status; - UInt32 enableIOFlag; - AudioStreamBasicDescription bestFormat; - ma_uint32 actualBufferSizeInFrames; - AURenderCallbackStruct callbackInfo; -#if defined(MA_APPLE_DESKTOP) - AudioObjectID deviceObjectID; -#endif - - /* This API should only be used for a single device type: playback or capture. No full-duplex mode. */ - if (deviceType == ma_device_type_duplex) { - return MA_INVALID_ARGS; - } - ma_assert(pContext != NULL); - ma_assert(deviceType == ma_device_type_playback || deviceType == ma_device_type_capture); - -#if defined(MA_APPLE_DESKTOP) - pData->deviceObjectID = 0; -#endif - pData->component = NULL; - pData->audioUnit = NULL; - pData->pAudioBufferList = NULL; + MA_ASSERT(pContext != NULL); + MA_ASSERT(pSampleRateOut != NULL); -#if defined(MA_APPLE_DESKTOP) - result = ma_find_AudioObjectID(pContext, deviceType, pDeviceID, &deviceObjectID); + *pSampleRateOut = 0; /* Safety. */ + + result = ma_get_AudioObject_sample_rates(pContext, deviceObjectID, deviceType, &sampleRateRangeCount, &pSampleRateRanges); if (result != MA_SUCCESS) { return result; } - pData->deviceObjectID = deviceObjectID; -#endif - - /* Core audio doesn't really use the notion of a period so we can leave this unmodified, but not too over the top. */ - pData->periodsOut = pData->periodsIn; - if (pData->periodsOut == 0) { - pData->periodsOut = MA_DEFAULT_PERIODS; - } - if (pData->periodsOut > 16) { - pData->periodsOut = 16; + if (sampleRateRangeCount == 0) { + ma_free(pSampleRateRanges, &pContext->allocationCallbacks); + return MA_ERROR; /* Should never hit this case should we? */ } - - /* Audio unit. */ - status = ((ma_AudioComponentInstanceNew_proc)pContext->coreaudio.AudioComponentInstanceNew)((AudioComponent)pContext->coreaudio.component, (AudioUnit*)&pData->audioUnit); - if (status != noErr) { - return ma_result_from_OSStatus(status); + if (sampleRateIn == 0) { + /* Search in order of miniaudio's preferred priority. */ + UInt32 iMALSampleRate; + for (iMALSampleRate = 0; iMALSampleRate < ma_countof(g_maStandardSampleRatePriorities); ++iMALSampleRate) { + ma_uint32 malSampleRate = g_maStandardSampleRatePriorities[iMALSampleRate]; + UInt32 iCASampleRate; + for (iCASampleRate = 0; iCASampleRate < sampleRateRangeCount; ++iCASampleRate) { + AudioValueRange caSampleRate = pSampleRateRanges[iCASampleRate]; + if (caSampleRate.mMinimum <= malSampleRate && caSampleRate.mMaximum >= malSampleRate) { + *pSampleRateOut = malSampleRate; + ma_free(pSampleRateRanges, &pContext->allocationCallbacks); + return MA_SUCCESS; + } + } + } + + /* + If we get here it means none of miniaudio's standard sample rates matched any of the supported sample rates from the device. In this + case we just fall back to the first one reported by Core Audio. + */ + MA_ASSERT(sampleRateRangeCount > 0); + + *pSampleRateOut = pSampleRateRanges[0].mMinimum; + ma_free(pSampleRateRanges, &pContext->allocationCallbacks); + return MA_SUCCESS; + } else { + /* Find the closest match to this sample rate. */ + UInt32 currentAbsoluteDifference = INT32_MAX; + UInt32 iCurrentClosestRange = (UInt32)-1; + UInt32 iRange; + for (iRange = 0; iRange < sampleRateRangeCount; ++iRange) { + if (pSampleRateRanges[iRange].mMinimum <= sampleRateIn && pSampleRateRanges[iRange].mMaximum >= sampleRateIn) { + *pSampleRateOut = sampleRateIn; + ma_free(pSampleRateRanges, &pContext->allocationCallbacks); + return MA_SUCCESS; + } else { + UInt32 absoluteDifference; + if (pSampleRateRanges[iRange].mMinimum > sampleRateIn) { + absoluteDifference = pSampleRateRanges[iRange].mMinimum - sampleRateIn; + } else { + absoluteDifference = sampleRateIn - pSampleRateRanges[iRange].mMaximum; + } + + if (currentAbsoluteDifference > absoluteDifference) { + currentAbsoluteDifference = absoluteDifference; + iCurrentClosestRange = iRange; + } + } + } + + MA_ASSERT(iCurrentClosestRange != (UInt32)-1); + + *pSampleRateOut = pSampleRateRanges[iCurrentClosestRange].mMinimum; + ma_free(pSampleRateRanges, &pContext->allocationCallbacks); + return MA_SUCCESS; } + /* Should never get here, but it would mean we weren't able to find any suitable sample rates. */ + /*ma_free(pSampleRateRanges, &pContext->allocationCallbacks);*/ + /*return MA_ERROR;*/ +} +#endif + +static ma_result ma_get_AudioObject_closest_buffer_size_in_frames(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32 bufferSizeInFramesIn, ma_uint32* pBufferSizeInFramesOut) +{ + AudioObjectPropertyAddress propAddress; + AudioValueRange bufferSizeRange; + UInt32 dataSize; + OSStatus status; + + MA_ASSERT(pContext != NULL); + MA_ASSERT(pBufferSizeInFramesOut != NULL); - /* The input/output buses need to be explicitly enabled and disabled. We set the flag based on the output unit first, then we just swap it for input. */ - enableIOFlag = 1; - if (deviceType == ma_device_type_capture) { - enableIOFlag = 0; - } + *pBufferSizeInFramesOut = 0; /* Safety. */ - status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, MA_COREAUDIO_OUTPUT_BUS, &enableIOFlag, sizeof(enableIOFlag)); + propAddress.mSelector = kAudioDevicePropertyBufferFrameSizeRange; + propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput; + propAddress.mElement = kAudioObjectPropertyElementMaster; + + dataSize = sizeof(bufferSizeRange); + status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, &bufferSizeRange); if (status != noErr) { - ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); return ma_result_from_OSStatus(status); } - enableIOFlag = (enableIOFlag == 0) ? 1 : 0; - status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, MA_COREAUDIO_INPUT_BUS, &enableIOFlag, sizeof(enableIOFlag)); - if (status != noErr) { - ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); - return ma_result_from_OSStatus(status); + /* This is just a clamp. */ + if (bufferSizeInFramesIn < bufferSizeRange.mMinimum) { + *pBufferSizeInFramesOut = (ma_uint32)bufferSizeRange.mMinimum; + } else if (bufferSizeInFramesIn > bufferSizeRange.mMaximum) { + *pBufferSizeInFramesOut = (ma_uint32)bufferSizeRange.mMaximum; + } else { + *pBufferSizeInFramesOut = bufferSizeInFramesIn; + } + + return MA_SUCCESS; +} + +static ma_result ma_set_AudioObject_buffer_size_in_frames(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32* pPeriodSizeInOut) +{ + ma_result result; + ma_uint32 chosenBufferSizeInFrames; + AudioObjectPropertyAddress propAddress; + UInt32 dataSize; + OSStatus status; + + MA_ASSERT(pContext != NULL); + + result = ma_get_AudioObject_closest_buffer_size_in_frames(pContext, deviceObjectID, deviceType, *pPeriodSizeInOut, &chosenBufferSizeInFrames); + if (result != MA_SUCCESS) { + return result; } + + /* Try setting the size of the buffer... If this fails we just use whatever is currently set. */ + propAddress.mSelector = kAudioDevicePropertyBufferFrameSize; + propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput; + propAddress.mElement = kAudioObjectPropertyElementMaster; + ((ma_AudioObjectSetPropertyData_proc)pContext->coreaudio.AudioObjectSetPropertyData)(deviceObjectID, &propAddress, 0, NULL, sizeof(chosenBufferSizeInFrames), &chosenBufferSizeInFrames); - /* Set the device to use with this audio unit. This is only used on desktop since we are using defaults on mobile. */ -#if defined(MA_APPLE_DESKTOP) - status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, (deviceType == ma_device_type_playback) ? MA_COREAUDIO_OUTPUT_BUS : MA_COREAUDIO_INPUT_BUS, &deviceObjectID, sizeof(AudioDeviceID)); + /* Get the actual size of the buffer. */ + dataSize = sizeof(*pPeriodSizeInOut); + status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, &chosenBufferSizeInFrames); if (status != noErr) { - ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); - return ma_result_from_OSStatus(result); + return ma_result_from_OSStatus(status); } -#endif - - /* - Format. This is the hardest part of initialization because there's a few variables to take into account. - 1) The format must be supported by the device. - 2) The format must be supported miniaudio. - 3) There's a priority that miniaudio prefers. - - Ideally we would like to use a format that's as close to the hardware as possible so we can get as close to a passthrough as possible. The - most important property is the sample rate. miniaudio can do format conversion for any sample rate and channel count, but cannot do the same - for the sample data format. If the sample data format is not supported by miniaudio it must be ignored completely. - On mobile platforms this is a bit different. We just force the use of whatever the audio unit's current format is set to. - */ - { - AudioUnitScope formatScope = (deviceType == ma_device_type_playback) ? kAudioUnitScope_Input : kAudioUnitScope_Output; - AudioUnitElement formatElement = (deviceType == ma_device_type_playback) ? MA_COREAUDIO_OUTPUT_BUS : MA_COREAUDIO_INPUT_BUS; + *pPeriodSizeInOut = chosenBufferSizeInFrames; + return MA_SUCCESS; +} - #if defined(MA_APPLE_DESKTOP) - AudioStreamBasicDescription origFormat; - UInt32 origFormatSize; - result = ma_find_best_format__coreaudio(pContext, deviceObjectID, deviceType, pData->formatIn, pData->channelsIn, pData->sampleRateIn, pData->usingDefaultFormat, pData->usingDefaultChannels, pData->usingDefaultSampleRate, &bestFormat); - if (result != MA_SUCCESS) { - ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); - return result; - } - - /* From what I can see, Apple's documentation implies that we should keep the sample rate consistent. */ - origFormatSize = sizeof(origFormat); +static ma_result ma_find_AudioObjectID(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, AudioObjectID* pDeviceObjectID) +{ + MA_ASSERT(pContext != NULL); + MA_ASSERT(pDeviceObjectID != NULL); + + /* Safety. */ + *pDeviceObjectID = 0; + + if (pDeviceID == NULL) { + /* Default device. */ + AudioObjectPropertyAddress propAddressDefaultDevice; + UInt32 defaultDeviceObjectIDSize = sizeof(AudioObjectID); + AudioObjectID defaultDeviceObjectID; + OSStatus status; + + propAddressDefaultDevice.mScope = kAudioObjectPropertyScopeGlobal; + propAddressDefaultDevice.mElement = kAudioObjectPropertyElementMaster; if (deviceType == ma_device_type_playback) { - status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, MA_COREAUDIO_OUTPUT_BUS, &origFormat, &origFormatSize); + propAddressDefaultDevice.mSelector = kAudioHardwarePropertyDefaultOutputDevice; } else { - status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, MA_COREAUDIO_INPUT_BUS, &origFormat, &origFormatSize); - } - - if (status != noErr) { - ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); - return result; - } - - bestFormat.mSampleRate = origFormat.mSampleRate; - - status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, sizeof(bestFormat)); - if (status != noErr) { - /* We failed to set the format, so fall back to the current format of the audio unit. */ - bestFormat = origFormat; - } - #else - UInt32 propSize = sizeof(bestFormat); - status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, &propSize); - if (status != noErr) { - ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); - return ma_result_from_OSStatus(status); - } - - /* - Sample rate is a little different here because for some reason kAudioUnitProperty_StreamFormat returns 0... Oh well. We need to instead try - setting the sample rate to what the user has requested and then just see the results of it. Need to use some Objective-C here for this since - it depends on Apple's AVAudioSession API. To do this we just get the shared AVAudioSession instance and then set it. Note that from what I - can tell, it looks like the sample rate is shared between playback and capture for everything. - */ - @autoreleasepool { - AVAudioSession* pAudioSession = [AVAudioSession sharedInstance]; - ma_assert(pAudioSession != NULL); - - [pAudioSession setPreferredSampleRate:(double)pData->sampleRateIn error:nil]; - bestFormat.mSampleRate = pAudioSession.sampleRate; + propAddressDefaultDevice.mSelector = kAudioHardwarePropertyDefaultInputDevice; } - status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, sizeof(bestFormat)); - if (status != noErr) { - ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); - return ma_result_from_OSStatus(status); + defaultDeviceObjectIDSize = sizeof(AudioObjectID); + status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(kAudioObjectSystemObject, &propAddressDefaultDevice, 0, NULL, &defaultDeviceObjectIDSize, &defaultDeviceObjectID); + if (status == noErr) { + *pDeviceObjectID = defaultDeviceObjectID; + return MA_SUCCESS; } - #endif - - result = ma_format_from_AudioStreamBasicDescription(&bestFormat, &pData->formatOut); + } else { + /* Explicit device. */ + UInt32 deviceCount; + AudioObjectID* pDeviceObjectIDs; + ma_result result; + UInt32 iDevice; + + result = ma_get_device_object_ids__coreaudio(pContext, &deviceCount, &pDeviceObjectIDs); if (result != MA_SUCCESS) { - ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); return result; } - if (pData->formatOut == ma_format_unknown) { - ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); - return MA_FORMAT_NOT_SUPPORTED; + for (iDevice = 0; iDevice < deviceCount; ++iDevice) { + AudioObjectID deviceObjectID = pDeviceObjectIDs[iDevice]; + + char uid[256]; + if (ma_get_AudioObject_uid(pContext, deviceObjectID, sizeof(uid), uid) != MA_SUCCESS) { + continue; + } + + if (deviceType == ma_device_type_playback) { + if (ma_does_AudioObject_support_playback(pContext, deviceObjectID)) { + if (strcmp(uid, pDeviceID->coreaudio) == 0) { + *pDeviceObjectID = deviceObjectID; + ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks); + return MA_SUCCESS; + } + } + } else { + if (ma_does_AudioObject_support_capture(pContext, deviceObjectID)) { + if (strcmp(uid, pDeviceID->coreaudio) == 0) { + *pDeviceObjectID = deviceObjectID; + ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks); + return MA_SUCCESS; + } + } + } } - - pData->channelsOut = bestFormat.mChannelsPerFrame; - pData->sampleRateOut = bestFormat.mSampleRate; + + ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks); } - /* - Internal channel map. This is weird in my testing. If I use the AudioObject to get the - channel map, the channel descriptions are set to "Unknown" for some reason. To work around - this it looks like retrieving it from the AudioUnit will work. However, and this is where - it gets weird, it doesn't seem to work with capture devices, nor at all on iOS... Therefore - I'm going to fall back to a default assumption in these cases. - */ -#if defined(MA_APPLE_DESKTOP) - result = ma_get_AudioUnit_channel_map(pContext, pData->audioUnit, deviceType, pData->channelMapOut); + /* If we get here it means we couldn't find the device. */ + return MA_NO_DEVICE; +} + + +static ma_result ma_find_best_format__coreaudio(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_bool32 usingDefaultFormat, ma_bool32 usingDefaultChannels, ma_bool32 usingDefaultSampleRate, AudioStreamBasicDescription* pFormat) +{ + UInt32 deviceFormatDescriptionCount; + AudioStreamRangedDescription* pDeviceFormatDescriptions; + ma_result result; + ma_uint32 desiredSampleRate; + ma_uint32 desiredChannelCount; + ma_format desiredFormat; + AudioStreamBasicDescription bestDeviceFormatSoFar; + ma_bool32 hasSupportedFormat; + UInt32 iFormat; + + result = ma_get_AudioObject_stream_descriptions(pContext, deviceObjectID, deviceType, &deviceFormatDescriptionCount, &pDeviceFormatDescriptions); if (result != MA_SUCCESS) { - #if 0 - /* Try falling back to the channel map from the AudioObject. */ - result = ma_get_AudioObject_channel_map(pContext, deviceObjectID, deviceType, pData->channelMapOut); - if (result != MA_SUCCESS) { - return result; - } - #else - /* Fall back to default assumptions. */ - ma_get_standard_channel_map(ma_standard_channel_map_default, pData->channelsOut, pData->channelMapOut); - #endif + return result; } -#else - /* TODO: Figure out how to get the channel map using AVAudioSession. */ - ma_get_standard_channel_map(ma_standard_channel_map_default, pData->channelsOut, pData->channelMapOut); -#endif + desiredSampleRate = sampleRate; + if (usingDefaultSampleRate) { + /* + When using the device's default sample rate, we get the highest priority standard rate supported by the device. Otherwise + we just use the pre-set rate. + */ + ma_uint32 iStandardRate; + for (iStandardRate = 0; iStandardRate < ma_countof(g_maStandardSampleRatePriorities); ++iStandardRate) { + ma_uint32 standardRate = g_maStandardSampleRatePriorities[iStandardRate]; + ma_bool32 foundRate = MA_FALSE; + UInt32 iDeviceRate; - /* Buffer size. Not allowing this to be configurable on iOS. */ - actualBufferSizeInFrames = pData->bufferSizeInFramesIn; + for (iDeviceRate = 0; iDeviceRate < deviceFormatDescriptionCount; ++iDeviceRate) { + ma_uint32 deviceRate = (ma_uint32)pDeviceFormatDescriptions[iDeviceRate].mFormat.mSampleRate; + + if (deviceRate == standardRate) { + desiredSampleRate = standardRate; + foundRate = MA_TRUE; + break; + } + } + + if (foundRate) { + break; + } + } + } -#if defined(MA_APPLE_DESKTOP) - if (actualBufferSizeInFrames == 0) { - actualBufferSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pData->bufferSizeInMillisecondsIn, pData->sampleRateOut); + desiredChannelCount = channels; + if (usingDefaultChannels) { + ma_get_AudioObject_channel_count(pContext, deviceObjectID, deviceType, &desiredChannelCount); /* <-- Not critical if this fails. */ } - actualBufferSizeInFrames = actualBufferSizeInFrames / pData->periodsOut; - result = ma_set_AudioObject_buffer_size_in_frames(pContext, deviceObjectID, deviceType, &actualBufferSizeInFrames); - if (result != MA_SUCCESS) { - return result; + desiredFormat = format; + if (usingDefaultFormat) { + desiredFormat = g_maFormatPriorities[0]; } - pData->bufferSizeInFramesOut = actualBufferSizeInFrames * pData->periodsOut; -#else - actualBufferSizeInFrames = 4096; - pData->bufferSizeInFramesOut = actualBufferSizeInFrames; -#endif - - /* - During testing I discovered that the buffer size can be too big. You'll get an error like this: - - kAudioUnitErr_TooManyFramesToProcess : inFramesToProcess=4096, mMaxFramesPerSlice=512 - - Note how inFramesToProcess is smaller than mMaxFramesPerSlice. To fix, we need to set kAudioUnitProperty_MaximumFramesPerSlice to that - of the size of our buffer, or do it the other way around and set our buffer size to the kAudioUnitProperty_MaximumFramesPerSlice. + If we get here it means we don't have an exact match to what the client is asking for. We'll need to find the closest one. The next + loop will check for formats that have the same sample rate to what we're asking for. If there is, we prefer that one in all cases. */ - { - /*AudioUnitScope propScope = (deviceType == ma_device_type_playback) ? kAudioUnitScope_Input : kAudioUnitScope_Output; - AudioUnitElement propBus = (deviceType == ma_device_type_playback) ? MA_COREAUDIO_OUTPUT_BUS : MA_COREAUDIO_INPUT_BUS; + MA_ZERO_OBJECT(&bestDeviceFormatSoFar); - status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, propScope, propBus, &actualBufferSizeInFrames, sizeof(actualBufferSizeInFrames)); - if (status != noErr) { - ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); - return ma_result_from_OSStatus(status); - }*/ - - status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, 0, &actualBufferSizeInFrames, sizeof(actualBufferSizeInFrames)); - if (status != noErr) { - ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); - return ma_result_from_OSStatus(status); + hasSupportedFormat = MA_FALSE; + for (iFormat = 0; iFormat < deviceFormatDescriptionCount; ++iFormat) { + ma_format format; + ma_result formatResult = ma_format_from_AudioStreamBasicDescription(&pDeviceFormatDescriptions[iFormat].mFormat, &format); + if (formatResult == MA_SUCCESS && format != ma_format_unknown) { + hasSupportedFormat = MA_TRUE; + bestDeviceFormatSoFar = pDeviceFormatDescriptions[iFormat].mFormat; + break; } } - /* We need a buffer list if this is an input device. We render into this in the input callback. */ - if (deviceType == ma_device_type_capture) { - ma_bool32 isInterleaved = (bestFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved) == 0; - size_t allocationSize; - AudioBufferList* pBufferList; + if (!hasSupportedFormat) { + ma_free(pDeviceFormatDescriptions, &pContext->allocationCallbacks); + return MA_FORMAT_NOT_SUPPORTED; + } + + + for (iFormat = 0; iFormat < deviceFormatDescriptionCount; ++iFormat) { + AudioStreamBasicDescription thisDeviceFormat = pDeviceFormatDescriptions[iFormat].mFormat; + ma_format thisSampleFormat; + ma_result formatResult; + ma_format bestSampleFormatSoFar; - allocationSize = sizeof(AudioBufferList) - sizeof(AudioBuffer); /* Subtract sizeof(AudioBuffer) because that part is dynamically sized. */ - if (isInterleaved) { - /* Interleaved case. This is the simple case because we just have one buffer. */ - allocationSize += sizeof(AudioBuffer) * 1; - allocationSize += actualBufferSizeInFrames * ma_get_bytes_per_frame(pData->formatOut, pData->channelsOut); - } else { - /* Non-interleaved case. This is the more complex case because there's more than one buffer. */ - allocationSize += sizeof(AudioBuffer) * pData->channelsOut; - allocationSize += actualBufferSizeInFrames * ma_get_bytes_per_sample(pData->formatOut) * pData->channelsOut; + /* If the format is not supported by miniaudio we need to skip this one entirely. */ + formatResult = ma_format_from_AudioStreamBasicDescription(&pDeviceFormatDescriptions[iFormat].mFormat, &thisSampleFormat); + if (formatResult != MA_SUCCESS || thisSampleFormat == ma_format_unknown) { + continue; /* The format is not supported by miniaudio. Skip. */ } - pBufferList = (AudioBufferList*)ma_malloc(allocationSize); - if (pBufferList == NULL) { - ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); - return MA_OUT_OF_MEMORY; - } + ma_format_from_AudioStreamBasicDescription(&bestDeviceFormatSoFar, &bestSampleFormatSoFar); - if (isInterleaved) { - pBufferList->mNumberBuffers = 1; - pBufferList->mBuffers[0].mNumberChannels = pData->channelsOut; - pBufferList->mBuffers[0].mDataByteSize = actualBufferSizeInFrames * ma_get_bytes_per_frame(pData->formatOut, pData->channelsOut); - pBufferList->mBuffers[0].mData = (ma_uint8*)pBufferList + sizeof(AudioBufferList); + /* Getting here means the format is supported by miniaudio which makes this format a candidate. */ + if (thisDeviceFormat.mSampleRate != desiredSampleRate) { + /* + The sample rate does not match, but this format could still be usable, although it's a very low priority. If the best format + so far has an equal sample rate we can just ignore this one. + */ + if (bestDeviceFormatSoFar.mSampleRate == desiredSampleRate) { + continue; /* The best sample rate so far has the same sample rate as what we requested which means it's still the best so far. Skip this format. */ + } else { + /* In this case, neither the best format so far nor this one have the same sample rate. Check the channel count next. */ + if (thisDeviceFormat.mChannelsPerFrame != desiredChannelCount) { + /* This format has a different sample rate _and_ a different channel count. */ + if (bestDeviceFormatSoFar.mChannelsPerFrame == desiredChannelCount) { + continue; /* No change to the best format. */ + } else { + /* + Both this format and the best so far have different sample rates and different channel counts. Whichever has the + best format is the new best. + */ + if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) { + bestDeviceFormatSoFar = thisDeviceFormat; + continue; + } else { + continue; /* No change to the best format. */ + } + } + } else { + /* This format has a different sample rate but the desired channel count. */ + if (bestDeviceFormatSoFar.mChannelsPerFrame == desiredChannelCount) { + /* Both this format and the best so far have the desired channel count. Whichever has the best format is the new best. */ + if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) { + bestDeviceFormatSoFar = thisDeviceFormat; + continue; + } else { + continue; /* No change to the best format for now. */ + } + } else { + /* This format has the desired channel count, but the best so far does not. We have a new best. */ + bestDeviceFormatSoFar = thisDeviceFormat; + continue; + } + } + } } else { - ma_uint32 iBuffer; - pBufferList->mNumberBuffers = pData->channelsOut; - for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; ++iBuffer) { - pBufferList->mBuffers[iBuffer].mNumberChannels = 1; - pBufferList->mBuffers[iBuffer].mDataByteSize = actualBufferSizeInFrames * ma_get_bytes_per_sample(pData->formatOut); - pBufferList->mBuffers[iBuffer].mData = (ma_uint8*)pBufferList + ((sizeof(AudioBufferList) - sizeof(AudioBuffer)) + (sizeof(AudioBuffer) * pData->channelsOut)) + (actualBufferSizeInFrames * ma_get_bytes_per_sample(pData->formatOut) * iBuffer); + /* + The sample rates match which makes this format a very high priority contender. If the best format so far has a different + sample rate it needs to be replaced with this one. + */ + if (bestDeviceFormatSoFar.mSampleRate != desiredSampleRate) { + bestDeviceFormatSoFar = thisDeviceFormat; + continue; + } else { + /* In this case both this format and the best format so far have the same sample rate. Check the channel count next. */ + if (thisDeviceFormat.mChannelsPerFrame == desiredChannelCount) { + /* + In this case this format has the same channel count as what the client is requesting. If the best format so far has + a different count, this one becomes the new best. + */ + if (bestDeviceFormatSoFar.mChannelsPerFrame != desiredChannelCount) { + bestDeviceFormatSoFar = thisDeviceFormat; + continue; + } else { + /* In this case both this format and the best so far have the ideal sample rate and channel count. Check the format. */ + if (thisSampleFormat == desiredFormat) { + bestDeviceFormatSoFar = thisDeviceFormat; + break; /* Found the exact match. */ + } else { + /* The formats are different. The new best format is the one with the highest priority format according to miniaudio. */ + if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) { + bestDeviceFormatSoFar = thisDeviceFormat; + continue; + } else { + continue; /* No change to the best format for now. */ + } + } + } + } else { + /* + In this case the channel count is different to what the client has requested. If the best so far has the same channel + count as the requested count then it remains the best. + */ + if (bestDeviceFormatSoFar.mChannelsPerFrame == desiredChannelCount) { + continue; + } else { + /* + This is the case where both have the same sample rate (good) but different channel counts. Right now both have about + the same priority, but we need to compare the format now. + */ + if (thisSampleFormat == bestSampleFormatSoFar) { + if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) { + bestDeviceFormatSoFar = thisDeviceFormat; + continue; + } else { + continue; /* No change to the best format for now. */ + } + } + } + } } } - - pData->pAudioBufferList = pBufferList; } - /* Callbacks. */ - callbackInfo.inputProcRefCon = pDevice_DoNotReference; + *pFormat = bestDeviceFormatSoFar; + + ma_free(pDeviceFormatDescriptions, &pContext->allocationCallbacks); + return MA_SUCCESS; +} +#endif + +static ma_result ma_get_AudioUnit_channel_map(ma_context* pContext, AudioUnit audioUnit, ma_device_type deviceType, ma_channel channelMap[MA_MAX_CHANNELS]) +{ + AudioUnitScope deviceScope; + AudioUnitElement deviceBus; + UInt32 channelLayoutSize; + OSStatus status; + AudioChannelLayout* pChannelLayout; + ma_result result; + + MA_ASSERT(pContext != NULL); + if (deviceType == ma_device_type_playback) { - callbackInfo.inputProc = ma_on_output__coreaudio; - status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Global, MA_COREAUDIO_OUTPUT_BUS, &callbackInfo, sizeof(callbackInfo)); - if (status != noErr) { - ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); - return ma_result_from_OSStatus(status); - } + deviceScope = kAudioUnitScope_Output; + deviceBus = MA_COREAUDIO_OUTPUT_BUS; } else { - callbackInfo.inputProc = ma_on_input__coreaudio; - status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, MA_COREAUDIO_INPUT_BUS, &callbackInfo, sizeof(callbackInfo)); - if (status != noErr) { - ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); - return ma_result_from_OSStatus(status); - } + deviceScope = kAudioUnitScope_Input; + deviceBus = MA_COREAUDIO_INPUT_BUS; } - /* We need to listen for stop events. */ - if (pData->registerStopEvent) { - status = ((ma_AudioUnitAddPropertyListener_proc)pContext->coreaudio.AudioUnitAddPropertyListener)(pData->audioUnit, kAudioOutputUnitProperty_IsRunning, on_start_stop__coreaudio, pDevice_DoNotReference); - if (status != noErr) { - ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); - return ma_result_from_OSStatus(status); - } + status = ((ma_AudioUnitGetPropertyInfo_proc)pContext->coreaudio.AudioUnitGetPropertyInfo)(audioUnit, kAudioUnitProperty_AudioChannelLayout, deviceScope, deviceBus, &channelLayoutSize, NULL); + if (status != noErr) { + return ma_result_from_OSStatus(status); } - /* Initialize the audio unit. */ - status = ((ma_AudioUnitInitialize_proc)pContext->coreaudio.AudioUnitInitialize)(pData->audioUnit); + pChannelLayout = (AudioChannelLayout*)ma__malloc_from_callbacks(channelLayoutSize, &pContext->allocationCallbacks); + if (pChannelLayout == NULL) { + return MA_OUT_OF_MEMORY; + } + + status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(audioUnit, kAudioUnitProperty_AudioChannelLayout, deviceScope, deviceBus, pChannelLayout, &channelLayoutSize); if (status != noErr) { - ma_free(pData->pAudioBufferList); - pData->pAudioBufferList = NULL; - ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); + ma__free_from_callbacks(pChannelLayout, &pContext->allocationCallbacks); return ma_result_from_OSStatus(status); } - /* Grab the name. */ -#if defined(MA_APPLE_DESKTOP) - ma_get_AudioObject_name(pContext, deviceObjectID, sizeof(pData->deviceName), pData->deviceName); -#else - if (deviceType == ma_device_type_playback) { - ma_strcpy_s(pData->deviceName, sizeof(pData->deviceName), MA_DEFAULT_PLAYBACK_DEVICE_NAME); - } else { - ma_strcpy_s(pData->deviceName, sizeof(pData->deviceName), MA_DEFAULT_CAPTURE_DEVICE_NAME); + result = ma_get_channel_map_from_AudioChannelLayout(pChannelLayout, channelMap); + if (result != MA_SUCCESS) { + ma__free_from_callbacks(pChannelLayout, &pContext->allocationCallbacks); + return result; } -#endif - - return result; + + ma__free_from_callbacks(pChannelLayout, &pContext->allocationCallbacks); + return MA_SUCCESS; } -ma_result ma_device_reinit_internal__coreaudio(ma_device* pDevice, ma_device_type deviceType, ma_bool32 disposePreviousAudioUnit) +static ma_bool32 ma_context_is_device_id_equal__coreaudio(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) { - ma_device_init_internal_data__coreaudio data; - ma_result result; - - /* This should only be called for playback or capture, not duplex. */ - if (deviceType == ma_device_type_duplex) { - return MA_INVALID_ARGS; - } + MA_ASSERT(pContext != NULL); + MA_ASSERT(pID0 != NULL); + MA_ASSERT(pID1 != NULL); + (void)pContext; - if (deviceType == ma_device_type_capture) { - data.formatIn = pDevice->capture.format; - data.channelsIn = pDevice->capture.channels; - data.sampleRateIn = pDevice->sampleRate; - ma_copy_memory(data.channelMapIn, pDevice->capture.channelMap, sizeof(pDevice->capture.channelMap)); - data.usingDefaultFormat = pDevice->capture.usingDefaultFormat; - data.usingDefaultChannels = pDevice->capture.usingDefaultChannels; - data.usingDefaultSampleRate = pDevice->usingDefaultSampleRate; - data.usingDefaultChannelMap = pDevice->capture.usingDefaultChannelMap; - data.shareMode = pDevice->capture.shareMode; - data.registerStopEvent = MA_TRUE; - - if (disposePreviousAudioUnit) { - ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture); - ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitCapture); - } - if (pDevice->coreaudio.pAudioBufferList) { - ma_free(pDevice->coreaudio.pAudioBufferList); - } - } else if (deviceType == ma_device_type_playback) { - data.formatIn = pDevice->playback.format; - data.channelsIn = pDevice->playback.channels; - data.sampleRateIn = pDevice->sampleRate; - ma_copy_memory(data.channelMapIn, pDevice->playback.channelMap, sizeof(pDevice->playback.channelMap)); - data.usingDefaultFormat = pDevice->playback.usingDefaultFormat; - data.usingDefaultChannels = pDevice->playback.usingDefaultChannels; - data.usingDefaultSampleRate = pDevice->usingDefaultSampleRate; - data.usingDefaultChannelMap = pDevice->playback.usingDefaultChannelMap; - data.shareMode = pDevice->playback.shareMode; - data.registerStopEvent = (pDevice->type != ma_device_type_duplex); - - if (disposePreviousAudioUnit) { - ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); - ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); - } - } - data.bufferSizeInFramesIn = pDevice->coreaudio.originalBufferSizeInFrames; - data.bufferSizeInMillisecondsIn = pDevice->coreaudio.originalBufferSizeInMilliseconds; - data.periodsIn = pDevice->coreaudio.originalPeriods; + return strcmp(pID0->coreaudio, pID1->coreaudio) == 0; +} - /* Need at least 3 periods for duplex. */ - if (data.periodsIn < 3 && pDevice->type == ma_device_type_duplex) { - data.periodsIn = 3; - } +static ma_result ma_context_enumerate_devices__coreaudio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +{ +#if defined(MA_APPLE_DESKTOP) + UInt32 deviceCount; + AudioObjectID* pDeviceObjectIDs; + ma_result result; + UInt32 iDevice; - result = ma_device_init_internal__coreaudio(pDevice->pContext, deviceType, NULL, &data, (void*)pDevice); + result = ma_get_device_object_ids__coreaudio(pContext, &deviceCount, &pDeviceObjectIDs); if (result != MA_SUCCESS) { return result; } + + for (iDevice = 0; iDevice < deviceCount; ++iDevice) { + AudioObjectID deviceObjectID = pDeviceObjectIDs[iDevice]; + ma_device_info info; + + MA_ZERO_OBJECT(&info); + if (ma_get_AudioObject_uid(pContext, deviceObjectID, sizeof(info.id.coreaudio), info.id.coreaudio) != MA_SUCCESS) { + continue; + } + if (ma_get_AudioObject_name(pContext, deviceObjectID, sizeof(info.name), info.name) != MA_SUCCESS) { + continue; + } + + if (ma_does_AudioObject_support_playback(pContext, deviceObjectID)) { + if (!callback(pContext, ma_device_type_playback, &info, pUserData)) { + break; + } + } + if (ma_does_AudioObject_support_capture(pContext, deviceObjectID)) { + if (!callback(pContext, ma_device_type_capture, &info, pUserData)) { + break; + } + } + } - if (deviceType == ma_device_type_capture) { - #if defined(MA_APPLE_DESKTOP) - pDevice->coreaudio.deviceObjectIDCapture = (ma_uint32)data.deviceObjectID; - #endif - pDevice->coreaudio.audioUnitCapture = (ma_ptr)data.audioUnit; - pDevice->coreaudio.pAudioBufferList = (ma_ptr)data.pAudioBufferList; - - pDevice->capture.internalFormat = data.formatOut; - pDevice->capture.internalChannels = data.channelsOut; - pDevice->capture.internalSampleRate = data.sampleRateOut; - ma_copy_memory(pDevice->capture.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); - pDevice->capture.internalBufferSizeInFrames = data.bufferSizeInFramesOut; - pDevice->capture.internalPeriods = data.periodsOut; - } else if (deviceType == ma_device_type_playback) { - #if defined(MA_APPLE_DESKTOP) - pDevice->coreaudio.deviceObjectIDPlayback = (ma_uint32)data.deviceObjectID; - #endif - pDevice->coreaudio.audioUnitPlayback = (ma_ptr)data.audioUnit; - - pDevice->playback.internalFormat = data.formatOut; - pDevice->playback.internalChannels = data.channelsOut; - pDevice->playback.internalSampleRate = data.sampleRateOut; - ma_copy_memory(pDevice->playback.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); - pDevice->playback.internalBufferSizeInFrames = data.bufferSizeInFramesOut; - pDevice->playback.internalPeriods = data.periodsOut; + ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks); +#else + /* Only supporting default devices on non-Desktop platforms. */ + ma_device_info info; + + MA_ZERO_OBJECT(&info); + ma_strncpy_s(info.name, sizeof(info.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); + if (!callback(pContext, ma_device_type_playback, &info, pUserData)) { + return MA_SUCCESS; + } + + MA_ZERO_OBJECT(&info); + ma_strncpy_s(info.name, sizeof(info.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); + if (!callback(pContext, ma_device_type_capture, &info, pUserData)) { + return MA_SUCCESS; } +#endif return MA_SUCCESS; } - -ma_result ma_device_init__coreaudio(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) +static ma_result ma_context_get_device_info__coreaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) { ma_result result; - ma_assert(pContext != NULL); - ma_assert(pConfig != NULL); - ma_assert(pDevice != NULL); - - if (pConfig->deviceType == ma_device_type_loopback) { - return MA_DEVICE_TYPE_NOT_SUPPORTED; - } + MA_ASSERT(pContext != NULL); /* No exclusive mode with the Core Audio backend for now. */ - if (((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive) || - ((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive)) { + if (shareMode == ma_share_mode_exclusive) { return MA_SHARE_MODE_NOT_SUPPORTED; } - /* Capture needs to be initialized first. */ - if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { - ma_device_init_internal_data__coreaudio data; - data.formatIn = pConfig->capture.format; - data.channelsIn = pConfig->capture.channels; - data.sampleRateIn = pConfig->sampleRate; - ma_copy_memory(data.channelMapIn, pConfig->capture.channelMap, sizeof(pConfig->capture.channelMap)); - data.usingDefaultFormat = pDevice->capture.usingDefaultFormat; - data.usingDefaultChannels = pDevice->capture.usingDefaultChannels; - data.usingDefaultSampleRate = pDevice->usingDefaultSampleRate; - data.usingDefaultChannelMap = pDevice->capture.usingDefaultChannelMap; - data.shareMode = pConfig->capture.shareMode; - data.bufferSizeInFramesIn = pConfig->bufferSizeInFrames; - data.bufferSizeInMillisecondsIn = pConfig->bufferSizeInMilliseconds; - data.periodsIn = pConfig->periods; - data.registerStopEvent = MA_TRUE; +#if defined(MA_APPLE_DESKTOP) + /* Desktop */ + { + AudioObjectID deviceObjectID; + UInt32 streamDescriptionCount; + AudioStreamRangedDescription* pStreamDescriptions; + UInt32 iStreamDescription; + UInt32 sampleRateRangeCount; + AudioValueRange* pSampleRateRanges; - /* Need at least 3 periods for duplex. */ - if (data.periodsIn < 3 && pConfig->deviceType == ma_device_type_duplex) { - data.periodsIn = 3; - } - - result = ma_device_init_internal__coreaudio(pDevice->pContext, ma_device_type_capture, pConfig->capture.pDeviceID, &data, (void*)pDevice); + result = ma_find_AudioObjectID(pContext, deviceType, pDeviceID, &deviceObjectID); if (result != MA_SUCCESS) { return result; } - - pDevice->coreaudio.isDefaultCaptureDevice = (pConfig->capture.pDeviceID == NULL); - #if defined(MA_APPLE_DESKTOP) - pDevice->coreaudio.deviceObjectIDCapture = (ma_uint32)data.deviceObjectID; - #endif - pDevice->coreaudio.audioUnitCapture = (ma_ptr)data.audioUnit; - pDevice->coreaudio.pAudioBufferList = (ma_ptr)data.pAudioBufferList; - - pDevice->capture.internalFormat = data.formatOut; - pDevice->capture.internalChannels = data.channelsOut; - pDevice->capture.internalSampleRate = data.sampleRateOut; - ma_copy_memory(pDevice->capture.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); - pDevice->capture.internalBufferSizeInFrames = data.bufferSizeInFramesOut; - pDevice->capture.internalPeriods = data.periodsOut; - - #if defined(MA_APPLE_DESKTOP) - /* - If we are using the default device we'll need to listen for changes to the system's default device so we can seemlessly - switch the device in the background. - */ - if (pConfig->capture.pDeviceID == NULL) { - ma_device__track__coreaudio(pDevice); + + result = ma_get_AudioObject_uid(pContext, deviceObjectID, sizeof(pDeviceInfo->id.coreaudio), pDeviceInfo->id.coreaudio); + if (result != MA_SUCCESS) { + return result; } - #endif - } - /* Playback. */ - if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { - ma_device_init_internal_data__coreaudio data; - data.formatIn = pConfig->playback.format; - data.channelsIn = pConfig->playback.channels; - data.sampleRateIn = pConfig->sampleRate; - ma_copy_memory(data.channelMapIn, pConfig->playback.channelMap, sizeof(pConfig->playback.channelMap)); - data.usingDefaultFormat = pDevice->playback.usingDefaultFormat; - data.usingDefaultChannels = pDevice->playback.usingDefaultChannels; - data.usingDefaultSampleRate = pDevice->usingDefaultSampleRate; - data.usingDefaultChannelMap = pDevice->playback.usingDefaultChannelMap; - data.shareMode = pConfig->playback.shareMode; - - /* In full-duplex mode we want the playback buffer to be the same size as the capture buffer. */ - if (pConfig->deviceType == ma_device_type_duplex) { - data.bufferSizeInFramesIn = pDevice->capture.internalBufferSizeInFrames; - data.periodsIn = pDevice->capture.internalPeriods; - data.registerStopEvent = MA_FALSE; - } else { - data.bufferSizeInFramesIn = pConfig->bufferSizeInFrames; - data.bufferSizeInMillisecondsIn = pConfig->bufferSizeInMilliseconds; - data.periodsIn = pConfig->periods; - data.registerStopEvent = MA_TRUE; + result = ma_get_AudioObject_name(pContext, deviceObjectID, sizeof(pDeviceInfo->name), pDeviceInfo->name); + if (result != MA_SUCCESS) { + return result; } - - result = ma_device_init_internal__coreaudio(pDevice->pContext, ma_device_type_playback, pConfig->playback.pDeviceID, &data, (void*)pDevice); + + /* Formats. */ + result = ma_get_AudioObject_stream_descriptions(pContext, deviceObjectID, deviceType, &streamDescriptionCount, &pStreamDescriptions); if (result != MA_SUCCESS) { - if (pConfig->deviceType == ma_device_type_duplex) { - ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitCapture); - if (pDevice->coreaudio.pAudioBufferList) { - ma_free(pDevice->coreaudio.pAudioBufferList); - } - } return result; } + + for (iStreamDescription = 0; iStreamDescription < streamDescriptionCount; ++iStreamDescription) { + ma_format format; + ma_bool32 formatExists = MA_FALSE; + ma_uint32 iOutputFormat; + + result = ma_format_from_AudioStreamBasicDescription(&pStreamDescriptions[iStreamDescription].mFormat, &format); + if (result != MA_SUCCESS) { + continue; + } - pDevice->coreaudio.isDefaultPlaybackDevice = (pConfig->playback.pDeviceID == NULL); - #if defined(MA_APPLE_DESKTOP) - pDevice->coreaudio.deviceObjectIDPlayback = (ma_uint32)data.deviceObjectID; - #endif - pDevice->coreaudio.audioUnitPlayback = (ma_ptr)data.audioUnit; + MA_ASSERT(format != ma_format_unknown); - pDevice->playback.internalFormat = data.formatOut; - pDevice->playback.internalChannels = data.channelsOut; - pDevice->playback.internalSampleRate = data.sampleRateOut; - ma_copy_memory(pDevice->playback.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); - pDevice->playback.internalBufferSizeInFrames = data.bufferSizeInFramesOut; - pDevice->playback.internalPeriods = data.periodsOut; + /* Make sure the format isn't already in the output list. */ + for (iOutputFormat = 0; iOutputFormat < pDeviceInfo->formatCount; ++iOutputFormat) { + if (pDeviceInfo->formats[iOutputFormat] == format) { + formatExists = MA_TRUE; + break; + } + } - #if defined(MA_APPLE_DESKTOP) - /* - If we are using the default device we'll need to listen for changes to the system's default device so we can seemlessly - switch the device in the background. - */ - if (pConfig->playback.pDeviceID == NULL && (pConfig->deviceType != ma_device_type_duplex || pConfig->capture.pDeviceID != NULL)) { - ma_device__track__coreaudio(pDevice); + if (!formatExists) { + pDeviceInfo->formats[pDeviceInfo->formatCount++] = format; + } } - #endif - } - pDevice->coreaudio.originalBufferSizeInFrames = pConfig->bufferSizeInFrames; - pDevice->coreaudio.originalBufferSizeInMilliseconds = pConfig->bufferSizeInMilliseconds; - pDevice->coreaudio.originalPeriods = pConfig->periods; + ma_free(pStreamDescriptions, &pContext->allocationCallbacks); - /* - When stopping the device, a callback is called on another thread. We need to wait for this callback - before returning from ma_device_stop(). This event is used for this. - */ - ma_event_init(pContext, &pDevice->coreaudio.stopEvent); - - /* Need a ring buffer for duplex mode. */ - if (pConfig->deviceType == ma_device_type_duplex) { - ma_uint32 rbSizeInFrames = (ma_uint32)ma_calculate_frame_count_after_src(pDevice->sampleRate, pDevice->capture.internalSampleRate, pDevice->capture.internalBufferSizeInFrames); - ma_result result = ma_pcm_rb_init(pDevice->capture.format, pDevice->capture.channels, rbSizeInFrames, NULL, &pDevice->coreaudio.duplexRB); + + /* Channels. */ + result = ma_get_AudioObject_channel_count(pContext, deviceObjectID, deviceType, &pDeviceInfo->minChannels); if (result != MA_SUCCESS) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[Core Audio] Failed to initialize ring buffer.", result); - } - - /* We need a period to act as a buffer for cases where the playback and capture device's end up desyncing. */ - { - ma_uint32 bufferSizeInFrames = rbSizeInFrames / pDevice->capture.internalPeriods; - void* pBufferData; - ma_pcm_rb_acquire_write(&pDevice->coreaudio.duplexRB, &bufferSizeInFrames, &pBufferData); - { - ma_zero_memory(pBufferData, bufferSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels)); - } - ma_pcm_rb_commit_write(&pDevice->coreaudio.duplexRB, bufferSizeInFrames, pBufferData); + return result; } - } - - return MA_SUCCESS; -} - - -ma_result ma_device_start__coreaudio(ma_device* pDevice) -{ - ma_assert(pDevice != NULL); + pDeviceInfo->maxChannels = pDeviceInfo->minChannels; - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - OSStatus status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitCapture); - if (status != noErr) { - return ma_result_from_OSStatus(status); + + /* Sample rates. */ + result = ma_get_AudioObject_sample_rates(pContext, deviceObjectID, deviceType, &sampleRateRangeCount, &pSampleRateRanges); + if (result != MA_SUCCESS) { + return result; } - } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - OSStatus status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); - if (status != noErr) { - if (pDevice->type == ma_device_type_duplex) { - ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture); + if (sampleRateRangeCount > 0) { + UInt32 iSampleRate; + pDeviceInfo->minSampleRate = UINT32_MAX; + pDeviceInfo->maxSampleRate = 0; + for (iSampleRate = 0; iSampleRate < sampleRateRangeCount; ++iSampleRate) { + if (pDeviceInfo->minSampleRate > pSampleRateRanges[iSampleRate].mMinimum) { + pDeviceInfo->minSampleRate = pSampleRateRanges[iSampleRate].mMinimum; + } + if (pDeviceInfo->maxSampleRate < pSampleRateRanges[iSampleRate].mMaximum) { + pDeviceInfo->maxSampleRate = pSampleRateRanges[iSampleRate].mMaximum; + } } - return ma_result_from_OSStatus(status); } - } + } +#else + /* Mobile */ + { + AudioComponentDescription desc; + AudioComponent component; + AudioUnit audioUnit; + OSStatus status; + AudioUnitScope formatScope; + AudioUnitElement formatElement; + AudioStreamBasicDescription bestFormat; + UInt32 propSize; + + if (deviceType == ma_device_type_playback) { + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); + } else { + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); + } - return MA_SUCCESS; -} - -ma_result ma_device_stop__coreaudio(ma_device* pDevice) -{ - ma_assert(pDevice != NULL); + /* + Retrieving device information is more annoying on mobile than desktop. For simplicity I'm locking this down to whatever format is + reported on a temporary I/O unit. The problem, however, is that this doesn't return a value for the sample rate which we need to + retrieve from the AVAudioSession shared instance. + */ + desc.componentType = kAudioUnitType_Output; + desc.componentSubType = kAudioUnitSubType_RemoteIO; + desc.componentManufacturer = kAudioUnitManufacturer_Apple; + desc.componentFlags = 0; + desc.componentFlagsMask = 0; - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - OSStatus status = ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture); - if (status != noErr) { - return ma_result_from_OSStatus(status); + component = ((ma_AudioComponentFindNext_proc)pContext->coreaudio.AudioComponentFindNext)(NULL, &desc); + if (component == NULL) { + return MA_FAILED_TO_INIT_BACKEND; } - } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - OSStatus status = ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); + status = ((ma_AudioComponentInstanceNew_proc)pContext->coreaudio.AudioComponentInstanceNew)(component, &audioUnit); if (status != noErr) { return ma_result_from_OSStatus(status); } - } - /* We need to wait for the callback to finish before returning. */ - ma_event_wait(&pDevice->coreaudio.stopEvent); - return MA_SUCCESS; -} - - -ma_result ma_context_uninit__coreaudio(ma_context* pContext) -{ - ma_assert(pContext != NULL); - ma_assert(pContext->backend == ma_backend_coreaudio); + formatScope = (deviceType == ma_device_type_playback) ? kAudioUnitScope_Input : kAudioUnitScope_Output; + formatElement = (deviceType == ma_device_type_playback) ? MA_COREAUDIO_OUTPUT_BUS : MA_COREAUDIO_INPUT_BUS; -#if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE) - ma_dlclose(pContext, pContext->coreaudio.hAudioUnit); - ma_dlclose(pContext, pContext->coreaudio.hCoreAudio); - ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation); -#endif - - (void)pContext; - return MA_SUCCESS; -} - -ma_result ma_context_init__coreaudio(const ma_context_config* pConfig, ma_context* pContext) -{ - ma_assert(pContext != NULL); - - (void)pConfig; - -#if defined(MA_APPLE_MOBILE) - @autoreleasepool { - AVAudioSession* pAudioSession = [AVAudioSession sharedInstance]; - ma_assert(pAudioSession != NULL); - - [pAudioSession setCategory: AVAudioSessionCategoryPlayAndRecord error:nil]; - - /* By default we want miniaudio to use the speakers instead of the receiver. In the future this may be customizable. */ - ma_bool32 useSpeakers = MA_TRUE; - if (useSpeakers) { - [pAudioSession overrideOutputAudioPort:AVAudioSessionPortOverrideSpeaker error:nil]; + propSize = sizeof(bestFormat); + status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, &propSize); + if (status != noErr) { + ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(audioUnit); + return ma_result_from_OSStatus(status); } - } -#endif - -#if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE) - pContext->coreaudio.hCoreFoundation = ma_dlopen(pContext, "CoreFoundation.framework/CoreFoundation"); - if (pContext->coreaudio.hCoreFoundation == NULL) { - return MA_API_NOT_FOUND; - } - - pContext->coreaudio.CFStringGetCString = ma_dlsym(pContext, pContext->coreaudio.hCoreFoundation, "CFStringGetCString"); - pContext->coreaudio.CFRelease = ma_dlsym(pContext, pContext->coreaudio.hCoreFoundation, "CFRelease"); + ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(audioUnit); + audioUnit = NULL; - pContext->coreaudio.hCoreAudio = ma_dlopen(pContext, "CoreAudio.framework/CoreAudio"); - if (pContext->coreaudio.hCoreAudio == NULL) { - ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation); - return MA_API_NOT_FOUND; - } - pContext->coreaudio.AudioObjectGetPropertyData = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectGetPropertyData"); - pContext->coreaudio.AudioObjectGetPropertyDataSize = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectGetPropertyDataSize"); - pContext->coreaudio.AudioObjectSetPropertyData = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectSetPropertyData"); - pContext->coreaudio.AudioObjectAddPropertyListener = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectAddPropertyListener"); - pContext->coreaudio.AudioObjectRemovePropertyListener = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectRemovePropertyListener"); - - /* - It looks like Apple has moved some APIs from AudioUnit into AudioToolbox on more recent versions of macOS. They are still - defined in AudioUnit, but just in case they decide to remove them from there entirely I'm going to implement a fallback. - The way it'll work is that it'll first try AudioUnit, and if the required symbols are not present there we'll fall back to - AudioToolbox. - */ - pContext->coreaudio.hAudioUnit = ma_dlopen(pContext, "AudioUnit.framework/AudioUnit"); - if (pContext->coreaudio.hAudioUnit == NULL) { - ma_dlclose(pContext, pContext->coreaudio.hCoreAudio); - ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation); - return MA_API_NOT_FOUND; - } + pDeviceInfo->minChannels = bestFormat.mChannelsPerFrame; + pDeviceInfo->maxChannels = bestFormat.mChannelsPerFrame; - if (ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentFindNext") == NULL) { - /* Couldn't find the required symbols in AudioUnit, so fall back to AudioToolbox. */ - ma_dlclose(pContext, pContext->coreaudio.hAudioUnit); - pContext->coreaudio.hAudioUnit = ma_dlopen(pContext, "AudioToolbox.framework/AudioToolbox"); - if (pContext->coreaudio.hAudioUnit == NULL) { - ma_dlclose(pContext, pContext->coreaudio.hCoreAudio); - ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation); - return MA_API_NOT_FOUND; + pDeviceInfo->formatCount = 1; + result = ma_format_from_AudioStreamBasicDescription(&bestFormat, &pDeviceInfo->formats[0]); + if (result != MA_SUCCESS) { + return result; } - } - - pContext->coreaudio.AudioComponentFindNext = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentFindNext"); - pContext->coreaudio.AudioComponentInstanceDispose = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentInstanceDispose"); - pContext->coreaudio.AudioComponentInstanceNew = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentInstanceNew"); - pContext->coreaudio.AudioOutputUnitStart = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioOutputUnitStart"); - pContext->coreaudio.AudioOutputUnitStop = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioOutputUnitStop"); - pContext->coreaudio.AudioUnitAddPropertyListener = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitAddPropertyListener"); - pContext->coreaudio.AudioUnitGetPropertyInfo = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitGetPropertyInfo"); - pContext->coreaudio.AudioUnitGetProperty = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitGetProperty"); - pContext->coreaudio.AudioUnitSetProperty = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitSetProperty"); - pContext->coreaudio.AudioUnitInitialize = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitInitialize"); - pContext->coreaudio.AudioUnitRender = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitRender"); -#else - pContext->coreaudio.CFStringGetCString = (ma_proc)CFStringGetCString; - pContext->coreaudio.CFRelease = (ma_proc)CFRelease; - - #if defined(MA_APPLE_DESKTOP) - pContext->coreaudio.AudioObjectGetPropertyData = (ma_proc)AudioObjectGetPropertyData; - pContext->coreaudio.AudioObjectGetPropertyDataSize = (ma_proc)AudioObjectGetPropertyDataSize; - pContext->coreaudio.AudioObjectSetPropertyData = (ma_proc)AudioObjectSetPropertyData; - pContext->coreaudio.AudioObjectAddPropertyListener = (ma_proc)AudioObjectAddPropertyListener; - pContext->coreaudio.AudioObjectRemovePropertyListener = (ma_proc)AudioObjectRemovePropertyListener; - #endif - pContext->coreaudio.AudioComponentFindNext = (ma_proc)AudioComponentFindNext; - pContext->coreaudio.AudioComponentInstanceDispose = (ma_proc)AudioComponentInstanceDispose; - pContext->coreaudio.AudioComponentInstanceNew = (ma_proc)AudioComponentInstanceNew; - pContext->coreaudio.AudioOutputUnitStart = (ma_proc)AudioOutputUnitStart; - pContext->coreaudio.AudioOutputUnitStop = (ma_proc)AudioOutputUnitStop; - pContext->coreaudio.AudioUnitAddPropertyListener = (ma_proc)AudioUnitAddPropertyListener; - pContext->coreaudio.AudioUnitGetPropertyInfo = (ma_proc)AudioUnitGetPropertyInfo; - pContext->coreaudio.AudioUnitGetProperty = (ma_proc)AudioUnitGetProperty; - pContext->coreaudio.AudioUnitSetProperty = (ma_proc)AudioUnitSetProperty; - pContext->coreaudio.AudioUnitInitialize = (ma_proc)AudioUnitInitialize; - pContext->coreaudio.AudioUnitRender = (ma_proc)AudioUnitRender; -#endif + /* + It looks like Apple are wanting to push the whole AVAudioSession thing. Thus, we need to use that to determine device settings. To do + this we just get the shared instance and inspect. + */ + @autoreleasepool { + AVAudioSession* pAudioSession = [AVAudioSession sharedInstance]; + MA_ASSERT(pAudioSession != NULL); - pContext->isBackendAsynchronous = MA_TRUE; - - pContext->onUninit = ma_context_uninit__coreaudio; - pContext->onDeviceIDEqual = ma_context_is_device_id_equal__coreaudio; - pContext->onEnumDevices = ma_context_enumerate_devices__coreaudio; - pContext->onGetDeviceInfo = ma_context_get_device_info__coreaudio; - pContext->onDeviceInit = ma_device_init__coreaudio; - pContext->onDeviceUninit = ma_device_uninit__coreaudio; - pContext->onDeviceStart = ma_device_start__coreaudio; - pContext->onDeviceStop = ma_device_stop__coreaudio; - - /* Audio component. */ - { - AudioComponentDescription desc; - desc.componentType = kAudioUnitType_Output; - #if defined(MA_APPLE_DESKTOP) - desc.componentSubType = kAudioUnitSubType_HALOutput; - #else - desc.componentSubType = kAudioUnitSubType_RemoteIO; - #endif - desc.componentManufacturer = kAudioUnitManufacturer_Apple; - desc.componentFlags = 0; - desc.componentFlagsMask = 0; - - pContext->coreaudio.component = ((ma_AudioComponentFindNext_proc)pContext->coreaudio.AudioComponentFindNext)(NULL, &desc); - if (pContext->coreaudio.component == NULL) { - #if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE) - ma_dlclose(pContext, pContext->coreaudio.hAudioUnit); - ma_dlclose(pContext, pContext->coreaudio.hCoreAudio); - ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation); - #endif - return MA_FAILED_TO_INIT_BACKEND; + pDeviceInfo->minSampleRate = (ma_uint32)pAudioSession.sampleRate; + pDeviceInfo->maxSampleRate = pDeviceInfo->minSampleRate; } } - - return MA_SUCCESS; -} -#endif /* Core Audio */ - - - -/****************************************************************************** - -sndio Backend - -******************************************************************************/ -#ifdef MA_HAS_SNDIO -#include -#include - -/* -Only supporting OpenBSD. This did not work very well at all on FreeBSD when I tried it. Not sure if this is due -to miniaudio's implementation or if it's some kind of system configuration issue, but basically the default device -just doesn't emit any sound, or at times you'll hear tiny pieces. I will consider enabling this when there's -demand for it or if I can get it tested and debugged more thoroughly. -*/ -#if 0 -#if defined(__NetBSD__) || defined(__OpenBSD__) -#include -#endif -#if defined(__FreeBSD__) || defined(__DragonFly__) -#include -#endif #endif + + (void)pDeviceInfo; /* Unused. */ + return MA_SUCCESS; +} -#define MA_SIO_DEVANY "default" -#define MA_SIO_PLAY 1 -#define MA_SIO_REC 2 -#define MA_SIO_NENC 8 -#define MA_SIO_NCHAN 8 -#define MA_SIO_NRATE 16 -#define MA_SIO_NCONF 4 - -struct ma_sio_hdl; /* <-- Opaque */ - -struct ma_sio_par -{ - unsigned int bits; - unsigned int bps; - unsigned int sig; - unsigned int le; - unsigned int msb; - unsigned int rchan; - unsigned int pchan; - unsigned int rate; - unsigned int bufsz; - unsigned int xrun; - unsigned int round; - unsigned int appbufsz; - int __pad[3]; - unsigned int __magic; -}; - -struct ma_sio_enc -{ - unsigned int bits; - unsigned int bps; - unsigned int sig; - unsigned int le; - unsigned int msb; -}; -struct ma_sio_conf +static OSStatus ma_on_output__coreaudio(void* pUserData, AudioUnitRenderActionFlags* pActionFlags, const AudioTimeStamp* pTimeStamp, UInt32 busNumber, UInt32 frameCount, AudioBufferList* pBufferList) { - unsigned int enc; - unsigned int rchan; - unsigned int pchan; - unsigned int rate; -}; + ma_device* pDevice = (ma_device*)pUserData; + ma_stream_layout layout; -struct ma_sio_cap -{ - struct ma_sio_enc enc[MA_SIO_NENC]; - unsigned int rchan[MA_SIO_NCHAN]; - unsigned int pchan[MA_SIO_NCHAN]; - unsigned int rate[MA_SIO_NRATE]; - int __pad[7]; - unsigned int nconf; - struct ma_sio_conf confs[MA_SIO_NCONF]; -}; + MA_ASSERT(pDevice != NULL); -typedef struct ma_sio_hdl* (* ma_sio_open_proc) (const char*, unsigned int, int); -typedef void (* ma_sio_close_proc) (struct ma_sio_hdl*); -typedef int (* ma_sio_setpar_proc) (struct ma_sio_hdl*, struct ma_sio_par*); -typedef int (* ma_sio_getpar_proc) (struct ma_sio_hdl*, struct ma_sio_par*); -typedef int (* ma_sio_getcap_proc) (struct ma_sio_hdl*, struct ma_sio_cap*); -typedef size_t (* ma_sio_write_proc) (struct ma_sio_hdl*, const void*, size_t); -typedef size_t (* ma_sio_read_proc) (struct ma_sio_hdl*, void*, size_t); -typedef int (* ma_sio_start_proc) (struct ma_sio_hdl*); -typedef int (* ma_sio_stop_proc) (struct ma_sio_hdl*); -typedef int (* ma_sio_initpar_proc)(struct ma_sio_par*); +#if defined(MA_DEBUG_OUTPUT) + printf("INFO: Output Callback: busNumber=%d, frameCount=%d, mNumberBuffers=%d\n", busNumber, frameCount, pBufferList->mNumberBuffers); +#endif -ma_format ma_format_from_sio_enc__sndio(unsigned int bits, unsigned int bps, unsigned int sig, unsigned int le, unsigned int msb) -{ - /* We only support native-endian right now. */ - if ((ma_is_little_endian() && le == 0) || (ma_is_big_endian() && le == 1)) { - return ma_format_unknown; + /* We need to check whether or not we are outputting interleaved or non-interleaved samples. The way we do this is slightly different for each type. */ + layout = ma_stream_layout_interleaved; + if (pBufferList->mBuffers[0].mNumberChannels != pDevice->playback.internalChannels) { + layout = ma_stream_layout_deinterleaved; } - if (bits == 8 && bps == 1 && sig == 0) { - return ma_format_u8; - } - if (bits == 16 && bps == 2 && sig == 1) { - return ma_format_s16; - } - if (bits == 24 && bps == 3 && sig == 1) { - return ma_format_s24; - } - if (bits == 24 && bps == 4 && sig == 1 && msb == 0) { - /*return ma_format_s24_32;*/ - } - if (bits == 32 && bps == 4 && sig == 1) { - return ma_format_s32; + if (layout == ma_stream_layout_interleaved) { + /* For now we can assume everything is interleaved. */ + UInt32 iBuffer; + for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; ++iBuffer) { + if (pBufferList->mBuffers[iBuffer].mNumberChannels == pDevice->playback.internalChannels) { + ma_uint32 frameCountForThisBuffer = pBufferList->mBuffers[iBuffer].mDataByteSize / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + if (frameCountForThisBuffer > 0) { + if (pDevice->type == ma_device_type_duplex) { + ma_device__handle_duplex_callback_playback(pDevice, frameCountForThisBuffer, pBufferList->mBuffers[iBuffer].mData, &pDevice->coreaudio.duplexRB); + } else { + ma_device__read_frames_from_client(pDevice, frameCountForThisBuffer, pBufferList->mBuffers[iBuffer].mData); + } + } + + #if defined(MA_DEBUG_OUTPUT) + printf(" frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", frameCount, pBufferList->mBuffers[iBuffer].mNumberChannels, pBufferList->mBuffers[iBuffer].mDataByteSize); + #endif + } else { + /* + This case is where the number of channels in the output buffer do not match our internal channels. It could mean that it's + not interleaved, in which case we can't handle right now since miniaudio does not yet support non-interleaved streams. We just + output silence here. + */ + MA_ZERO_MEMORY(pBufferList->mBuffers[iBuffer].mData, pBufferList->mBuffers[iBuffer].mDataByteSize); + + #if defined(MA_DEBUG_OUTPUT) + printf(" WARNING: Outputting silence. frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", frameCount, pBufferList->mBuffers[iBuffer].mNumberChannels, pBufferList->mBuffers[iBuffer].mDataByteSize); + #endif + } + } + } else { + /* This is the deinterleaved case. We need to update each buffer in groups of internalChannels. This assumes each buffer is the same size. */ + + /* + For safety we'll check that the internal channels is a multiple of the buffer count. If it's not it means something + very strange has happened and we're not going to support it. + */ + if ((pBufferList->mNumberBuffers % pDevice->playback.internalChannels) == 0) { + ma_uint8 tempBuffer[4096]; + UInt32 iBuffer; + + for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; iBuffer += pDevice->playback.internalChannels) { + ma_uint32 frameCountPerBuffer = pBufferList->mBuffers[iBuffer].mDataByteSize / ma_get_bytes_per_sample(pDevice->playback.internalFormat); + ma_uint32 framesRemaining = frameCountPerBuffer; + + while (framesRemaining > 0) { + void* ppDeinterleavedBuffers[MA_MAX_CHANNELS]; + ma_uint32 iChannel; + ma_uint32 framesToRead = sizeof(tempBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + if (framesToRead > framesRemaining) { + framesToRead = framesRemaining; + } + + if (pDevice->type == ma_device_type_duplex) { + ma_device__handle_duplex_callback_playback(pDevice, framesToRead, tempBuffer, &pDevice->coreaudio.duplexRB); + } else { + ma_device__read_frames_from_client(pDevice, framesToRead, tempBuffer); + } + + for (iChannel = 0; iChannel < pDevice->playback.internalChannels; ++iChannel) { + ppDeinterleavedBuffers[iChannel] = (void*)ma_offset_ptr(pBufferList->mBuffers[iBuffer+iChannel].mData, (frameCountPerBuffer - framesRemaining) * ma_get_bytes_per_sample(pDevice->playback.internalFormat)); + } + + ma_deinterleave_pcm_frames(pDevice->playback.internalFormat, pDevice->playback.internalChannels, framesToRead, tempBuffer, ppDeinterleavedBuffers); + + framesRemaining -= framesToRead; + } + } + } } - return ma_format_unknown; + (void)pActionFlags; + (void)pTimeStamp; + (void)busNumber; + + return noErr; } -ma_format ma_find_best_format_from_sio_cap__sndio(struct ma_sio_cap* caps) +static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFlags* pActionFlags, const AudioTimeStamp* pTimeStamp, UInt32 busNumber, UInt32 frameCount, AudioBufferList* pUnusedBufferList) { - ma_format bestFormat; - unsigned int iConfig; + ma_device* pDevice = (ma_device*)pUserData; + AudioBufferList* pRenderedBufferList; + ma_stream_layout layout; + OSStatus status; - ma_assert(caps != NULL); + MA_ASSERT(pDevice != NULL); - bestFormat = ma_format_unknown; - for (iConfig = 0; iConfig < caps->nconf; iConfig += 1) { - unsigned int iEncoding; - for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) { - unsigned int bits; - unsigned int bps; - unsigned int sig; - unsigned int le; - unsigned int msb; - ma_format format; + pRenderedBufferList = (AudioBufferList*)pDevice->coreaudio.pAudioBufferList; + MA_ASSERT(pRenderedBufferList); + + /* We need to check whether or not we are outputting interleaved or non-interleaved samples. The way we do this is slightly different for each type. */ + layout = ma_stream_layout_interleaved; + if (pRenderedBufferList->mBuffers[0].mNumberChannels != pDevice->capture.internalChannels) { + layout = ma_stream_layout_deinterleaved; + } + +#if defined(MA_DEBUG_OUTPUT) + printf("INFO: Input Callback: busNumber=%d, frameCount=%d, mNumberBuffers=%d\n", busNumber, frameCount, pRenderedBufferList->mNumberBuffers); +#endif + + status = ((ma_AudioUnitRender_proc)pDevice->pContext->coreaudio.AudioUnitRender)((AudioUnit)pDevice->coreaudio.audioUnitCapture, pActionFlags, pTimeStamp, busNumber, frameCount, pRenderedBufferList); + if (status != noErr) { + #if defined(MA_DEBUG_OUTPUT) + printf(" ERROR: AudioUnitRender() failed with %d\n", status); + #endif + return status; + } + + if (layout == ma_stream_layout_interleaved) { + UInt32 iBuffer; + for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; ++iBuffer) { + if (pRenderedBufferList->mBuffers[iBuffer].mNumberChannels == pDevice->capture.internalChannels) { + if (pDevice->type == ma_device_type_duplex) { + ma_device__handle_duplex_callback_capture(pDevice, frameCount, pRenderedBufferList->mBuffers[iBuffer].mData, &pDevice->coreaudio.duplexRB); + } else { + ma_device__send_frames_to_client(pDevice, frameCount, pRenderedBufferList->mBuffers[iBuffer].mData); + } + #if defined(MA_DEBUG_OUTPUT) + printf(" mDataByteSize=%d\n", pRenderedBufferList->mBuffers[iBuffer].mDataByteSize); + #endif + } else { + /* + This case is where the number of channels in the output buffer do not match our internal channels. It could mean that it's + not interleaved, in which case we can't handle right now since miniaudio does not yet support non-interleaved streams. + */ + ma_uint8 silentBuffer[4096]; + ma_uint32 framesRemaining; + + MA_ZERO_MEMORY(silentBuffer, sizeof(silentBuffer)); + + framesRemaining = frameCount; + while (framesRemaining > 0) { + ma_uint32 framesToSend = sizeof(silentBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + if (framesToSend > framesRemaining) { + framesToSend = framesRemaining; + } + + if (pDevice->type == ma_device_type_duplex) { + ma_device__handle_duplex_callback_capture(pDevice, framesToSend, silentBuffer, &pDevice->coreaudio.duplexRB); + } else { + ma_device__send_frames_to_client(pDevice, framesToSend, silentBuffer); + } + + framesRemaining -= framesToSend; + } + + #if defined(MA_DEBUG_OUTPUT) + printf(" WARNING: Outputting silence. frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", frameCount, pRenderedBufferList->mBuffers[iBuffer].mNumberChannels, pRenderedBufferList->mBuffers[iBuffer].mDataByteSize); + #endif + } + } + } else { + /* This is the deinterleaved case. We need to interleave the audio data before sending it to the client. This assumes each buffer is the same size. */ + + /* + For safety we'll check that the internal channels is a multiple of the buffer count. If it's not it means something + very strange has happened and we're not going to support it. + */ + if ((pRenderedBufferList->mNumberBuffers % pDevice->capture.internalChannels) == 0) { + ma_uint8 tempBuffer[4096]; + UInt32 iBuffer; + for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; iBuffer += pDevice->capture.internalChannels) { + ma_uint32 framesRemaining = frameCount; + while (framesRemaining > 0) { + void* ppDeinterleavedBuffers[MA_MAX_CHANNELS]; + ma_uint32 iChannel; + ma_uint32 framesToSend = sizeof(tempBuffer) / ma_get_bytes_per_sample(pDevice->capture.internalFormat); + if (framesToSend > framesRemaining) { + framesToSend = framesRemaining; + } + + for (iChannel = 0; iChannel < pDevice->capture.internalChannels; ++iChannel) { + ppDeinterleavedBuffers[iChannel] = (void*)ma_offset_ptr(pRenderedBufferList->mBuffers[iBuffer+iChannel].mData, (frameCount - framesRemaining) * ma_get_bytes_per_sample(pDevice->capture.internalFormat)); + } + + ma_interleave_pcm_frames(pDevice->capture.internalFormat, pDevice->capture.internalChannels, framesToSend, (const void**)ppDeinterleavedBuffers, tempBuffer); - if ((caps->confs[iConfig].enc & (1UL << iEncoding)) == 0) { - continue; + if (pDevice->type == ma_device_type_duplex) { + ma_device__handle_duplex_callback_capture(pDevice, framesToSend, tempBuffer, &pDevice->coreaudio.duplexRB); + } else { + ma_device__send_frames_to_client(pDevice, framesToSend, tempBuffer); + } + + framesRemaining -= framesToSend; + } } + } + } + + (void)pActionFlags; + (void)pTimeStamp; + (void)busNumber; + (void)frameCount; + (void)pUnusedBufferList; + + return noErr; +} + +static void on_start_stop__coreaudio(void* pUserData, AudioUnit audioUnit, AudioUnitPropertyID propertyID, AudioUnitScope scope, AudioUnitElement element) +{ + ma_device* pDevice = (ma_device*)pUserData; + MA_ASSERT(pDevice != NULL); + + /* + There's been a report of a deadlock here when triggered by ma_device_uninit(). It looks like + AudioUnitGetProprty (called below) and AudioComponentInstanceDispose (called in ma_device_uninit) + can try waiting on the same lock. I'm going to try working around this by not calling any Core + Audio APIs in the callback when the device has been stopped or uninitialized. + */ + if (ma_device__get_state(pDevice) == MA_STATE_UNINITIALIZED || ma_device__get_state(pDevice) == MA_STATE_STOPPING || ma_device__get_state(pDevice) == MA_STATE_STOPPED) { + ma_stop_proc onStop = pDevice->onStop; + if (onStop) { + onStop(pDevice); + } + + ma_event_signal(&pDevice->coreaudio.stopEvent); + } else { + UInt32 isRunning; + UInt32 isRunningSize = sizeof(isRunning); + OSStatus status = ((ma_AudioUnitGetProperty_proc)pDevice->pContext->coreaudio.AudioUnitGetProperty)(audioUnit, kAudioOutputUnitProperty_IsRunning, scope, element, &isRunning, &isRunningSize); + if (status != noErr) { + return; /* Don't really know what to do in this case... just ignore it, I suppose... */ + } + + if (!isRunning) { + ma_stop_proc onStop; + + /* + The stop event is a bit annoying in Core Audio because it will be called when we automatically switch the default device. Some scenarios to consider: - bits = caps->enc[iEncoding].bits; - bps = caps->enc[iEncoding].bps; - sig = caps->enc[iEncoding].sig; - le = caps->enc[iEncoding].le; - msb = caps->enc[iEncoding].msb; - format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb); - if (format == ma_format_unknown) { - continue; /* Format not supported. */ - } + 1) When the device is unplugged, this will be called _before_ the default device change notification. + 2) When the device is changed via the default device change notification, this will be called _after_ the switch. - if (bestFormat == ma_format_unknown) { - bestFormat = format; - } else { - if (ma_get_format_priority_index(bestFormat) > ma_get_format_priority_index(format)) { /* <-- Lower = better. */ - bestFormat = format; + For case #1, we just check if there's a new default device available. If so, we just ignore the stop event. For case #2 we check a flag. + */ + if (((audioUnit == pDevice->coreaudio.audioUnitPlayback) && pDevice->coreaudio.isDefaultPlaybackDevice) || + ((audioUnit == pDevice->coreaudio.audioUnitCapture) && pDevice->coreaudio.isDefaultCaptureDevice)) { + /* + It looks like the device is switching through an external event, such as the user unplugging the device or changing the default device + via the operating system's sound settings. If we're re-initializing the device, we just terminate because we want the stopping of the + device to be seamless to the client (we don't want them receiving the onStop event and thinking that the device has stopped when it + hasn't!). + */ + if (((audioUnit == pDevice->coreaudio.audioUnitPlayback) && pDevice->coreaudio.isSwitchingPlaybackDevice) || + ((audioUnit == pDevice->coreaudio.audioUnitCapture) && pDevice->coreaudio.isSwitchingCaptureDevice)) { + return; } + + /* + Getting here means the device is not reinitializing which means it may have been unplugged. From what I can see, it looks like Core Audio + will try switching to the new default device seamlessly. We need to somehow find a way to determine whether or not Core Audio will most + likely be successful in switching to the new device. + + TODO: Try to predict if Core Audio will switch devices. If not, the onStop callback needs to be posted. + */ + return; + } + + /* Getting here means we need to stop the device. */ + onStop = pDevice->onStop; + if (onStop) { + onStop(pDevice); } } } - - return ma_format_unknown; + + (void)propertyID; /* Unused. */ } -ma_uint32 ma_find_best_channels_from_sio_cap__sndio(struct ma_sio_cap* caps, ma_device_type deviceType, ma_format requiredFormat) -{ - ma_uint32 maxChannels; - unsigned int iConfig; +#if defined(MA_APPLE_DESKTOP) +static ma_uint32 g_DeviceTrackingInitCounter_CoreAudio = 0; +static ma_mutex g_DeviceTrackingMutex_CoreAudio; +static ma_device** g_ppTrackedDevices_CoreAudio = NULL; +static ma_uint32 g_TrackedDeviceCap_CoreAudio = 0; +static ma_uint32 g_TrackedDeviceCount_CoreAudio = 0; - ma_assert(caps != NULL); - ma_assert(requiredFormat != ma_format_unknown); +static OSStatus ma_default_device_changed__coreaudio(AudioObjectID objectID, UInt32 addressCount, const AudioObjectPropertyAddress* pAddresses, void* pUserData) +{ + ma_device_type deviceType; - /* Just pick whatever configuration has the most channels. */ - maxChannels = 0; - for (iConfig = 0; iConfig < caps->nconf; iConfig += 1) { - /* The encoding should be of requiredFormat. */ - unsigned int iEncoding; - for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) { - unsigned int iChannel; - unsigned int bits; - unsigned int bps; - unsigned int sig; - unsigned int le; - unsigned int msb; - ma_format format; - - if ((caps->confs[iConfig].enc & (1UL << iEncoding)) == 0) { - continue; - } - - bits = caps->enc[iEncoding].bits; - bps = caps->enc[iEncoding].bps; - sig = caps->enc[iEncoding].sig; - le = caps->enc[iEncoding].le; - msb = caps->enc[iEncoding].msb; - format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb); - if (format != requiredFormat) { - continue; - } - - /* Getting here means the format is supported. Iterate over each channel count and grab the biggest one. */ - for (iChannel = 0; iChannel < MA_SIO_NCHAN; iChannel += 1) { - unsigned int chan = 0; - unsigned int channels; - - if (deviceType == ma_device_type_playback) { - chan = caps->confs[iConfig].pchan; - } else { - chan = caps->confs[iConfig].rchan; - } + /* Not sure if I really need to check this, but it makes me feel better. */ + if (addressCount == 0) { + return noErr; + } + + if (pAddresses[0].mSelector == kAudioHardwarePropertyDefaultOutputDevice) { + deviceType = ma_device_type_playback; + } else if (pAddresses[0].mSelector == kAudioHardwarePropertyDefaultInputDevice) { + deviceType = ma_device_type_capture; + } else { + return noErr; /* Should never hit this. */ + } + + ma_mutex_lock(&g_DeviceTrackingMutex_CoreAudio); + { + ma_uint32 iDevice; + for (iDevice = 0; iDevice < g_TrackedDeviceCount_CoreAudio; iDevice += 1) { + ma_result reinitResult; + ma_device* pDevice; - if ((chan & (1UL << iChannel)) == 0) { - continue; - } - + pDevice = g_ppTrackedDevices_CoreAudio[iDevice]; + if (pDevice->type == deviceType || pDevice->type == ma_device_type_duplex) { if (deviceType == ma_device_type_playback) { - channels = caps->pchan[iChannel]; + pDevice->coreaudio.isSwitchingPlaybackDevice = MA_TRUE; + reinitResult = ma_device_reinit_internal__coreaudio(pDevice, deviceType, MA_TRUE); + pDevice->coreaudio.isSwitchingPlaybackDevice = MA_FALSE; } else { - channels = caps->rchan[iChannel]; + pDevice->coreaudio.isSwitchingCaptureDevice = MA_TRUE; + reinitResult = ma_device_reinit_internal__coreaudio(pDevice, deviceType, MA_TRUE); + pDevice->coreaudio.isSwitchingCaptureDevice = MA_FALSE; } - if (maxChannels < channels) { - maxChannels = channels; + if (reinitResult == MA_SUCCESS) { + ma_device__post_init_setup(pDevice, deviceType); + + /* Restart the device if required. If this fails we need to stop the device entirely. */ + if (ma_device__get_state(pDevice) == MA_STATE_STARTED) { + OSStatus status; + if (deviceType == ma_device_type_playback) { + status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); + if (status != noErr) { + if (pDevice->type == ma_device_type_duplex) { + ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture); + } + ma_device__set_state(pDevice, MA_STATE_STOPPED); + } + } else if (deviceType == ma_device_type_capture) { + status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitCapture); + if (status != noErr) { + if (pDevice->type == ma_device_type_duplex) { + ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); + } + ma_device__set_state(pDevice, MA_STATE_STOPPED); + } + } + } } } } } + ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio); - return maxChannels; + (void)objectID; /* Unused. */ + return noErr; } -ma_uint32 ma_find_best_sample_rate_from_sio_cap__sndio(struct ma_sio_cap* caps, ma_device_type deviceType, ma_format requiredFormat, ma_uint32 requiredChannels) +static ma_result ma_context__init_device_tracking__coreaudio(ma_context* pContext) { - ma_uint32 firstSampleRate; - ma_uint32 bestSampleRate; - unsigned int iConfig; + MA_ASSERT(pContext != NULL); + + if (ma_atomic_increment_32(&g_DeviceTrackingInitCounter_CoreAudio) == 1) { + AudioObjectPropertyAddress propAddress; + propAddress.mScope = kAudioObjectPropertyScopeGlobal; + propAddress.mElement = kAudioObjectPropertyElementMaster; + + ma_mutex_init(pContext, &g_DeviceTrackingMutex_CoreAudio); + + propAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice; + ((ma_AudioObjectAddPropertyListener_proc)pContext->coreaudio.AudioObjectAddPropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL); + + propAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice; + ((ma_AudioObjectAddPropertyListener_proc)pContext->coreaudio.AudioObjectAddPropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL); + } + + return MA_SUCCESS; +} - ma_assert(caps != NULL); - ma_assert(requiredFormat != ma_format_unknown); - ma_assert(requiredChannels > 0); - ma_assert(requiredChannels <= MA_MAX_CHANNELS); +static ma_result ma_context__uninit_device_tracking__coreaudio(ma_context* pContext) +{ + MA_ASSERT(pContext != NULL); - firstSampleRate = 0; /* <-- If the device does not support a standard rate we'll fall back to the first one that's found. */ - bestSampleRate = 0; + if (ma_atomic_decrement_32(&g_DeviceTrackingInitCounter_CoreAudio) == 0) { + AudioObjectPropertyAddress propAddress; + propAddress.mScope = kAudioObjectPropertyScopeGlobal; + propAddress.mElement = kAudioObjectPropertyElementMaster; + + propAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice; + ((ma_AudioObjectRemovePropertyListener_proc)pContext->coreaudio.AudioObjectRemovePropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL); + + propAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice; + ((ma_AudioObjectRemovePropertyListener_proc)pContext->coreaudio.AudioObjectRemovePropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL); + + /* At this point there should be no tracked devices. If so there's an error somewhere. */ + MA_ASSERT(g_ppTrackedDevices_CoreAudio == NULL); + MA_ASSERT(g_TrackedDeviceCount_CoreAudio == 0); + + ma_mutex_uninit(&g_DeviceTrackingMutex_CoreAudio); + } + + return MA_SUCCESS; +} - for (iConfig = 0; iConfig < caps->nconf; iConfig += 1) { - /* The encoding should be of requiredFormat. */ - unsigned int iEncoding; - for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) { - unsigned int iChannel; - unsigned int bits; - unsigned int bps; - unsigned int sig; - unsigned int le; - unsigned int msb; - ma_format format; +static ma_result ma_device__track__coreaudio(ma_device* pDevice) +{ + ma_result result; - if ((caps->confs[iConfig].enc & (1UL << iEncoding)) == 0) { - continue; - } + MA_ASSERT(pDevice != NULL); + + result = ma_context__init_device_tracking__coreaudio(pDevice->pContext); + if (result != MA_SUCCESS) { + return result; + } + + ma_mutex_lock(&g_DeviceTrackingMutex_CoreAudio); + { + /* Allocate memory if required. */ + if (g_TrackedDeviceCap_CoreAudio <= g_TrackedDeviceCount_CoreAudio) { + ma_uint32 oldCap; + ma_uint32 newCap; + ma_device** ppNewDevices; - bits = caps->enc[iEncoding].bits; - bps = caps->enc[iEncoding].bps; - sig = caps->enc[iEncoding].sig; - le = caps->enc[iEncoding].le; - msb = caps->enc[iEncoding].msb; - format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb); - if (format != requiredFormat) { - continue; + oldCap = g_TrackedDeviceCap_CoreAudio; + newCap = g_TrackedDeviceCap_CoreAudio * 2; + if (newCap == 0) { + newCap = 1; } - /* Getting here means the format is supported. Iterate over each channel count and grab the biggest one. */ - for (iChannel = 0; iChannel < MA_SIO_NCHAN; iChannel += 1) { - unsigned int chan = 0; - unsigned int channels; - unsigned int iRate; - - if (deviceType == ma_device_type_playback) { - chan = caps->confs[iConfig].pchan; - } else { - chan = caps->confs[iConfig].rchan; - } + ppNewDevices = (ma_device**)ma__realloc_from_callbacks(g_ppTrackedDevices_CoreAudio, sizeof(*g_ppTrackedDevices_CoreAudio)*newCap, sizeof(*g_ppTrackedDevices_CoreAudio)*oldCap, &pDevice->pContext->allocationCallbacks); + if (ppNewDevices == NULL) { + ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio); + return MA_OUT_OF_MEMORY; + } - if ((chan & (1UL << iChannel)) == 0) { - continue; - } - - if (deviceType == ma_device_type_playback) { - channels = caps->pchan[iChannel]; - } else { - channels = caps->rchan[iChannel]; - } - - if (channels != requiredChannels) { - continue; + g_ppTrackedDevices_CoreAudio = ppNewDevices; + g_TrackedDeviceCap_CoreAudio = newCap; + } + + g_ppTrackedDevices_CoreAudio[g_TrackedDeviceCount_CoreAudio] = pDevice; + g_TrackedDeviceCount_CoreAudio += 1; + } + ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio); + + return MA_SUCCESS; +} + +static ma_result ma_device__untrack__coreaudio(ma_device* pDevice) +{ + ma_result result; + + MA_ASSERT(pDevice != NULL); + + ma_mutex_lock(&g_DeviceTrackingMutex_CoreAudio); + { + ma_uint32 iDevice; + for (iDevice = 0; iDevice < g_TrackedDeviceCount_CoreAudio; iDevice += 1) { + if (g_ppTrackedDevices_CoreAudio[iDevice] == pDevice) { + /* We've found the device. We now need to remove it from the list. */ + ma_uint32 jDevice; + for (jDevice = iDevice; jDevice < g_TrackedDeviceCount_CoreAudio-1; jDevice += 1) { + g_ppTrackedDevices_CoreAudio[jDevice] = g_ppTrackedDevices_CoreAudio[jDevice+1]; } - /* Getting here means we have found a compatible encoding/channel pair. */ - for (iRate = 0; iRate < MA_SIO_NRATE; iRate += 1) { - ma_uint32 rate = (ma_uint32)caps->rate[iRate]; - ma_uint32 ratePriority; + g_TrackedDeviceCount_CoreAudio -= 1; - if (firstSampleRate == 0) { - firstSampleRate = rate; - } - - /* Disregard this rate if it's not a standard one. */ - ratePriority = ma_get_standard_sample_rate_priority_index(rate); - if (ratePriority == (ma_uint32)-1) { - continue; - } - - if (ma_get_standard_sample_rate_priority_index(bestSampleRate) > ratePriority) { /* Lower = better. */ - bestSampleRate = rate; - } + /* If there's nothing else in the list we need to free memory. */ + if (g_TrackedDeviceCount_CoreAudio == 0) { + ma__free_from_callbacks(g_ppTrackedDevices_CoreAudio, &pDevice->pContext->allocationCallbacks); + g_ppTrackedDevices_CoreAudio = NULL; + g_TrackedDeviceCap_CoreAudio = 0; } + + break; } } } - - /* If a standard sample rate was not found just fall back to the first one that was iterated. */ - if (bestSampleRate == 0) { - bestSampleRate = firstSampleRate; + ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio); + + result = ma_context__uninit_device_tracking__coreaudio(pDevice->pContext); + if (result != MA_SUCCESS) { + return result; } - return bestSampleRate; + return MA_SUCCESS; } +#endif +#if defined(MA_APPLE_MOBILE) +@interface ma_router_change_handler:NSObject { + ma_device* m_pDevice; +} +@end -ma_bool32 ma_context_is_device_id_equal__sndio(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) +@implementation ma_router_change_handler +-(id)init:(ma_device*)pDevice { - ma_assert(pContext != NULL); - ma_assert(pID0 != NULL); - ma_assert(pID1 != NULL); - (void)pContext; + self = [super init]; + m_pDevice = pDevice; - return ma_strcmp(pID0->sndio, pID1->sndio) == 0; + [[NSNotificationCenter defaultCenter] addObserver:self selector:@selector(handle_route_change:) name:AVAudioSessionRouteChangeNotification object:[AVAudioSession sharedInstance]]; + + return self; } -ma_result ma_context_enumerate_devices__sndio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +-(void)dealloc { - ma_bool32 isTerminating = MA_FALSE; - struct ma_sio_hdl* handle; + [self remove_handler]; +} + +-(void)remove_handler +{ + [[NSNotificationCenter defaultCenter] removeObserver:self name:@"AVAudioSessionRouteChangeNotification" object:nil]; +} + +-(void)handle_route_change:(NSNotification*)pNotification +{ + AVAudioSession* pSession = [AVAudioSession sharedInstance]; + + NSInteger reason = [[[pNotification userInfo] objectForKey:AVAudioSessionRouteChangeReasonKey] integerValue]; + switch (reason) + { + case AVAudioSessionRouteChangeReasonOldDeviceUnavailable: + { + #if defined(MA_DEBUG_OUTPUT) + printf("[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonOldDeviceUnavailable\n"); + #endif + } break; + + case AVAudioSessionRouteChangeReasonNewDeviceAvailable: + { + #if defined(MA_DEBUG_OUTPUT) + printf("[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonNewDeviceAvailable\n"); + #endif + } break; + + case AVAudioSessionRouteChangeReasonNoSuitableRouteForCategory: + { + #if defined(MA_DEBUG_OUTPUT) + printf("[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonNoSuitableRouteForCategory\n"); + #endif + } break; + + case AVAudioSessionRouteChangeReasonWakeFromSleep: + { + #if defined(MA_DEBUG_OUTPUT) + printf("[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonWakeFromSleep\n"); + #endif + } break; + + case AVAudioSessionRouteChangeReasonOverride: + { + #if defined(MA_DEBUG_OUTPUT) + printf("[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonOverride\n"); + #endif + } break; + + case AVAudioSessionRouteChangeReasonCategoryChange: + { + #if defined(MA_DEBUG_OUTPUT) + printf("[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonCategoryChange\n"); + #endif + } break; + + case AVAudioSessionRouteChangeReasonUnknown: + default: + { + #if defined(MA_DEBUG_OUTPUT) + printf("[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonUnknown\n"); + #endif + } break; + } + + m_pDevice->sampleRate = (ma_uint32)pSession.sampleRate; - ma_assert(pContext != NULL); - ma_assert(callback != NULL); + if (m_pDevice->type == ma_device_type_capture || m_pDevice->type == ma_device_type_duplex) { + m_pDevice->capture.channels = (ma_uint32)pSession.inputNumberOfChannels; + ma_device__post_init_setup(m_pDevice, ma_device_type_capture); + } + if (m_pDevice->type == ma_device_type_playback || m_pDevice->type == ma_device_type_duplex) { + m_pDevice->playback.channels = (ma_uint32)pSession.outputNumberOfChannels; + ma_device__post_init_setup(m_pDevice, ma_device_type_playback); + } +} +@end +#endif + +static void ma_device_uninit__coreaudio(ma_device* pDevice) +{ + MA_ASSERT(pDevice != NULL); + MA_ASSERT(ma_device__get_state(pDevice) == MA_STATE_UNINITIALIZED); - /* sndio doesn't seem to have a good device enumeration API, so I'm therefore only enumerating over default devices for now. */ +#if defined(MA_APPLE_DESKTOP) + /* + Make sure we're no longer tracking the device. It doesn't matter if we call this for a non-default device because it'll + just gracefully ignore it. + */ + ma_device__untrack__coreaudio(pDevice); +#endif +#if defined(MA_APPLE_MOBILE) + if (pDevice->coreaudio.pRouteChangeHandler != NULL) { + ma_router_change_handler* pRouteChangeHandler = (__bridge_transfer ma_router_change_handler*)pDevice->coreaudio.pRouteChangeHandler; + [pRouteChangeHandler remove_handler]; + } +#endif - /* Playback. */ - if (!isTerminating) { - handle = ((ma_sio_open_proc)pContext->sndio.sio_open)(MA_SIO_DEVANY, MA_SIO_PLAY, 0); - if (handle != NULL) { - /* Supports playback. */ - ma_device_info deviceInfo; - ma_zero_object(&deviceInfo); - ma_strcpy_s(deviceInfo.id.sndio, sizeof(deviceInfo.id.sndio), MA_SIO_DEVANY); - ma_strcpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME); - - isTerminating = !callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); - - ((ma_sio_close_proc)pContext->sndio.sio_close)(handle); - } + if (pDevice->coreaudio.audioUnitCapture != NULL) { + ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitCapture); + } + if (pDevice->coreaudio.audioUnitPlayback != NULL) { + ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); } - /* Capture. */ - if (!isTerminating) { - handle = ((ma_sio_open_proc)pContext->sndio.sio_open)(MA_SIO_DEVANY, MA_SIO_REC, 0); - if (handle != NULL) { - /* Supports capture. */ - ma_device_info deviceInfo; - ma_zero_object(&deviceInfo); - ma_strcpy_s(deviceInfo.id.sndio, sizeof(deviceInfo.id.sndio), "default"); - ma_strcpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME); + if (pDevice->coreaudio.pAudioBufferList) { + ma__free_from_callbacks(pDevice->coreaudio.pAudioBufferList, &pDevice->pContext->allocationCallbacks); + } - isTerminating = !callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); - - ((ma_sio_close_proc)pContext->sndio.sio_close)(handle); - } + if (pDevice->type == ma_device_type_duplex) { + ma_pcm_rb_uninit(&pDevice->coreaudio.duplexRB); + } +} + +typedef struct +{ + /* Input. */ + ma_format formatIn; + ma_uint32 channelsIn; + ma_uint32 sampleRateIn; + ma_channel channelMapIn[MA_MAX_CHANNELS]; + ma_uint32 periodSizeInFramesIn; + ma_uint32 periodSizeInMillisecondsIn; + ma_uint32 periodsIn; + ma_bool32 usingDefaultFormat; + ma_bool32 usingDefaultChannels; + ma_bool32 usingDefaultSampleRate; + ma_bool32 usingDefaultChannelMap; + ma_share_mode shareMode; + ma_bool32 registerStopEvent; + + /* Output. */ +#if defined(MA_APPLE_DESKTOP) + AudioObjectID deviceObjectID; +#endif + AudioComponent component; + AudioUnit audioUnit; + AudioBufferList* pAudioBufferList; /* Only used for input devices. */ + ma_format formatOut; + ma_uint32 channelsOut; + ma_uint32 sampleRateOut; + ma_channel channelMapOut[MA_MAX_CHANNELS]; + ma_uint32 periodSizeInFramesOut; + ma_uint32 periodsOut; + char deviceName[256]; +} ma_device_init_internal_data__coreaudio; + +static ma_result ma_device_init_internal__coreaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_init_internal_data__coreaudio* pData, void* pDevice_DoNotReference) /* <-- pDevice is typed as void* intentionally so as to avoid accidentally referencing it. */ +{ + ma_result result; + OSStatus status; + UInt32 enableIOFlag; + AudioStreamBasicDescription bestFormat; + ma_uint32 actualPeriodSizeInFrames; + AURenderCallbackStruct callbackInfo; +#if defined(MA_APPLE_DESKTOP) + AudioObjectID deviceObjectID; +#endif + + /* This API should only be used for a single device type: playback or capture. No full-duplex mode. */ + if (deviceType == ma_device_type_duplex) { + return MA_INVALID_ARGS; + } + + MA_ASSERT(pContext != NULL); + MA_ASSERT(deviceType == ma_device_type_playback || deviceType == ma_device_type_capture); + +#if defined(MA_APPLE_DESKTOP) + pData->deviceObjectID = 0; +#endif + pData->component = NULL; + pData->audioUnit = NULL; + pData->pAudioBufferList = NULL; + +#if defined(MA_APPLE_DESKTOP) + result = ma_find_AudioObjectID(pContext, deviceType, pDeviceID, &deviceObjectID); + if (result != MA_SUCCESS) { + return result; + } + + pData->deviceObjectID = deviceObjectID; +#endif + + /* Core audio doesn't really use the notion of a period so we can leave this unmodified, but not too over the top. */ + pData->periodsOut = pData->periodsIn; + if (pData->periodsOut == 0) { + pData->periodsOut = MA_DEFAULT_PERIODS; + } + if (pData->periodsOut > 16) { + pData->periodsOut = 16; + } + + + /* Audio unit. */ + status = ((ma_AudioComponentInstanceNew_proc)pContext->coreaudio.AudioComponentInstanceNew)((AudioComponent)pContext->coreaudio.component, (AudioUnit*)&pData->audioUnit); + if (status != noErr) { + return ma_result_from_OSStatus(status); + } + + + /* The input/output buses need to be explicitly enabled and disabled. We set the flag based on the output unit first, then we just swap it for input. */ + enableIOFlag = 1; + if (deviceType == ma_device_type_capture) { + enableIOFlag = 0; } - return MA_SUCCESS; -} - -ma_result ma_context_get_device_info__sndio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) -{ - char devid[256]; - struct ma_sio_hdl* handle; - struct ma_sio_cap caps; - unsigned int iConfig; - - ma_assert(pContext != NULL); - (void)shareMode; - - /* We need to open the device before we can get information about it. */ - if (pDeviceID == NULL) { - ma_strcpy_s(devid, sizeof(devid), MA_SIO_DEVANY); - ma_strcpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), (deviceType == ma_device_type_playback) ? MA_DEFAULT_PLAYBACK_DEVICE_NAME : MA_DEFAULT_CAPTURE_DEVICE_NAME); - } else { - ma_strcpy_s(devid, sizeof(devid), pDeviceID->sndio); - ma_strcpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), devid); + status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, MA_COREAUDIO_OUTPUT_BUS, &enableIOFlag, sizeof(enableIOFlag)); + if (status != noErr) { + ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); + return ma_result_from_OSStatus(status); } - handle = ((ma_sio_open_proc)pContext->sndio.sio_open)(devid, (deviceType == ma_device_type_playback) ? MA_SIO_PLAY : MA_SIO_REC, 0); - if (handle == NULL) { - return MA_NO_DEVICE; + enableIOFlag = (enableIOFlag == 0) ? 1 : 0; + status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, MA_COREAUDIO_INPUT_BUS, &enableIOFlag, sizeof(enableIOFlag)); + if (status != noErr) { + ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); + return ma_result_from_OSStatus(status); } - if (((ma_sio_getcap_proc)pContext->sndio.sio_getcap)(handle, &caps) == 0) { - return MA_ERROR; + + /* Set the device to use with this audio unit. This is only used on desktop since we are using defaults on mobile. */ +#if defined(MA_APPLE_DESKTOP) + status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, (deviceType == ma_device_type_playback) ? MA_COREAUDIO_OUTPUT_BUS : MA_COREAUDIO_INPUT_BUS, &deviceObjectID, sizeof(AudioDeviceID)); + if (status != noErr) { + ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); + return ma_result_from_OSStatus(result); } +#endif - for (iConfig = 0; iConfig < caps.nconf; iConfig += 1) { - /* - The main thing we care about is that the encoding is supported by miniaudio. If it is, we want to give - preference to some formats over others. - */ - unsigned int iEncoding; - unsigned int iChannel; - unsigned int iRate; + /* + Format. This is the hardest part of initialization because there's a few variables to take into account. + 1) The format must be supported by the device. + 2) The format must be supported miniaudio. + 3) There's a priority that miniaudio prefers. + + Ideally we would like to use a format that's as close to the hardware as possible so we can get as close to a passthrough as possible. The + most important property is the sample rate. miniaudio can do format conversion for any sample rate and channel count, but cannot do the same + for the sample data format. If the sample data format is not supported by miniaudio it must be ignored completely. + + On mobile platforms this is a bit different. We just force the use of whatever the audio unit's current format is set to. + */ + { + AudioUnitScope formatScope = (deviceType == ma_device_type_playback) ? kAudioUnitScope_Input : kAudioUnitScope_Output; + AudioUnitElement formatElement = (deviceType == ma_device_type_playback) ? MA_COREAUDIO_OUTPUT_BUS : MA_COREAUDIO_INPUT_BUS; - for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) { - unsigned int bits; - unsigned int bps; - unsigned int sig; - unsigned int le; - unsigned int msb; - ma_format format; - ma_bool32 formatExists = MA_FALSE; - ma_uint32 iExistingFormat; + #if defined(MA_APPLE_DESKTOP) + AudioStreamBasicDescription origFormat; + UInt32 origFormatSize; - if ((caps.confs[iConfig].enc & (1UL << iEncoding)) == 0) { - continue; - } - - bits = caps.enc[iEncoding].bits; - bps = caps.enc[iEncoding].bps; - sig = caps.enc[iEncoding].sig; - le = caps.enc[iEncoding].le; - msb = caps.enc[iEncoding].msb; - format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb); - if (format == ma_format_unknown) { - continue; /* Format not supported. */ - } - - /* Add this format if it doesn't already exist. */ - for (iExistingFormat = 0; iExistingFormat < pDeviceInfo->formatCount; iExistingFormat += 1) { - if (pDeviceInfo->formats[iExistingFormat] == format) { - formatExists = MA_TRUE; - break; - } - } - - if (!formatExists) { - pDeviceInfo->formats[pDeviceInfo->formatCount++] = format; - } + result = ma_find_best_format__coreaudio(pContext, deviceObjectID, deviceType, pData->formatIn, pData->channelsIn, pData->sampleRateIn, pData->usingDefaultFormat, pData->usingDefaultChannels, pData->usingDefaultSampleRate, &bestFormat); + if (result != MA_SUCCESS) { + ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); + return result; } - /* Channels. */ - for (iChannel = 0; iChannel < MA_SIO_NCHAN; iChannel += 1) { - unsigned int chan = 0; - unsigned int channels; - - if (deviceType == ma_device_type_playback) { - chan = caps.confs[iConfig].pchan; - } else { - chan = caps.confs[iConfig].rchan; - } + /* From what I can see, Apple's documentation implies that we should keep the sample rate consistent. */ + origFormatSize = sizeof(origFormat); + if (deviceType == ma_device_type_playback) { + status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, MA_COREAUDIO_OUTPUT_BUS, &origFormat, &origFormatSize); + } else { + status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, MA_COREAUDIO_INPUT_BUS, &origFormat, &origFormatSize); + } - if ((chan & (1UL << iChannel)) == 0) { - continue; - } + if (status != noErr) { + ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); + return result; + } + + bestFormat.mSampleRate = origFormat.mSampleRate; + + status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, sizeof(bestFormat)); + if (status != noErr) { + /* We failed to set the format, so fall back to the current format of the audio unit. */ + bestFormat = origFormat; + } + #else + UInt32 propSize = sizeof(bestFormat); + status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, &propSize); + if (status != noErr) { + ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); + return ma_result_from_OSStatus(status); + } + + /* + Sample rate is a little different here because for some reason kAudioUnitProperty_StreamFormat returns 0... Oh well. We need to instead try + setting the sample rate to what the user has requested and then just see the results of it. Need to use some Objective-C here for this since + it depends on Apple's AVAudioSession API. To do this we just get the shared AVAudioSession instance and then set it. Note that from what I + can tell, it looks like the sample rate is shared between playback and capture for everything. + */ + @autoreleasepool { + AVAudioSession* pAudioSession = [AVAudioSession sharedInstance]; + MA_ASSERT(pAudioSession != NULL); + [pAudioSession setPreferredSampleRate:(double)pData->sampleRateIn error:nil]; + bestFormat.mSampleRate = pAudioSession.sampleRate; + + /* + I've had a report that the channel count returned by AudioUnitGetProperty above is inconsistent with + AVAudioSession outputNumberOfChannels. I'm going to try using the AVAudioSession values instead. + */ if (deviceType == ma_device_type_playback) { - channels = caps.pchan[iChannel]; - } else { - channels = caps.rchan[iChannel]; - } - - if (pDeviceInfo->minChannels > channels) { - pDeviceInfo->minChannels = channels; + bestFormat.mChannelsPerFrame = (UInt32)pAudioSession.outputNumberOfChannels; } - if (pDeviceInfo->maxChannels < channels) { - pDeviceInfo->maxChannels = channels; + if (deviceType == ma_device_type_capture) { + bestFormat.mChannelsPerFrame = (UInt32)pAudioSession.inputNumberOfChannels; } } - /* Sample rates. */ - for (iRate = 0; iRate < MA_SIO_NRATE; iRate += 1) { - if ((caps.confs[iConfig].rate & (1UL << iRate)) != 0) { - unsigned int rate = caps.rate[iRate]; - if (pDeviceInfo->minSampleRate > rate) { - pDeviceInfo->minSampleRate = rate; - } - if (pDeviceInfo->maxSampleRate < rate) { - pDeviceInfo->maxSampleRate = rate; - } - } + status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, sizeof(bestFormat)); + if (status != noErr) { + ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); + return ma_result_from_OSStatus(status); } + #endif + + result = ma_format_from_AudioStreamBasicDescription(&bestFormat, &pData->formatOut); + if (result != MA_SUCCESS) { + ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); + return result; + } + + if (pData->formatOut == ma_format_unknown) { + ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); + return MA_FORMAT_NOT_SUPPORTED; + } + + pData->channelsOut = bestFormat.mChannelsPerFrame; + pData->sampleRateOut = bestFormat.mSampleRate; } - - ((ma_sio_close_proc)pContext->sndio.sio_close)(handle); - return MA_SUCCESS; -} - -void ma_device_uninit__sndio(ma_device* pDevice) -{ - ma_assert(pDevice != NULL); - - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)pDevice->sndio.handleCapture); - } - - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback); - } -} - -ma_result ma_device_init_handle__sndio(ma_context* pContext, const ma_device_config* pConfig, ma_device_type deviceType, ma_device* pDevice) -{ - const char* pDeviceName; - ma_ptr handle; - int openFlags = 0; - struct ma_sio_cap caps; - struct ma_sio_par par; - ma_device_id* pDeviceID; - ma_format format; - ma_uint32 channels; - ma_uint32 sampleRate; - ma_format internalFormat; - ma_uint32 internalChannels; - ma_uint32 internalSampleRate; - ma_uint32 internalBufferSizeInFrames; - ma_uint32 internalPeriods; - - ma_assert(pContext != NULL); - ma_assert(pConfig != NULL); - ma_assert(deviceType != ma_device_type_duplex); - ma_assert(pDevice != NULL); - - if (deviceType == ma_device_type_capture) { - openFlags = MA_SIO_REC; - pDeviceID = pConfig->capture.pDeviceID; - format = pConfig->capture.format; - channels = pConfig->capture.channels; - sampleRate = pConfig->sampleRate; - } else { - openFlags = MA_SIO_PLAY; - pDeviceID = pConfig->playback.pDeviceID; - format = pConfig->playback.format; - channels = pConfig->playback.channels; - sampleRate = pConfig->sampleRate; + + /* + Internal channel map. This is weird in my testing. If I use the AudioObject to get the + channel map, the channel descriptions are set to "Unknown" for some reason. To work around + this it looks like retrieving it from the AudioUnit will work. However, and this is where + it gets weird, it doesn't seem to work with capture devices, nor at all on iOS... Therefore + I'm going to fall back to a default assumption in these cases. + */ +#if defined(MA_APPLE_DESKTOP) + result = ma_get_AudioUnit_channel_map(pContext, pData->audioUnit, deviceType, pData->channelMapOut); + if (result != MA_SUCCESS) { + #if 0 + /* Try falling back to the channel map from the AudioObject. */ + result = ma_get_AudioObject_channel_map(pContext, deviceObjectID, deviceType, pData->channelMapOut); + if (result != MA_SUCCESS) { + return result; + } + #else + /* Fall back to default assumptions. */ + ma_get_standard_channel_map(ma_standard_channel_map_default, pData->channelsOut, pData->channelMapOut); + #endif } +#else + /* TODO: Figure out how to get the channel map using AVAudioSession. */ + ma_get_standard_channel_map(ma_standard_channel_map_default, pData->channelsOut, pData->channelMapOut); +#endif + - pDeviceName = MA_SIO_DEVANY; - if (pDeviceID != NULL) { - pDeviceName = pDeviceID->sndio; + /* Buffer size. Not allowing this to be configurable on iOS. */ + actualPeriodSizeInFrames = pData->periodSizeInFramesIn; + +#if defined(MA_APPLE_DESKTOP) + if (actualPeriodSizeInFrames == 0) { + actualPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pData->periodSizeInMillisecondsIn, pData->sampleRateOut); } - - handle = (ma_ptr)((ma_sio_open_proc)pContext->sndio.sio_open)(pDeviceName, openFlags, 0); - if (handle == NULL) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[sndio] Failed to open device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + + result = ma_set_AudioObject_buffer_size_in_frames(pContext, deviceObjectID, deviceType, &actualPeriodSizeInFrames); + if (result != MA_SUCCESS) { + return result; } + + pData->periodSizeInFramesOut = actualPeriodSizeInFrames; +#else + actualPeriodSizeInFrames = 2048; + pData->periodSizeInFramesOut = actualPeriodSizeInFrames; +#endif - /* We need to retrieve the device caps to determine the most appropriate format to use. */ - if (((ma_sio_getcap_proc)pContext->sndio.sio_getcap)((struct ma_sio_hdl*)handle, &caps) == 0) { - ((ma_sio_close_proc)pContext->sndio.sio_close)((struct ma_sio_hdl*)handle); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[sndio] Failed to retrieve device caps.", MA_ERROR); - } /* - Note: sndio reports a huge range of available channels. This is inconvenient for us because there's no real - way, as far as I can tell, to get the _actual_ channel count of the device. I'm therefore restricting this - to the requested channels, regardless of whether or not the default channel count is requested. + During testing I discovered that the buffer size can be too big. You'll get an error like this: - For hardware devices, I'm suspecting only a single channel count will be reported and we can safely use the - value returned by ma_find_best_channels_from_sio_cap__sndio(). + kAudioUnitErr_TooManyFramesToProcess : inFramesToProcess=4096, mMaxFramesPerSlice=512 + + Note how inFramesToProcess is smaller than mMaxFramesPerSlice. To fix, we need to set kAudioUnitProperty_MaximumFramesPerSlice to that + of the size of our buffer, or do it the other way around and set our buffer size to the kAudioUnitProperty_MaximumFramesPerSlice. */ - if (deviceType == ma_device_type_capture) { - if (pDevice->capture.usingDefaultFormat) { - format = ma_find_best_format_from_sio_cap__sndio(&caps); - } - if (pDevice->capture.usingDefaultChannels) { - if (strlen(pDeviceName) > strlen("rsnd/") && strncmp(pDeviceName, "rsnd/", strlen("rsnd/")) == 0) { - channels = ma_find_best_channels_from_sio_cap__sndio(&caps, deviceType, format); - } - } - } else { - if (pDevice->playback.usingDefaultFormat) { - format = ma_find_best_format_from_sio_cap__sndio(&caps); - } - if (pDevice->playback.usingDefaultChannels) { - if (strlen(pDeviceName) > strlen("rsnd/") && strncmp(pDeviceName, "rsnd/", strlen("rsnd/")) == 0) { - channels = ma_find_best_channels_from_sio_cap__sndio(&caps, deviceType, format); - } + { + /*AudioUnitScope propScope = (deviceType == ma_device_type_playback) ? kAudioUnitScope_Input : kAudioUnitScope_Output; + AudioUnitElement propBus = (deviceType == ma_device_type_playback) ? MA_COREAUDIO_OUTPUT_BUS : MA_COREAUDIO_INPUT_BUS; + + status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, propScope, propBus, &actualBufferSizeInFrames, sizeof(actualBufferSizeInFrames)); + if (status != noErr) { + ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); + return ma_result_from_OSStatus(status); + }*/ + + status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, 0, &actualPeriodSizeInFrames, sizeof(actualPeriodSizeInFrames)); + if (status != noErr) { + ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); + return ma_result_from_OSStatus(status); } } - if (pDevice->usingDefaultSampleRate) { - sampleRate = ma_find_best_sample_rate_from_sio_cap__sndio(&caps, pConfig->deviceType, format, channels); - } - + /* We need a buffer list if this is an input device. We render into this in the input callback. */ + if (deviceType == ma_device_type_capture) { + ma_bool32 isInterleaved = (bestFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved) == 0; + size_t allocationSize; + AudioBufferList* pBufferList; - ((ma_sio_initpar_proc)pDevice->pContext->sndio.sio_initpar)(&par); - par.msb = 0; - par.le = ma_is_little_endian(); - - switch (format) { - case ma_format_u8: - { - par.bits = 8; - par.bps = 1; - par.sig = 0; - } break; + allocationSize = sizeof(AudioBufferList) - sizeof(AudioBuffer); /* Subtract sizeof(AudioBuffer) because that part is dynamically sized. */ + if (isInterleaved) { + /* Interleaved case. This is the simple case because we just have one buffer. */ + allocationSize += sizeof(AudioBuffer) * 1; + allocationSize += actualPeriodSizeInFrames * ma_get_bytes_per_frame(pData->formatOut, pData->channelsOut); + } else { + /* Non-interleaved case. This is the more complex case because there's more than one buffer. */ + allocationSize += sizeof(AudioBuffer) * pData->channelsOut; + allocationSize += actualPeriodSizeInFrames * ma_get_bytes_per_sample(pData->formatOut) * pData->channelsOut; + } - case ma_format_s24: - { - par.bits = 24; - par.bps = 3; - par.sig = 1; - } break; + pBufferList = (AudioBufferList*)ma__malloc_from_callbacks(allocationSize, &pContext->allocationCallbacks); + if (pBufferList == NULL) { + ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); + return MA_OUT_OF_MEMORY; + } - case ma_format_s32: - { - par.bits = 32; - par.bps = 4; - par.sig = 1; - } break; + if (isInterleaved) { + pBufferList->mNumberBuffers = 1; + pBufferList->mBuffers[0].mNumberChannels = pData->channelsOut; + pBufferList->mBuffers[0].mDataByteSize = actualPeriodSizeInFrames * ma_get_bytes_per_frame(pData->formatOut, pData->channelsOut); + pBufferList->mBuffers[0].mData = (ma_uint8*)pBufferList + sizeof(AudioBufferList); + } else { + ma_uint32 iBuffer; + pBufferList->mNumberBuffers = pData->channelsOut; + for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; ++iBuffer) { + pBufferList->mBuffers[iBuffer].mNumberChannels = 1; + pBufferList->mBuffers[iBuffer].mDataByteSize = actualPeriodSizeInFrames * ma_get_bytes_per_sample(pData->formatOut); + pBufferList->mBuffers[iBuffer].mData = (ma_uint8*)pBufferList + ((sizeof(AudioBufferList) - sizeof(AudioBuffer)) + (sizeof(AudioBuffer) * pData->channelsOut)) + (actualPeriodSizeInFrames * ma_get_bytes_per_sample(pData->formatOut) * iBuffer); + } + } - case ma_format_s16: - case ma_format_f32: - default: - { - par.bits = 16; - par.bps = 2; - par.sig = 1; - } break; - } - - if (deviceType == ma_device_type_capture) { - par.rchan = channels; - } else { - par.pchan = channels; - } - - par.rate = sampleRate; - - internalBufferSizeInFrames = pConfig->bufferSizeInFrames; - if (internalBufferSizeInFrames == 0) { - internalBufferSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->bufferSizeInMilliseconds, par.rate); + pData->pAudioBufferList = pBufferList; } - - par.round = internalBufferSizeInFrames / pConfig->periods; - par.appbufsz = par.round * pConfig->periods; - if (((ma_sio_setpar_proc)pContext->sndio.sio_setpar)((struct ma_sio_hdl*)handle, &par) == 0) { - ((ma_sio_close_proc)pContext->sndio.sio_close)((struct ma_sio_hdl*)handle); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[sndio] Failed to set buffer size.", MA_FORMAT_NOT_SUPPORTED); - } - if (((ma_sio_getpar_proc)pContext->sndio.sio_getpar)((struct ma_sio_hdl*)handle, &par) == 0) { - ((ma_sio_close_proc)pContext->sndio.sio_close)((struct ma_sio_hdl*)handle); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[sndio] Failed to retrieve buffer size.", MA_FORMAT_NOT_SUPPORTED); - } - - internalFormat = ma_format_from_sio_enc__sndio(par.bits, par.bps, par.sig, par.le, par.msb); - internalChannels = (deviceType == ma_device_type_capture) ? par.rchan : par.pchan; - internalSampleRate = par.rate; - internalPeriods = par.appbufsz / par.round; - internalBufferSizeInFrames = par.appbufsz; - - if (deviceType == ma_device_type_capture) { - pDevice->sndio.handleCapture = handle; - pDevice->capture.internalFormat = internalFormat; - pDevice->capture.internalChannels = internalChannels; - pDevice->capture.internalSampleRate = internalSampleRate; - ma_get_standard_channel_map(ma_standard_channel_map_sndio, pDevice->capture.internalChannels, pDevice->capture.internalChannelMap); - pDevice->capture.internalBufferSizeInFrames = internalBufferSizeInFrames; - pDevice->capture.internalPeriods = internalPeriods; + /* Callbacks. */ + callbackInfo.inputProcRefCon = pDevice_DoNotReference; + if (deviceType == ma_device_type_playback) { + callbackInfo.inputProc = ma_on_output__coreaudio; + status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Global, MA_COREAUDIO_OUTPUT_BUS, &callbackInfo, sizeof(callbackInfo)); + if (status != noErr) { + ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); + return ma_result_from_OSStatus(status); + } } else { - pDevice->sndio.handlePlayback = handle; - pDevice->playback.internalFormat = internalFormat; - pDevice->playback.internalChannels = internalChannels; - pDevice->playback.internalSampleRate = internalSampleRate; - ma_get_standard_channel_map(ma_standard_channel_map_sndio, pDevice->playback.internalChannels, pDevice->playback.internalChannelMap); - pDevice->playback.internalBufferSizeInFrames = internalBufferSizeInFrames; - pDevice->playback.internalPeriods = internalPeriods; - } - -#ifdef MA_DEBUG_OUTPUT - printf("DEVICE INFO\n"); - printf(" Format: %s\n", ma_get_format_name(internalFormat)); - printf(" Channels: %d\n", internalChannels); - printf(" Sample Rate: %d\n", internalSampleRate); - printf(" Buffer Size: %d\n", internalBufferSizeInFrames); - printf(" Periods: %d\n", internalPeriods); - printf(" appbufsz: %d\n", par.appbufsz); - printf(" round: %d\n", par.round); -#endif - - return MA_SUCCESS; -} - -ma_result ma_device_init__sndio(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) -{ - ma_assert(pDevice != NULL); - - ma_zero_object(&pDevice->sndio); - - if (pConfig->deviceType == ma_device_type_loopback) { - return MA_DEVICE_TYPE_NOT_SUPPORTED; - } - - if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { - ma_result result = ma_device_init_handle__sndio(pContext, pConfig, ma_device_type_capture, pDevice); - if (result != MA_SUCCESS) { - return result; + callbackInfo.inputProc = ma_on_input__coreaudio; + status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, MA_COREAUDIO_INPUT_BUS, &callbackInfo, sizeof(callbackInfo)); + if (status != noErr) { + ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); + return ma_result_from_OSStatus(status); } } - - if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { - ma_result result = ma_device_init_handle__sndio(pContext, pConfig, ma_device_type_playback, pDevice); - if (result != MA_SUCCESS) { - return result; + + /* We need to listen for stop events. */ + if (pData->registerStopEvent) { + status = ((ma_AudioUnitAddPropertyListener_proc)pContext->coreaudio.AudioUnitAddPropertyListener)(pData->audioUnit, kAudioOutputUnitProperty_IsRunning, on_start_stop__coreaudio, pDevice_DoNotReference); + if (status != noErr) { + ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); + return ma_result_from_OSStatus(status); } } - - return MA_SUCCESS; -} - -ma_result ma_device_stop__sndio(ma_device* pDevice) -{ - ma_assert(pDevice != NULL); - - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - ((ma_sio_stop_proc)pDevice->pContext->sndio.sio_stop)((struct ma_sio_hdl*)pDevice->sndio.handleCapture); + + /* Initialize the audio unit. */ + status = ((ma_AudioUnitInitialize_proc)pContext->coreaudio.AudioUnitInitialize)(pData->audioUnit); + if (status != noErr) { + ma__free_from_callbacks(pData->pAudioBufferList, &pContext->allocationCallbacks); + pData->pAudioBufferList = NULL; + ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit); + return ma_result_from_OSStatus(status); } - - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - ((ma_sio_stop_proc)pDevice->pContext->sndio.sio_stop)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback); + + /* Grab the name. */ +#if defined(MA_APPLE_DESKTOP) + ma_get_AudioObject_name(pContext, deviceObjectID, sizeof(pData->deviceName), pData->deviceName); +#else + if (deviceType == ma_device_type_playback) { + ma_strcpy_s(pData->deviceName, sizeof(pData->deviceName), MA_DEFAULT_PLAYBACK_DEVICE_NAME); + } else { + ma_strcpy_s(pData->deviceName, sizeof(pData->deviceName), MA_DEFAULT_CAPTURE_DEVICE_NAME); } - - return MA_SUCCESS; +#endif + + return result; } -ma_result ma_device_write__sndio(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) +static ma_result ma_device_reinit_internal__coreaudio(ma_device* pDevice, ma_device_type deviceType, ma_bool32 disposePreviousAudioUnit) { - int result; + ma_device_init_internal_data__coreaudio data; + ma_result result; - if (pFramesWritten != NULL) { - *pFramesWritten = 0; + /* This should only be called for playback or capture, not duplex. */ + if (deviceType == ma_device_type_duplex) { + return MA_INVALID_ARGS; } - result = ((ma_sio_write_proc)pDevice->pContext->sndio.sio_write)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); - if (result == 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[sndio] Failed to send data from the client to the device.", MA_FAILED_TO_SEND_DATA_TO_DEVICE); + if (deviceType == ma_device_type_capture) { + data.formatIn = pDevice->capture.format; + data.channelsIn = pDevice->capture.channels; + data.sampleRateIn = pDevice->sampleRate; + MA_COPY_MEMORY(data.channelMapIn, pDevice->capture.channelMap, sizeof(pDevice->capture.channelMap)); + data.usingDefaultFormat = pDevice->capture.usingDefaultFormat; + data.usingDefaultChannels = pDevice->capture.usingDefaultChannels; + data.usingDefaultSampleRate = pDevice->usingDefaultSampleRate; + data.usingDefaultChannelMap = pDevice->capture.usingDefaultChannelMap; + data.shareMode = pDevice->capture.shareMode; + data.registerStopEvent = MA_TRUE; + + if (disposePreviousAudioUnit) { + ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture); + ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitCapture); + } + if (pDevice->coreaudio.pAudioBufferList) { + ma__free_from_callbacks(pDevice->coreaudio.pAudioBufferList, &pDevice->pContext->allocationCallbacks); + } + } else if (deviceType == ma_device_type_playback) { + data.formatIn = pDevice->playback.format; + data.channelsIn = pDevice->playback.channels; + data.sampleRateIn = pDevice->sampleRate; + MA_COPY_MEMORY(data.channelMapIn, pDevice->playback.channelMap, sizeof(pDevice->playback.channelMap)); + data.usingDefaultFormat = pDevice->playback.usingDefaultFormat; + data.usingDefaultChannels = pDevice->playback.usingDefaultChannels; + data.usingDefaultSampleRate = pDevice->usingDefaultSampleRate; + data.usingDefaultChannelMap = pDevice->playback.usingDefaultChannelMap; + data.shareMode = pDevice->playback.shareMode; + data.registerStopEvent = (pDevice->type != ma_device_type_duplex); + + if (disposePreviousAudioUnit) { + ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); + ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); + } } + data.periodSizeInFramesIn = pDevice->coreaudio.originalPeriodSizeInFrames; + data.periodSizeInMillisecondsIn = pDevice->coreaudio.originalPeriodSizeInMilliseconds; + data.periodsIn = pDevice->coreaudio.originalPeriods; - if (pFramesWritten != NULL) { - *pFramesWritten = frameCount; + /* Need at least 3 periods for duplex. */ + if (data.periodsIn < 3 && pDevice->type == ma_device_type_duplex) { + data.periodsIn = 3; + } + + result = ma_device_init_internal__coreaudio(pDevice->pContext, deviceType, NULL, &data, (void*)pDevice); + if (result != MA_SUCCESS) { + return result; + } + + if (deviceType == ma_device_type_capture) { + #if defined(MA_APPLE_DESKTOP) + pDevice->coreaudio.deviceObjectIDCapture = (ma_uint32)data.deviceObjectID; + #endif + pDevice->coreaudio.audioUnitCapture = (ma_ptr)data.audioUnit; + pDevice->coreaudio.pAudioBufferList = (ma_ptr)data.pAudioBufferList; + + pDevice->capture.internalFormat = data.formatOut; + pDevice->capture.internalChannels = data.channelsOut; + pDevice->capture.internalSampleRate = data.sampleRateOut; + MA_COPY_MEMORY(pDevice->capture.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); + pDevice->capture.internalPeriodSizeInFrames = data.periodSizeInFramesOut; + pDevice->capture.internalPeriods = data.periodsOut; + } else if (deviceType == ma_device_type_playback) { + #if defined(MA_APPLE_DESKTOP) + pDevice->coreaudio.deviceObjectIDPlayback = (ma_uint32)data.deviceObjectID; + #endif + pDevice->coreaudio.audioUnitPlayback = (ma_ptr)data.audioUnit; + + pDevice->playback.internalFormat = data.formatOut; + pDevice->playback.internalChannels = data.channelsOut; + pDevice->playback.internalSampleRate = data.sampleRateOut; + MA_COPY_MEMORY(pDevice->playback.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); + pDevice->playback.internalPeriodSizeInFrames = data.periodSizeInFramesOut; + pDevice->playback.internalPeriods = data.periodsOut; } return MA_SUCCESS; } -ma_result ma_device_read__sndio(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead) + +static ma_result ma_device_init__coreaudio(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) { - int result; + ma_result result; - if (pFramesRead != NULL) { - *pFramesRead = 0; - } + MA_ASSERT(pContext != NULL); + MA_ASSERT(pConfig != NULL); + MA_ASSERT(pDevice != NULL); - result = ((ma_sio_read_proc)pDevice->pContext->sndio.sio_read)((struct ma_sio_hdl*)pDevice->sndio.handleCapture, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); - if (result == 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[sndio] Failed to read data from the device to be sent to the device.", MA_FAILED_TO_SEND_DATA_TO_DEVICE); + if (pConfig->deviceType == ma_device_type_loopback) { + return MA_DEVICE_TYPE_NOT_SUPPORTED; } - if (pFramesRead != NULL) { - *pFramesRead = frameCount; + /* No exclusive mode with the Core Audio backend for now. */ + if (((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive) || + ((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive)) { + return MA_SHARE_MODE_NOT_SUPPORTED; } - return MA_SUCCESS; -} - -ma_result ma_device_main_loop__sndio(ma_device* pDevice) -{ - ma_result result = MA_SUCCESS; - ma_bool32 exitLoop = MA_FALSE; + /* Capture needs to be initialized first. */ + if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { + ma_device_init_internal_data__coreaudio data; + data.formatIn = pConfig->capture.format; + data.channelsIn = pConfig->capture.channels; + data.sampleRateIn = pConfig->sampleRate; + MA_COPY_MEMORY(data.channelMapIn, pConfig->capture.channelMap, sizeof(pConfig->capture.channelMap)); + data.usingDefaultFormat = pDevice->capture.usingDefaultFormat; + data.usingDefaultChannels = pDevice->capture.usingDefaultChannels; + data.usingDefaultSampleRate = pDevice->usingDefaultSampleRate; + data.usingDefaultChannelMap = pDevice->capture.usingDefaultChannelMap; + data.shareMode = pConfig->capture.shareMode; + data.periodSizeInFramesIn = pConfig->periodSizeInFrames; + data.periodSizeInMillisecondsIn = pConfig->periodSizeInMilliseconds; + data.periodsIn = pConfig->periods; + data.registerStopEvent = MA_TRUE; - /* Devices need to be started here. */ - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - ((ma_sio_start_proc)pDevice->pContext->sndio.sio_start)((struct ma_sio_hdl*)pDevice->sndio.handleCapture); - } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - ((ma_sio_start_proc)pDevice->pContext->sndio.sio_start)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback); /* <-- Doesn't actually playback until data is written. */ + /* Need at least 3 periods for duplex. */ + if (data.periodsIn < 3 && pConfig->deviceType == ma_device_type_duplex) { + data.periodsIn = 3; + } + + result = ma_device_init_internal__coreaudio(pDevice->pContext, ma_device_type_capture, pConfig->capture.pDeviceID, &data, (void*)pDevice); + if (result != MA_SUCCESS) { + return result; + } + + pDevice->coreaudio.isDefaultCaptureDevice = (pConfig->capture.pDeviceID == NULL); + #if defined(MA_APPLE_DESKTOP) + pDevice->coreaudio.deviceObjectIDCapture = (ma_uint32)data.deviceObjectID; + #endif + pDevice->coreaudio.audioUnitCapture = (ma_ptr)data.audioUnit; + pDevice->coreaudio.pAudioBufferList = (ma_ptr)data.pAudioBufferList; + + pDevice->capture.internalFormat = data.formatOut; + pDevice->capture.internalChannels = data.channelsOut; + pDevice->capture.internalSampleRate = data.sampleRateOut; + MA_COPY_MEMORY(pDevice->capture.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); + pDevice->capture.internalPeriodSizeInFrames = data.periodSizeInFramesOut; + pDevice->capture.internalPeriods = data.periodsOut; + + #if defined(MA_APPLE_DESKTOP) + /* + If we are using the default device we'll need to listen for changes to the system's default device so we can seemlessly + switch the device in the background. + */ + if (pConfig->capture.pDeviceID == NULL) { + ma_device__track__coreaudio(pDevice); + } + #endif } - - while (ma_device__get_state(pDevice) == MA_STATE_STARTED && !exitLoop) { - switch (pDevice->type) - { - case ma_device_type_duplex: - { - /* The process is: device_read -> convert -> callback -> convert -> device_write */ - ma_uint8 capturedDeviceData[8192]; - ma_uint8 playbackDeviceData[8192]; - ma_uint32 capturedDeviceDataCapInFrames = sizeof(capturedDeviceData) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 playbackDeviceDataCapInFrames = sizeof(playbackDeviceData) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - - ma_uint32 totalFramesProcessed = 0; - ma_uint32 periodSizeInFrames = ma_min(pDevice->capture.internalBufferSizeInFrames/pDevice->capture.internalPeriods, pDevice->playback.internalBufferSizeInFrames/pDevice->playback.internalPeriods); - - while (totalFramesProcessed < periodSizeInFrames) { - ma_uint32 framesRemaining = periodSizeInFrames - totalFramesProcessed; - ma_uint32 framesProcessed; - ma_uint32 framesToProcess = framesRemaining; - if (framesToProcess > capturedDeviceDataCapInFrames) { - framesToProcess = capturedDeviceDataCapInFrames; - } - - result = ma_device_read__sndio(pDevice, capturedDeviceData, framesToProcess, &framesProcessed); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } - - pDevice->capture._dspFrameCount = framesToProcess; - pDevice->capture._dspFrames = capturedDeviceData; - - for (;;) { - ma_uint8 capturedData[8192]; - ma_uint8 playbackData[8192]; - ma_uint32 capturedDataCapInFrames = sizeof(capturedData) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); - ma_uint32 playbackDataCapInFrames = sizeof(playbackData) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); - - ma_uint32 capturedFramesToTryProcessing = ma_min(capturedDataCapInFrames, playbackDataCapInFrames); - ma_uint32 capturedFramesToProcess = (ma_uint32)ma_pcm_converter_read(&pDevice->capture.converter, capturedData, capturedFramesToTryProcessing); - if (capturedFramesToProcess == 0) { - break; /* Don't fire the data callback with zero frames. */ - } - - ma_device__on_data(pDevice, playbackData, capturedData, capturedFramesToProcess); - - /* At this point the playbackData buffer should be holding data that needs to be written to the device. */ - pDevice->playback._dspFrameCount = capturedFramesToProcess; - pDevice->playback._dspFrames = playbackData; - for (;;) { - ma_uint32 playbackDeviceFramesCount = (ma_uint32)ma_pcm_converter_read(&pDevice->playback.converter, playbackDeviceData, playbackDeviceDataCapInFrames); - if (playbackDeviceFramesCount == 0) { - break; - } - - result = ma_device_write__sndio(pDevice, playbackDeviceData, playbackDeviceFramesCount, NULL); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } - - if (playbackDeviceFramesCount < playbackDeviceDataCapInFrames) { - break; - } - } - - if (capturedFramesToProcess < capturedFramesToTryProcessing) { - break; - } - - /* In case an error happened from ma_device_write2__alsa()... */ - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } - } - - totalFramesProcessed += framesProcessed; + + /* Playback. */ + if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { + ma_device_init_internal_data__coreaudio data; + data.formatIn = pConfig->playback.format; + data.channelsIn = pConfig->playback.channels; + data.sampleRateIn = pConfig->sampleRate; + MA_COPY_MEMORY(data.channelMapIn, pConfig->playback.channelMap, sizeof(pConfig->playback.channelMap)); + data.usingDefaultFormat = pDevice->playback.usingDefaultFormat; + data.usingDefaultChannels = pDevice->playback.usingDefaultChannels; + data.usingDefaultSampleRate = pDevice->usingDefaultSampleRate; + data.usingDefaultChannelMap = pDevice->playback.usingDefaultChannelMap; + data.shareMode = pConfig->playback.shareMode; + + /* In full-duplex mode we want the playback buffer to be the same size as the capture buffer. */ + if (pConfig->deviceType == ma_device_type_duplex) { + data.periodSizeInFramesIn = pDevice->capture.internalPeriodSizeInFrames; + data.periodsIn = pDevice->capture.internalPeriods; + data.registerStopEvent = MA_FALSE; + } else { + data.periodSizeInFramesIn = pConfig->periodSizeInFrames; + data.periodSizeInMillisecondsIn = pConfig->periodSizeInMilliseconds; + data.periodsIn = pConfig->periods; + data.registerStopEvent = MA_TRUE; + } + + result = ma_device_init_internal__coreaudio(pDevice->pContext, ma_device_type_playback, pConfig->playback.pDeviceID, &data, (void*)pDevice); + if (result != MA_SUCCESS) { + if (pConfig->deviceType == ma_device_type_duplex) { + ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitCapture); + if (pDevice->coreaudio.pAudioBufferList) { + ma__free_from_callbacks(pDevice->coreaudio.pAudioBufferList, &pDevice->pContext->allocationCallbacks); } - } break; - - case ma_device_type_capture: - { - /* We read in chunks of the period size, but use a stack allocated buffer for the intermediary. */ - ma_uint8 intermediaryBuffer[8192]; - ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 periodSizeInFrames = pDevice->capture.internalBufferSizeInFrames / pDevice->capture.internalPeriods; - ma_uint32 framesReadThisPeriod = 0; - while (framesReadThisPeriod < periodSizeInFrames) { - ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesReadThisPeriod; - ma_uint32 framesProcessed; - ma_uint32 framesToReadThisIteration = framesRemainingInPeriod; - if (framesToReadThisIteration > intermediaryBufferSizeInFrames) { - framesToReadThisIteration = intermediaryBufferSizeInFrames; - } - - result = ma_device_read__sndio(pDevice, intermediaryBuffer, framesToReadThisIteration, &framesProcessed); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } - - ma_device__send_frames_to_client(pDevice, framesProcessed, intermediaryBuffer); + } + return result; + } + + pDevice->coreaudio.isDefaultPlaybackDevice = (pConfig->playback.pDeviceID == NULL); + #if defined(MA_APPLE_DESKTOP) + pDevice->coreaudio.deviceObjectIDPlayback = (ma_uint32)data.deviceObjectID; + #endif + pDevice->coreaudio.audioUnitPlayback = (ma_ptr)data.audioUnit; + + pDevice->playback.internalFormat = data.formatOut; + pDevice->playback.internalChannels = data.channelsOut; + pDevice->playback.internalSampleRate = data.sampleRateOut; + MA_COPY_MEMORY(pDevice->playback.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut)); + pDevice->playback.internalPeriodSizeInFrames = data.periodSizeInFramesOut; + pDevice->playback.internalPeriods = data.periodsOut; + + #if defined(MA_APPLE_DESKTOP) + /* + If we are using the default device we'll need to listen for changes to the system's default device so we can seemlessly + switch the device in the background. + */ + if (pConfig->playback.pDeviceID == NULL && (pConfig->deviceType != ma_device_type_duplex || pConfig->capture.pDeviceID != NULL)) { + ma_device__track__coreaudio(pDevice); + } + #endif + } + + pDevice->coreaudio.originalPeriodSizeInFrames = pConfig->periodSizeInFrames; + pDevice->coreaudio.originalPeriodSizeInMilliseconds = pConfig->periodSizeInMilliseconds; + pDevice->coreaudio.originalPeriods = pConfig->periods; + + /* + When stopping the device, a callback is called on another thread. We need to wait for this callback + before returning from ma_device_stop(). This event is used for this. + */ + ma_event_init(pContext, &pDevice->coreaudio.stopEvent); - framesReadThisPeriod += framesProcessed; - } - } break; + /* Need a ring buffer for duplex mode. */ + if (pConfig->deviceType == ma_device_type_duplex) { + ma_uint32 rbSizeInFrames = (ma_uint32)ma_calculate_frame_count_after_resampling(pDevice->sampleRate, pDevice->capture.internalSampleRate, pDevice->capture.internalPeriodSizeInFrames * pDevice->capture.internalPeriods); + ma_result result = ma_pcm_rb_init(pDevice->capture.format, pDevice->capture.channels, rbSizeInFrames, NULL, &pDevice->pContext->allocationCallbacks, &pDevice->coreaudio.duplexRB); + if (result != MA_SUCCESS) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[Core Audio] Failed to initialize ring buffer.", result); + } - case ma_device_type_playback: + /* We need a period to act as a buffer for cases where the playback and capture device's end up desyncing. */ + { + ma_uint32 bufferSizeInFrames = rbSizeInFrames / pDevice->capture.internalPeriods; + void* pBufferData; + ma_pcm_rb_acquire_write(&pDevice->coreaudio.duplexRB, &bufferSizeInFrames, &pBufferData); { - /* We write in chunks of the period size, but use a stack allocated buffer for the intermediary. */ - ma_uint8 intermediaryBuffer[8192]; - ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - ma_uint32 periodSizeInFrames = pDevice->playback.internalBufferSizeInFrames / pDevice->playback.internalPeriods; - ma_uint32 framesWrittenThisPeriod = 0; - while (framesWrittenThisPeriod < periodSizeInFrames) { - ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesWrittenThisPeriod; - ma_uint32 framesProcessed; - ma_uint32 framesToWriteThisIteration = framesRemainingInPeriod; - if (framesToWriteThisIteration > intermediaryBufferSizeInFrames) { - framesToWriteThisIteration = intermediaryBufferSizeInFrames; - } + MA_ZERO_MEMORY(pBufferData, bufferSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels)); + } + ma_pcm_rb_commit_write(&pDevice->coreaudio.duplexRB, bufferSizeInFrames, pBufferData); + } + } - ma_device__read_frames_from_client(pDevice, framesToWriteThisIteration, intermediaryBuffer); + /* + We need to detect when a route has changed so we can update the data conversion pipeline accordingly. This is done + differently on non-Desktop Apple platforms. + */ +#if defined(MA_APPLE_MOBILE) + pDevice->coreaudio.pRouteChangeHandler = (__bridge_retained void*)[[ma_router_change_handler alloc] init:pDevice]; +#endif - result = ma_device_write__sndio(pDevice, intermediaryBuffer, framesToWriteThisIteration, &framesProcessed); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } + return MA_SUCCESS; +} - framesWrittenThisPeriod += framesProcessed; - } - } break; - /* To silence a warning. Will never hit this. */ - case ma_device_type_loopback: - default: break; +static ma_result ma_device_start__coreaudio(ma_device* pDevice) +{ + MA_ASSERT(pDevice != NULL); + + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + OSStatus status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitCapture); + if (status != noErr) { + return ma_result_from_OSStatus(status); + } + } + + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + OSStatus status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); + if (status != noErr) { + if (pDevice->type == ma_device_type_duplex) { + ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture); + } + return ma_result_from_OSStatus(status); } } + + return MA_SUCCESS; +} +static ma_result ma_device_stop__coreaudio(ma_device* pDevice) +{ + MA_ASSERT(pDevice != NULL); - /* Here is where the device is stopped. */ - ma_device_stop__sndio(pDevice); + /* It's not clear from the documentation whether or not AudioOutputUnitStop() actually drains the device or not. */ - return result; + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + OSStatus status = ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture); + if (status != noErr) { + return ma_result_from_OSStatus(status); + } + } + + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + OSStatus status = ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitPlayback); + if (status != noErr) { + return ma_result_from_OSStatus(status); + } + } + + /* We need to wait for the callback to finish before returning. */ + ma_event_wait(&pDevice->coreaudio.stopEvent); + return MA_SUCCESS; } -ma_result ma_context_uninit__sndio(ma_context* pContext) + +static ma_result ma_context_uninit__coreaudio(ma_context* pContext) { - ma_assert(pContext != NULL); - ma_assert(pContext->backend == ma_backend_sndio); + MA_ASSERT(pContext != NULL); + MA_ASSERT(pContext->backend == ma_backend_coreaudio); + +#if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE) + ma_dlclose(pContext, pContext->coreaudio.hAudioUnit); + ma_dlclose(pContext, pContext->coreaudio.hCoreAudio); + ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation); +#endif (void)pContext; return MA_SUCCESS; } -ma_result ma_context_init__sndio(const ma_context_config* pConfig, ma_context* pContext) +#if defined(MA_APPLE_MOBILE) +static AVAudioSessionCategory ma_to_AVAudioSessionCategory(ma_ios_session_category category) +{ + /* The "default" and "none" categories are treated different and should not be used as an input into this function. */ + MA_ASSERT(category != ma_ios_session_category_default); + MA_ASSERT(category != ma_ios_session_category_none); + + switch (category) { + case ma_ios_session_category_ambient: return AVAudioSessionCategoryAmbient; + case ma_ios_session_category_solo_ambient: return AVAudioSessionCategorySoloAmbient; + case ma_ios_session_category_playback: return AVAudioSessionCategoryPlayback; + case ma_ios_session_category_record: return AVAudioSessionCategoryRecord; + case ma_ios_session_category_play_and_record: return AVAudioSessionCategoryPlayAndRecord; + case ma_ios_session_category_multi_route: return AVAudioSessionCategoryMultiRoute; + case ma_ios_session_category_none: return AVAudioSessionCategoryAmbient; + case ma_ios_session_category_default: return AVAudioSessionCategoryAmbient; + default: return AVAudioSessionCategoryAmbient; + } +} +#endif + +static ma_result ma_context_init__coreaudio(const ma_context_config* pConfig, ma_context* pContext) { -#ifndef MA_NO_RUNTIME_LINKING - const char* libsndioNames[] = { - "libsndio.so" - }; - size_t i; + MA_ASSERT(pConfig != NULL); + MA_ASSERT(pContext != NULL); - for (i = 0; i < ma_countof(libsndioNames); ++i) { - pContext->sndio.sndioSO = ma_dlopen(pContext, libsndioNames[i]); - if (pContext->sndio.sndioSO != NULL) { - break; +#if defined(MA_APPLE_MOBILE) + @autoreleasepool { + AVAudioSession* pAudioSession = [AVAudioSession sharedInstance]; + AVAudioSessionCategoryOptions options = pConfig->coreaudio.sessionCategoryOptions; + + MA_ASSERT(pAudioSession != NULL); + + if (pConfig->coreaudio.sessionCategory == ma_ios_session_category_default) { + /* + I'm going to use trial and error to determine our default session category. First we'll try PlayAndRecord. If that fails + we'll try Playback and if that fails we'll try record. If all of these fail we'll just not set the category. + */ + #if !defined(MA_APPLE_TV) && !defined(MA_APPLE_WATCH) + options |= AVAudioSessionCategoryOptionDefaultToSpeaker; + #endif + + if ([pAudioSession setCategory: AVAudioSessionCategoryPlayAndRecord withOptions:options error:nil]) { + /* Using PlayAndRecord */ + } else if ([pAudioSession setCategory: AVAudioSessionCategoryPlayback withOptions:options error:nil]) { + /* Using Playback */ + } else if ([pAudioSession setCategory: AVAudioSessionCategoryRecord withOptions:options error:nil]) { + /* Using Record */ + } else { + /* Leave as default? */ + } + } else { + if (pConfig->coreaudio.sessionCategory != ma_ios_session_category_none) { + if (![pAudioSession setCategory: ma_to_AVAudioSessionCategory(pConfig->coreaudio.sessionCategory) withOptions:options error:nil]) { + return MA_INVALID_OPERATION; /* Failed to set session category. */ + } + } } } +#endif + +#if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE) + pContext->coreaudio.hCoreFoundation = ma_dlopen(pContext, "CoreFoundation.framework/CoreFoundation"); + if (pContext->coreaudio.hCoreFoundation == NULL) { + return MA_API_NOT_FOUND; + } + + pContext->coreaudio.CFStringGetCString = ma_dlsym(pContext, pContext->coreaudio.hCoreFoundation, "CFStringGetCString"); + pContext->coreaudio.CFRelease = ma_dlsym(pContext, pContext->coreaudio.hCoreFoundation, "CFRelease"); + + + pContext->coreaudio.hCoreAudio = ma_dlopen(pContext, "CoreAudio.framework/CoreAudio"); + if (pContext->coreaudio.hCoreAudio == NULL) { + ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation); + return MA_API_NOT_FOUND; + } + + pContext->coreaudio.AudioObjectGetPropertyData = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectGetPropertyData"); + pContext->coreaudio.AudioObjectGetPropertyDataSize = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectGetPropertyDataSize"); + pContext->coreaudio.AudioObjectSetPropertyData = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectSetPropertyData"); + pContext->coreaudio.AudioObjectAddPropertyListener = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectAddPropertyListener"); + pContext->coreaudio.AudioObjectRemovePropertyListener = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectRemovePropertyListener"); - if (pContext->sndio.sndioSO == NULL) { - return MA_NO_BACKEND; + /* + It looks like Apple has moved some APIs from AudioUnit into AudioToolbox on more recent versions of macOS. They are still + defined in AudioUnit, but just in case they decide to remove them from there entirely I'm going to implement a fallback. + The way it'll work is that it'll first try AudioUnit, and if the required symbols are not present there we'll fall back to + AudioToolbox. + */ + pContext->coreaudio.hAudioUnit = ma_dlopen(pContext, "AudioUnit.framework/AudioUnit"); + if (pContext->coreaudio.hAudioUnit == NULL) { + ma_dlclose(pContext, pContext->coreaudio.hCoreAudio); + ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation); + return MA_API_NOT_FOUND; } - pContext->sndio.sio_open = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_open"); - pContext->sndio.sio_close = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_close"); - pContext->sndio.sio_setpar = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_setpar"); - pContext->sndio.sio_getpar = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_getpar"); - pContext->sndio.sio_getcap = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_getcap"); - pContext->sndio.sio_write = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_write"); - pContext->sndio.sio_read = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_read"); - pContext->sndio.sio_start = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_start"); - pContext->sndio.sio_stop = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_stop"); - pContext->sndio.sio_initpar = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_initpar"); + if (ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentFindNext") == NULL) { + /* Couldn't find the required symbols in AudioUnit, so fall back to AudioToolbox. */ + ma_dlclose(pContext, pContext->coreaudio.hAudioUnit); + pContext->coreaudio.hAudioUnit = ma_dlopen(pContext, "AudioToolbox.framework/AudioToolbox"); + if (pContext->coreaudio.hAudioUnit == NULL) { + ma_dlclose(pContext, pContext->coreaudio.hCoreAudio); + ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation); + return MA_API_NOT_FOUND; + } + } + + pContext->coreaudio.AudioComponentFindNext = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentFindNext"); + pContext->coreaudio.AudioComponentInstanceDispose = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentInstanceDispose"); + pContext->coreaudio.AudioComponentInstanceNew = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentInstanceNew"); + pContext->coreaudio.AudioOutputUnitStart = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioOutputUnitStart"); + pContext->coreaudio.AudioOutputUnitStop = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioOutputUnitStop"); + pContext->coreaudio.AudioUnitAddPropertyListener = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitAddPropertyListener"); + pContext->coreaudio.AudioUnitGetPropertyInfo = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitGetPropertyInfo"); + pContext->coreaudio.AudioUnitGetProperty = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitGetProperty"); + pContext->coreaudio.AudioUnitSetProperty = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitSetProperty"); + pContext->coreaudio.AudioUnitInitialize = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitInitialize"); + pContext->coreaudio.AudioUnitRender = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitRender"); #else - pContext->sndio.sio_open = sio_open; - pContext->sndio.sio_close = sio_close; - pContext->sndio.sio_setpar = sio_setpar; - pContext->sndio.sio_getpar = sio_getpar; - pContext->sndio.sio_getcap = sio_getcap; - pContext->sndio.sio_write = sio_write; - pContext->sndio.sio_read = sio_read; - pContext->sndio.sio_start = sio_start; - pContext->sndio.sio_stop = sio_stop; - pContext->sndio.sio_initpar = sio_initpar; + pContext->coreaudio.CFStringGetCString = (ma_proc)CFStringGetCString; + pContext->coreaudio.CFRelease = (ma_proc)CFRelease; + + #if defined(MA_APPLE_DESKTOP) + pContext->coreaudio.AudioObjectGetPropertyData = (ma_proc)AudioObjectGetPropertyData; + pContext->coreaudio.AudioObjectGetPropertyDataSize = (ma_proc)AudioObjectGetPropertyDataSize; + pContext->coreaudio.AudioObjectSetPropertyData = (ma_proc)AudioObjectSetPropertyData; + pContext->coreaudio.AudioObjectAddPropertyListener = (ma_proc)AudioObjectAddPropertyListener; + pContext->coreaudio.AudioObjectRemovePropertyListener = (ma_proc)AudioObjectRemovePropertyListener; + #endif + + pContext->coreaudio.AudioComponentFindNext = (ma_proc)AudioComponentFindNext; + pContext->coreaudio.AudioComponentInstanceDispose = (ma_proc)AudioComponentInstanceDispose; + pContext->coreaudio.AudioComponentInstanceNew = (ma_proc)AudioComponentInstanceNew; + pContext->coreaudio.AudioOutputUnitStart = (ma_proc)AudioOutputUnitStart; + pContext->coreaudio.AudioOutputUnitStop = (ma_proc)AudioOutputUnitStop; + pContext->coreaudio.AudioUnitAddPropertyListener = (ma_proc)AudioUnitAddPropertyListener; + pContext->coreaudio.AudioUnitGetPropertyInfo = (ma_proc)AudioUnitGetPropertyInfo; + pContext->coreaudio.AudioUnitGetProperty = (ma_proc)AudioUnitGetProperty; + pContext->coreaudio.AudioUnitSetProperty = (ma_proc)AudioUnitSetProperty; + pContext->coreaudio.AudioUnitInitialize = (ma_proc)AudioUnitInitialize; + pContext->coreaudio.AudioUnitRender = (ma_proc)AudioUnitRender; #endif - pContext->onUninit = ma_context_uninit__sndio; - pContext->onDeviceIDEqual = ma_context_is_device_id_equal__sndio; - pContext->onEnumDevices = ma_context_enumerate_devices__sndio; - pContext->onGetDeviceInfo = ma_context_get_device_info__sndio; - pContext->onDeviceInit = ma_device_init__sndio; - pContext->onDeviceUninit = ma_device_uninit__sndio; - pContext->onDeviceStart = NULL; /* Not required for synchronous backends. */ - pContext->onDeviceStop = NULL; /* Not required for synchronous backends. */ - pContext->onDeviceMainLoop = ma_device_main_loop__sndio; + pContext->isBackendAsynchronous = MA_TRUE; + + pContext->onUninit = ma_context_uninit__coreaudio; + pContext->onDeviceIDEqual = ma_context_is_device_id_equal__coreaudio; + pContext->onEnumDevices = ma_context_enumerate_devices__coreaudio; + pContext->onGetDeviceInfo = ma_context_get_device_info__coreaudio; + pContext->onDeviceInit = ma_device_init__coreaudio; + pContext->onDeviceUninit = ma_device_uninit__coreaudio; + pContext->onDeviceStart = ma_device_start__coreaudio; + pContext->onDeviceStop = ma_device_stop__coreaudio; + + /* Audio component. */ + { + AudioComponentDescription desc; + desc.componentType = kAudioUnitType_Output; + #if defined(MA_APPLE_DESKTOP) + desc.componentSubType = kAudioUnitSubType_HALOutput; + #else + desc.componentSubType = kAudioUnitSubType_RemoteIO; + #endif + desc.componentManufacturer = kAudioUnitManufacturer_Apple; + desc.componentFlags = 0; + desc.componentFlagsMask = 0; + + pContext->coreaudio.component = ((ma_AudioComponentFindNext_proc)pContext->coreaudio.AudioComponentFindNext)(NULL, &desc); + if (pContext->coreaudio.component == NULL) { + #if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE) + ma_dlclose(pContext, pContext->coreaudio.hAudioUnit); + ma_dlclose(pContext, pContext->coreaudio.hCoreAudio); + ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation); + #endif + return MA_FAILED_TO_INIT_BACKEND; + } + } - (void)pConfig; return MA_SUCCESS; } -#endif /* sndio */ +#endif /* Core Audio */ /****************************************************************************** -audio(4) Backend +sndio Backend ******************************************************************************/ -#ifdef MA_HAS_AUDIO4 +#ifdef MA_HAS_SNDIO #include -#include -#include #include -#include -#include -#include -#if defined(__OpenBSD__) - #include - #if defined(OpenBSD) && OpenBSD >= 201709 - #define MA_AUDIO4_USE_NEW_API - #endif +/* +Only supporting OpenBSD. This did not work very well at all on FreeBSD when I tried it. Not sure if this is due +to miniaudio's implementation or if it's some kind of system configuration issue, but basically the default device +just doesn't emit any sound, or at times you'll hear tiny pieces. I will consider enabling this when there's +demand for it or if I can get it tested and debugged more thoroughly. +*/ +#if 0 +#if defined(__NetBSD__) || defined(__OpenBSD__) +#include +#endif +#if defined(__FreeBSD__) || defined(__DragonFly__) +#include +#endif #endif -void ma_construct_device_id__audio4(char* id, size_t idSize, const char* base, int deviceIndex) -{ - size_t baseLen; +#define MA_SIO_DEVANY "default" +#define MA_SIO_PLAY 1 +#define MA_SIO_REC 2 +#define MA_SIO_NENC 8 +#define MA_SIO_NCHAN 8 +#define MA_SIO_NRATE 16 +#define MA_SIO_NCONF 4 - ma_assert(id != NULL); - ma_assert(idSize > 0); - ma_assert(deviceIndex >= 0); - - baseLen = strlen(base); - ma_assert(idSize > baseLen); - - ma_strcpy_s(id, idSize, base); - ma_itoa_s(deviceIndex, id+baseLen, idSize-baseLen, 10); -} +struct ma_sio_hdl; /* <-- Opaque */ -ma_result ma_extract_device_index_from_id__audio4(const char* id, const char* base, int* pIndexOut) +struct ma_sio_par { - size_t idLen; - size_t baseLen; - const char* deviceIndexStr; + unsigned int bits; + unsigned int bps; + unsigned int sig; + unsigned int le; + unsigned int msb; + unsigned int rchan; + unsigned int pchan; + unsigned int rate; + unsigned int bufsz; + unsigned int xrun; + unsigned int round; + unsigned int appbufsz; + int __pad[3]; + unsigned int __magic; +}; - ma_assert(id != NULL); - ma_assert(base != NULL); - ma_assert(pIndexOut != NULL); - - idLen = strlen(id); - baseLen = strlen(base); - if (idLen <= baseLen) { - return MA_ERROR; /* Doesn't look like the id starts with the base. */ - } - - if (strncmp(id, base, baseLen) != 0) { - return MA_ERROR; /* ID does not begin with base. */ - } - - deviceIndexStr = id + baseLen; - if (deviceIndexStr[0] == '\0') { - return MA_ERROR; /* No index specified in the ID. */ - } - - if (pIndexOut) { - *pIndexOut = atoi(deviceIndexStr); - } - - return MA_SUCCESS; -} +struct ma_sio_enc +{ + unsigned int bits; + unsigned int bps; + unsigned int sig; + unsigned int le; + unsigned int msb; +}; -ma_bool32 ma_context_is_device_id_equal__audio4(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) +struct ma_sio_conf { - ma_assert(pContext != NULL); - ma_assert(pID0 != NULL); - ma_assert(pID1 != NULL); - (void)pContext; + unsigned int enc; + unsigned int rchan; + unsigned int pchan; + unsigned int rate; +}; - return ma_strcmp(pID0->audio4, pID1->audio4) == 0; -} +struct ma_sio_cap +{ + struct ma_sio_enc enc[MA_SIO_NENC]; + unsigned int rchan[MA_SIO_NCHAN]; + unsigned int pchan[MA_SIO_NCHAN]; + unsigned int rate[MA_SIO_NRATE]; + int __pad[7]; + unsigned int nconf; + struct ma_sio_conf confs[MA_SIO_NCONF]; +}; -#if !defined(MA_AUDIO4_USE_NEW_API) /* Old API */ -ma_format ma_format_from_encoding__audio4(unsigned int encoding, unsigned int precision) +typedef struct ma_sio_hdl* (* ma_sio_open_proc) (const char*, unsigned int, int); +typedef void (* ma_sio_close_proc) (struct ma_sio_hdl*); +typedef int (* ma_sio_setpar_proc) (struct ma_sio_hdl*, struct ma_sio_par*); +typedef int (* ma_sio_getpar_proc) (struct ma_sio_hdl*, struct ma_sio_par*); +typedef int (* ma_sio_getcap_proc) (struct ma_sio_hdl*, struct ma_sio_cap*); +typedef size_t (* ma_sio_write_proc) (struct ma_sio_hdl*, const void*, size_t); +typedef size_t (* ma_sio_read_proc) (struct ma_sio_hdl*, void*, size_t); +typedef int (* ma_sio_start_proc) (struct ma_sio_hdl*); +typedef int (* ma_sio_stop_proc) (struct ma_sio_hdl*); +typedef int (* ma_sio_initpar_proc)(struct ma_sio_par*); + +static ma_uint32 ma_get_standard_sample_rate_priority_index__sndio(ma_uint32 sampleRate) /* Lower = higher priority */ { - if (precision == 8 && (encoding == AUDIO_ENCODING_ULINEAR || encoding == AUDIO_ENCODING_ULINEAR || encoding == AUDIO_ENCODING_ULINEAR_LE || encoding == AUDIO_ENCODING_ULINEAR_BE)) { - return ma_format_u8; - } else { - if (ma_is_little_endian() && encoding == AUDIO_ENCODING_SLINEAR_LE) { - if (precision == 16) { - return ma_format_s16; - } else if (precision == 24) { - return ma_format_s24; - } else if (precision == 32) { - return ma_format_s32; - } - } else if (ma_is_big_endian() && encoding == AUDIO_ENCODING_SLINEAR_BE) { - if (precision == 16) { - return ma_format_s16; - } else if (precision == 24) { - return ma_format_s24; - } else if (precision == 32) { - return ma_format_s32; - } + ma_uint32 i; + for (i = 0; i < ma_countof(g_maStandardSampleRatePriorities); ++i) { + if (g_maStandardSampleRatePriorities[i] == sampleRate) { + return i; } } - return ma_format_unknown; /* Encoding not supported. */ + return (ma_uint32)-1; } -void ma_encoding_from_format__audio4(ma_format format, unsigned int* pEncoding, unsigned int* pPrecision) +static ma_format ma_format_from_sio_enc__sndio(unsigned int bits, unsigned int bps, unsigned int sig, unsigned int le, unsigned int msb) { - ma_assert(format != ma_format_unknown); - ma_assert(pEncoding != NULL); - ma_assert(pPrecision != NULL); - - switch (format) - { - case ma_format_u8: - { - *pEncoding = AUDIO_ENCODING_ULINEAR; - *pPrecision = 8; - } break; - - case ma_format_s24: - { - *pEncoding = (ma_is_little_endian()) ? AUDIO_ENCODING_SLINEAR_LE : AUDIO_ENCODING_SLINEAR_BE; - *pPrecision = 24; - } break; - - case ma_format_s32: - { - *pEncoding = (ma_is_little_endian()) ? AUDIO_ENCODING_SLINEAR_LE : AUDIO_ENCODING_SLINEAR_BE; - *pPrecision = 32; - } break; - - case ma_format_s16: - case ma_format_f32: - default: - { - *pEncoding = (ma_is_little_endian()) ? AUDIO_ENCODING_SLINEAR_LE : AUDIO_ENCODING_SLINEAR_BE; - *pPrecision = 16; - } break; + /* We only support native-endian right now. */ + if ((ma_is_little_endian() && le == 0) || (ma_is_big_endian() && le == 1)) { + return ma_format_unknown; } -} - -ma_format ma_format_from_prinfo__audio4(struct audio_prinfo* prinfo) -{ - return ma_format_from_encoding__audio4(prinfo->encoding, prinfo->precision); -} -#else -ma_format ma_format_from_swpar__audio4(struct audio_swpar* par) -{ - if (par->bits == 8 && par->bps == 1 && par->sig == 0) { + + if (bits == 8 && bps == 1 && sig == 0) { return ma_format_u8; } - if (par->bits == 16 && par->bps == 2 && par->sig == 1 && par->le == ma_is_little_endian()) { + if (bits == 16 && bps == 2 && sig == 1) { return ma_format_s16; } - if (par->bits == 24 && par->bps == 3 && par->sig == 1 && par->le == ma_is_little_endian()) { + if (bits == 24 && bps == 3 && sig == 1) { return ma_format_s24; } - if (par->bits == 32 && par->bps == 4 && par->sig == 1 && par->le == ma_is_little_endian()) { - return ma_format_f32; + if (bits == 24 && bps == 4 && sig == 1 && msb == 0) { + /*return ma_format_s24_32;*/ + } + if (bits == 32 && bps == 4 && sig == 1) { + return ma_format_s32; + } + + return ma_format_unknown; +} + +static ma_format ma_find_best_format_from_sio_cap__sndio(struct ma_sio_cap* caps) +{ + ma_format bestFormat; + unsigned int iConfig; + + MA_ASSERT(caps != NULL); + + bestFormat = ma_format_unknown; + for (iConfig = 0; iConfig < caps->nconf; iConfig += 1) { + unsigned int iEncoding; + for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) { + unsigned int bits; + unsigned int bps; + unsigned int sig; + unsigned int le; + unsigned int msb; + ma_format format; + + if ((caps->confs[iConfig].enc & (1UL << iEncoding)) == 0) { + continue; + } + + bits = caps->enc[iEncoding].bits; + bps = caps->enc[iEncoding].bps; + sig = caps->enc[iEncoding].sig; + le = caps->enc[iEncoding].le; + msb = caps->enc[iEncoding].msb; + format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb); + if (format == ma_format_unknown) { + continue; /* Format not supported. */ + } + + if (bestFormat == ma_format_unknown) { + bestFormat = format; + } else { + if (ma_get_format_priority_index(bestFormat) > ma_get_format_priority_index(format)) { /* <-- Lower = better. */ + bestFormat = format; + } + } + } } - - /* Format not supported. */ + return ma_format_unknown; } -#endif -ma_result ma_context_get_device_info_from_fd__audio4(ma_context* pContext, ma_device_type deviceType, int fd, ma_device_info* pInfoOut) +static ma_uint32 ma_find_best_channels_from_sio_cap__sndio(struct ma_sio_cap* caps, ma_device_type deviceType, ma_format requiredFormat) { - audio_device_t fdDevice; -#if !defined(MA_AUDIO4_USE_NEW_API) - int counter = 0; - audio_info_t fdInfo; -#else - struct audio_swpar fdPar; - ma_format format; -#endif + ma_uint32 maxChannels; + unsigned int iConfig; - ma_assert(pContext != NULL); - ma_assert(fd >= 0); - ma_assert(pInfoOut != NULL); + MA_ASSERT(caps != NULL); + MA_ASSERT(requiredFormat != ma_format_unknown); - (void)pContext; - (void)deviceType; - - if (ioctl(fd, AUDIO_GETDEV, &fdDevice) < 0) { - return MA_ERROR; /* Failed to retrieve device info. */ - } - - /* Name. */ - ma_strcpy_s(pInfoOut->name, sizeof(pInfoOut->name), fdDevice.name); - -#if !defined(MA_AUDIO4_USE_NEW_API) - /* Supported formats. We get this by looking at the encodings. */ - for (;;) { - audio_encoding_t encoding; - ma_format format; + /* Just pick whatever configuration has the most channels. */ + maxChannels = 0; + for (iConfig = 0; iConfig < caps->nconf; iConfig += 1) { + /* The encoding should be of requiredFormat. */ + unsigned int iEncoding; + for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) { + unsigned int iChannel; + unsigned int bits; + unsigned int bps; + unsigned int sig; + unsigned int le; + unsigned int msb; + ma_format format; - ma_zero_object(&encoding); - encoding.index = counter; - if (ioctl(fd, AUDIO_GETENC, &encoding) < 0) { - break; - } + if ((caps->confs[iConfig].enc & (1UL << iEncoding)) == 0) { + continue; + } + + bits = caps->enc[iEncoding].bits; + bps = caps->enc[iEncoding].bps; + sig = caps->enc[iEncoding].sig; + le = caps->enc[iEncoding].le; + msb = caps->enc[iEncoding].msb; + format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb); + if (format != requiredFormat) { + continue; + } + + /* Getting here means the format is supported. Iterate over each channel count and grab the biggest one. */ + for (iChannel = 0; iChannel < MA_SIO_NCHAN; iChannel += 1) { + unsigned int chan = 0; + unsigned int channels; - format = ma_format_from_encoding__audio4(encoding.encoding, encoding.precision); - if (format != ma_format_unknown) { - pInfoOut->formats[pInfoOut->formatCount++] = format; + if (deviceType == ma_device_type_playback) { + chan = caps->confs[iConfig].pchan; + } else { + chan = caps->confs[iConfig].rchan; + } + + if ((chan & (1UL << iChannel)) == 0) { + continue; + } + + if (deviceType == ma_device_type_playback) { + channels = caps->pchan[iChannel]; + } else { + channels = caps->rchan[iChannel]; + } + + if (maxChannels < channels) { + maxChannels = channels; + } + } } - - counter += 1; - } - - if (ioctl(fd, AUDIO_GETINFO, &fdInfo) < 0) { - return MA_ERROR; - } - - if (deviceType == ma_device_type_playback) { - pInfoOut->minChannels = fdInfo.play.channels; - pInfoOut->maxChannels = fdInfo.play.channels; - pInfoOut->minSampleRate = fdInfo.play.sample_rate; - pInfoOut->maxSampleRate = fdInfo.play.sample_rate; - } else { - pInfoOut->minChannels = fdInfo.record.channels; - pInfoOut->maxChannels = fdInfo.record.channels; - pInfoOut->minSampleRate = fdInfo.record.sample_rate; - pInfoOut->maxSampleRate = fdInfo.record.sample_rate; - } -#else - if (ioctl(fd, AUDIO_GETPAR, &fdPar) < 0) { - return MA_ERROR; - } - - format = ma_format_from_swpar__audio4(&fdPar); - if (format == ma_format_unknown) { - return MA_FORMAT_NOT_SUPPORTED; - } - pInfoOut->formats[pInfoOut->formatCount++] = format; - - if (deviceType == ma_device_type_playback) { - pInfoOut->minChannels = fdPar.pchan; - pInfoOut->maxChannels = fdPar.pchan; - } else { - pInfoOut->minChannels = fdPar.rchan; - pInfoOut->maxChannels = fdPar.rchan; } - pInfoOut->minSampleRate = fdPar.rate; - pInfoOut->maxSampleRate = fdPar.rate; -#endif - - return MA_SUCCESS; + return maxChannels; } -ma_result ma_context_enumerate_devices__audio4(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +static ma_uint32 ma_find_best_sample_rate_from_sio_cap__sndio(struct ma_sio_cap* caps, ma_device_type deviceType, ma_format requiredFormat, ma_uint32 requiredChannels) { - const int maxDevices = 64; - char devpath[256]; - int iDevice; + ma_uint32 firstSampleRate; + ma_uint32 bestSampleRate; + unsigned int iConfig; - ma_assert(pContext != NULL); - ma_assert(callback != NULL); + MA_ASSERT(caps != NULL); + MA_ASSERT(requiredFormat != ma_format_unknown); + MA_ASSERT(requiredChannels > 0); + MA_ASSERT(requiredChannels <= MA_MAX_CHANNELS); - /* - Every device will be named "/dev/audioN", with a "/dev/audioctlN" equivalent. We use the "/dev/audioctlN" - version here since we can open it even when another process has control of the "/dev/audioN" device. - */ - for (iDevice = 0; iDevice < maxDevices; ++iDevice) { - struct stat st; - int fd; - ma_bool32 isTerminating = MA_FALSE; + firstSampleRate = 0; /* <-- If the device does not support a standard rate we'll fall back to the first one that's found. */ + bestSampleRate = 0; - ma_strcpy_s(devpath, sizeof(devpath), "/dev/audioctl"); - ma_itoa_s(iDevice, devpath+strlen(devpath), sizeof(devpath)-strlen(devpath), 10); - - if (stat(devpath, &st) < 0) { - break; - } + for (iConfig = 0; iConfig < caps->nconf; iConfig += 1) { + /* The encoding should be of requiredFormat. */ + unsigned int iEncoding; + for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) { + unsigned int iChannel; + unsigned int bits; + unsigned int bps; + unsigned int sig; + unsigned int le; + unsigned int msb; + ma_format format; - /* The device exists, but we need to check if it's usable as playback and/or capture. */ - - /* Playback. */ - if (!isTerminating) { - fd = open(devpath, O_RDONLY, 0); - if (fd >= 0) { - /* Supports playback. */ - ma_device_info deviceInfo; - ma_zero_object(&deviceInfo); - ma_construct_device_id__audio4(deviceInfo.id.audio4, sizeof(deviceInfo.id.audio4), "/dev/audio", iDevice); - if (ma_context_get_device_info_from_fd__audio4(pContext, ma_device_type_playback, fd, &deviceInfo) == MA_SUCCESS) { - isTerminating = !callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); + if ((caps->confs[iConfig].enc & (1UL << iEncoding)) == 0) { + continue; + } + + bits = caps->enc[iEncoding].bits; + bps = caps->enc[iEncoding].bps; + sig = caps->enc[iEncoding].sig; + le = caps->enc[iEncoding].le; + msb = caps->enc[iEncoding].msb; + format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb); + if (format != requiredFormat) { + continue; + } + + /* Getting here means the format is supported. Iterate over each channel count and grab the biggest one. */ + for (iChannel = 0; iChannel < MA_SIO_NCHAN; iChannel += 1) { + unsigned int chan = 0; + unsigned int channels; + unsigned int iRate; + + if (deviceType == ma_device_type_playback) { + chan = caps->confs[iConfig].pchan; + } else { + chan = caps->confs[iConfig].rchan; + } + + if ((chan & (1UL << iChannel)) == 0) { + continue; } - close(fd); - } - } - - /* Capture. */ - if (!isTerminating) { - fd = open(devpath, O_WRONLY, 0); - if (fd >= 0) { - /* Supports capture. */ - ma_device_info deviceInfo; - ma_zero_object(&deviceInfo); - ma_construct_device_id__audio4(deviceInfo.id.audio4, sizeof(deviceInfo.id.audio4), "/dev/audio", iDevice); - if (ma_context_get_device_info_from_fd__audio4(pContext, ma_device_type_capture, fd, &deviceInfo) == MA_SUCCESS) { - isTerminating = !callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); + if (deviceType == ma_device_type_playback) { + channels = caps->pchan[iChannel]; + } else { + channels = caps->rchan[iChannel]; } - close(fd); + if (channels != requiredChannels) { + continue; + } + + /* Getting here means we have found a compatible encoding/channel pair. */ + for (iRate = 0; iRate < MA_SIO_NRATE; iRate += 1) { + ma_uint32 rate = (ma_uint32)caps->rate[iRate]; + ma_uint32 ratePriority; + + if (firstSampleRate == 0) { + firstSampleRate = rate; + } + + /* Disregard this rate if it's not a standard one. */ + ratePriority = ma_get_standard_sample_rate_priority_index__sndio(rate); + if (ratePriority == (ma_uint32)-1) { + continue; + } + + if (ma_get_standard_sample_rate_priority_index__sndio(bestSampleRate) > ratePriority) { /* Lower = better. */ + bestSampleRate = rate; + } + } } } - - if (isTerminating) { - break; + } + + /* If a standard sample rate was not found just fall back to the first one that was iterated. */ + if (bestSampleRate == 0) { + bestSampleRate = firstSampleRate; + } + + return bestSampleRate; +} + + +static ma_bool32 ma_context_is_device_id_equal__sndio(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) +{ + MA_ASSERT(pContext != NULL); + MA_ASSERT(pID0 != NULL); + MA_ASSERT(pID1 != NULL); + (void)pContext; + + return ma_strcmp(pID0->sndio, pID1->sndio) == 0; +} + +static ma_result ma_context_enumerate_devices__sndio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +{ + ma_bool32 isTerminating = MA_FALSE; + struct ma_sio_hdl* handle; + + MA_ASSERT(pContext != NULL); + MA_ASSERT(callback != NULL); + + /* sndio doesn't seem to have a good device enumeration API, so I'm therefore only enumerating over default devices for now. */ + + /* Playback. */ + if (!isTerminating) { + handle = ((ma_sio_open_proc)pContext->sndio.sio_open)(MA_SIO_DEVANY, MA_SIO_PLAY, 0); + if (handle != NULL) { + /* Supports playback. */ + ma_device_info deviceInfo; + MA_ZERO_OBJECT(&deviceInfo); + ma_strcpy_s(deviceInfo.id.sndio, sizeof(deviceInfo.id.sndio), MA_SIO_DEVANY); + ma_strcpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME); + + isTerminating = !callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); + + ((ma_sio_close_proc)pContext->sndio.sio_close)(handle); + } + } + + /* Capture. */ + if (!isTerminating) { + handle = ((ma_sio_open_proc)pContext->sndio.sio_open)(MA_SIO_DEVANY, MA_SIO_REC, 0); + if (handle != NULL) { + /* Supports capture. */ + ma_device_info deviceInfo; + MA_ZERO_OBJECT(&deviceInfo); + ma_strcpy_s(deviceInfo.id.sndio, sizeof(deviceInfo.id.sndio), "default"); + ma_strcpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME); + + isTerminating = !callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); + + ((ma_sio_close_proc)pContext->sndio.sio_close)(handle); } } return MA_SUCCESS; } -ma_result ma_context_get_device_info__audio4(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) +static ma_result ma_context_get_device_info__sndio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) { - int fd = -1; - int deviceIndex = -1; - char ctlid[256]; - ma_result result; + char devid[256]; + struct ma_sio_hdl* handle; + struct ma_sio_cap caps; + unsigned int iConfig; - ma_assert(pContext != NULL); + MA_ASSERT(pContext != NULL); (void)shareMode; - /* - We need to open the "/dev/audioctlN" device to get the info. To do this we need to extract the number - from the device ID which will be in "/dev/audioN" format. - */ + /* We need to open the device before we can get information about it. */ if (pDeviceID == NULL) { - /* Default device. */ - ma_strcpy_s(ctlid, sizeof(ctlid), "/dev/audioctl"); + ma_strcpy_s(devid, sizeof(devid), MA_SIO_DEVANY); + ma_strcpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), (deviceType == ma_device_type_playback) ? MA_DEFAULT_PLAYBACK_DEVICE_NAME : MA_DEFAULT_CAPTURE_DEVICE_NAME); } else { - /* Specific device. We need to convert from "/dev/audioN" to "/dev/audioctlN". */ - result = ma_extract_device_index_from_id__audio4(pDeviceID->audio4, "/dev/audio", &deviceIndex); - if (result != MA_SUCCESS) { - return result; - } - - ma_construct_device_id__audio4(ctlid, sizeof(ctlid), "/dev/audioctl", deviceIndex); + ma_strcpy_s(devid, sizeof(devid), pDeviceID->sndio); + ma_strcpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), devid); } - fd = open(ctlid, (deviceType == ma_device_type_playback) ? O_WRONLY : O_RDONLY, 0); - if (fd == -1) { + handle = ((ma_sio_open_proc)pContext->sndio.sio_open)(devid, (deviceType == ma_device_type_playback) ? MA_SIO_PLAY : MA_SIO_REC, 0); + if (handle == NULL) { return MA_NO_DEVICE; } - if (deviceIndex == -1) { - ma_strcpy_s(pDeviceInfo->id.audio4, sizeof(pDeviceInfo->id.audio4), "/dev/audio"); - } else { - ma_construct_device_id__audio4(pDeviceInfo->id.audio4, sizeof(pDeviceInfo->id.audio4), "/dev/audio", deviceIndex); + if (((ma_sio_getcap_proc)pContext->sndio.sio_getcap)(handle, &caps) == 0) { + return MA_ERROR; } - result = ma_context_get_device_info_from_fd__audio4(pContext, deviceType, fd, pDeviceInfo); - - close(fd); - return result; -} + for (iConfig = 0; iConfig < caps.nconf; iConfig += 1) { + /* + The main thing we care about is that the encoding is supported by miniaudio. If it is, we want to give + preference to some formats over others. + */ + unsigned int iEncoding; + unsigned int iChannel; + unsigned int iRate; -void ma_device_uninit__audio4(ma_device* pDevice) -{ - ma_assert(pDevice != NULL); + for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) { + unsigned int bits; + unsigned int bps; + unsigned int sig; + unsigned int le; + unsigned int msb; + ma_format format; + ma_bool32 formatExists = MA_FALSE; + ma_uint32 iExistingFormat; - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - close(pDevice->audio4.fdCapture); - } + if ((caps.confs[iConfig].enc & (1UL << iEncoding)) == 0) { + continue; + } + + bits = caps.enc[iEncoding].bits; + bps = caps.enc[iEncoding].bps; + sig = caps.enc[iEncoding].sig; + le = caps.enc[iEncoding].le; + msb = caps.enc[iEncoding].msb; + format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb); + if (format == ma_format_unknown) { + continue; /* Format not supported. */ + } + + /* Add this format if it doesn't already exist. */ + for (iExistingFormat = 0; iExistingFormat < pDeviceInfo->formatCount; iExistingFormat += 1) { + if (pDeviceInfo->formats[iExistingFormat] == format) { + formatExists = MA_TRUE; + break; + } + } + + if (!formatExists) { + pDeviceInfo->formats[pDeviceInfo->formatCount++] = format; + } + } + + /* Channels. */ + for (iChannel = 0; iChannel < MA_SIO_NCHAN; iChannel += 1) { + unsigned int chan = 0; + unsigned int channels; - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - close(pDevice->audio4.fdPlayback); + if (deviceType == ma_device_type_playback) { + chan = caps.confs[iConfig].pchan; + } else { + chan = caps.confs[iConfig].rchan; + } + + if ((chan & (1UL << iChannel)) == 0) { + continue; + } + + if (deviceType == ma_device_type_playback) { + channels = caps.pchan[iChannel]; + } else { + channels = caps.rchan[iChannel]; + } + + if (pDeviceInfo->minChannels > channels) { + pDeviceInfo->minChannels = channels; + } + if (pDeviceInfo->maxChannels < channels) { + pDeviceInfo->maxChannels = channels; + } + } + + /* Sample rates. */ + for (iRate = 0; iRate < MA_SIO_NRATE; iRate += 1) { + if ((caps.confs[iConfig].rate & (1UL << iRate)) != 0) { + unsigned int rate = caps.rate[iRate]; + if (pDeviceInfo->minSampleRate > rate) { + pDeviceInfo->minSampleRate = rate; + } + if (pDeviceInfo->maxSampleRate < rate) { + pDeviceInfo->maxSampleRate = rate; + } + } + } } + + ((ma_sio_close_proc)pContext->sndio.sio_close)(handle); + return MA_SUCCESS; } -ma_result ma_device_init_fd__audio4(ma_context* pContext, const ma_device_config* pConfig, ma_device_type deviceType, ma_device* pDevice) +static void ma_device_uninit__sndio(ma_device* pDevice) { - const char* pDefaultDeviceNames[] = { - "/dev/audio", - "/dev/audio0" - }; - int fd; - int fdFlags = 0; -#if !defined(MA_AUDIO4_USE_NEW_API) /* Old API */ - audio_info_t fdInfo; -#else - struct audio_swpar fdPar; -#endif - ma_format internalFormat; - ma_uint32 internalChannels; - ma_uint32 internalSampleRate; - ma_uint32 internalBufferSizeInFrames; - ma_uint32 internalPeriods; - - ma_assert(pContext != NULL); - ma_assert(pConfig != NULL); - ma_assert(deviceType != ma_device_type_duplex); - ma_assert(pDevice != NULL); - - (void)pContext; - - /* The first thing to do is open the file. */ - if (deviceType == ma_device_type_capture) { - fdFlags = O_RDONLY; - } else { - fdFlags = O_WRONLY; - } - /*fdFlags |= O_NONBLOCK;*/ + MA_ASSERT(pDevice != NULL); - if ((deviceType == ma_device_type_capture && pConfig->capture.pDeviceID == NULL) || (deviceType == ma_device_type_playback && pConfig->playback.pDeviceID == NULL)) { - /* Default device. */ - size_t iDevice; - for (iDevice = 0; iDevice < ma_countof(pDefaultDeviceNames); ++iDevice) { - fd = open(pDefaultDeviceNames[iDevice], fdFlags, 0); - if (fd != -1) { - break; - } - } - } else { - /* Specific device. */ - fd = open((deviceType == ma_device_type_capture) ? pConfig->capture.pDeviceID->audio4 : pConfig->playback.pDeviceID->audio4, fdFlags, 0); + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)pDevice->sndio.handleCapture); } - if (fd == -1) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to open device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback); } +} -#if !defined(MA_AUDIO4_USE_NEW_API) /* Old API */ - AUDIO_INITINFO(&fdInfo); - - /* We get the driver to do as much of the data conversion as possible. */ - if (deviceType == ma_device_type_capture) { - fdInfo.mode = AUMODE_RECORD; - ma_encoding_from_format__audio4(pConfig->capture.format, &fdInfo.record.encoding, &fdInfo.record.precision); - fdInfo.record.channels = pConfig->capture.channels; - fdInfo.record.sample_rate = pConfig->sampleRate; - } else { - fdInfo.mode = AUMODE_PLAY; - ma_encoding_from_format__audio4(pConfig->playback.format, &fdInfo.play.encoding, &fdInfo.play.precision); - fdInfo.play.channels = pConfig->playback.channels; - fdInfo.play.sample_rate = pConfig->sampleRate; - } +static ma_result ma_device_init_handle__sndio(ma_context* pContext, const ma_device_config* pConfig, ma_device_type deviceType, ma_device* pDevice) +{ + const char* pDeviceName; + ma_ptr handle; + int openFlags = 0; + struct ma_sio_cap caps; + struct ma_sio_par par; + ma_device_id* pDeviceID; + ma_format format; + ma_uint32 channels; + ma_uint32 sampleRate; + ma_format internalFormat; + ma_uint32 internalChannels; + ma_uint32 internalSampleRate; + ma_uint32 internalPeriodSizeInFrames; + ma_uint32 internalPeriods; - if (ioctl(fd, AUDIO_SETINFO, &fdInfo) < 0) { - close(fd); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to set device format. AUDIO_SETINFO failed.", MA_FORMAT_NOT_SUPPORTED); - } - - if (ioctl(fd, AUDIO_GETINFO, &fdInfo) < 0) { - close(fd); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] AUDIO_GETINFO failed.", MA_FORMAT_NOT_SUPPORTED); - } + MA_ASSERT(pContext != NULL); + MA_ASSERT(pConfig != NULL); + MA_ASSERT(deviceType != ma_device_type_duplex); + MA_ASSERT(pDevice != NULL); if (deviceType == ma_device_type_capture) { - internalFormat = ma_format_from_prinfo__audio4(&fdInfo.record); - internalChannels = fdInfo.record.channels; - internalSampleRate = fdInfo.record.sample_rate; + openFlags = MA_SIO_REC; + pDeviceID = pConfig->capture.pDeviceID; + format = pConfig->capture.format; + channels = pConfig->capture.channels; + sampleRate = pConfig->sampleRate; } else { - internalFormat = ma_format_from_prinfo__audio4(&fdInfo.play); - internalChannels = fdInfo.play.channels; - internalSampleRate = fdInfo.play.sample_rate; + openFlags = MA_SIO_PLAY; + pDeviceID = pConfig->playback.pDeviceID; + format = pConfig->playback.format; + channels = pConfig->playback.channels; + sampleRate = pConfig->sampleRate; } - if (internalFormat == ma_format_unknown) { - close(fd); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] The device's internal device format is not supported by miniaudio. The device is unusable.", MA_FORMAT_NOT_SUPPORTED); + pDeviceName = MA_SIO_DEVANY; + if (pDeviceID != NULL) { + pDeviceName = pDeviceID->sndio; } - /* Buffer. */ - { - ma_uint32 internalBufferSizeInBytes; + handle = (ma_ptr)((ma_sio_open_proc)pContext->sndio.sio_open)(pDeviceName, openFlags, 0); + if (handle == NULL) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[sndio] Failed to open device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - internalBufferSizeInFrames = pConfig->bufferSizeInFrames; - if (internalBufferSizeInFrames == 0) { - internalBufferSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->bufferSizeInMilliseconds, internalSampleRate); - } + /* We need to retrieve the device caps to determine the most appropriate format to use. */ + if (((ma_sio_getcap_proc)pContext->sndio.sio_getcap)((struct ma_sio_hdl*)handle, &caps) == 0) { + ((ma_sio_close_proc)pContext->sndio.sio_close)((struct ma_sio_hdl*)handle); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[sndio] Failed to retrieve device caps.", MA_ERROR); + } - internalBufferSizeInBytes = internalBufferSizeInFrames * ma_get_bytes_per_frame(internalFormat, internalChannels); - if (internalBufferSizeInBytes < 16) { - internalBufferSizeInBytes = 16; + /* + Note: sndio reports a huge range of available channels. This is inconvenient for us because there's no real + way, as far as I can tell, to get the _actual_ channel count of the device. I'm therefore restricting this + to the requested channels, regardless of whether or not the default channel count is requested. + + For hardware devices, I'm suspecting only a single channel count will be reported and we can safely use the + value returned by ma_find_best_channels_from_sio_cap__sndio(). + */ + if (deviceType == ma_device_type_capture) { + if (pDevice->capture.usingDefaultFormat) { + format = ma_find_best_format_from_sio_cap__sndio(&caps); } - - internalPeriods = pConfig->periods; - if (internalPeriods < 2) { - internalPeriods = 2; + if (pDevice->capture.usingDefaultChannels) { + if (strlen(pDeviceName) > strlen("rsnd/") && strncmp(pDeviceName, "rsnd/", strlen("rsnd/")) == 0) { + channels = ma_find_best_channels_from_sio_cap__sndio(&caps, deviceType, format); + } } - - /* What miniaudio calls a fragment, audio4 calls a block. */ - AUDIO_INITINFO(&fdInfo); - fdInfo.hiwat = internalPeriods; - fdInfo.lowat = internalPeriods-1; - fdInfo.blocksize = internalBufferSizeInBytes / internalPeriods; - if (ioctl(fd, AUDIO_SETINFO, &fdInfo) < 0) { - close(fd); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to set internal buffer size. AUDIO_SETINFO failed.", MA_FORMAT_NOT_SUPPORTED); + } else { + if (pDevice->playback.usingDefaultFormat) { + format = ma_find_best_format_from_sio_cap__sndio(&caps); + } + if (pDevice->playback.usingDefaultChannels) { + if (strlen(pDeviceName) > strlen("rsnd/") && strncmp(pDeviceName, "rsnd/", strlen("rsnd/")) == 0) { + channels = ma_find_best_channels_from_sio_cap__sndio(&caps, deviceType, format); + } } - - internalPeriods = fdInfo.hiwat; - internalBufferSizeInFrames = (fdInfo.blocksize * fdInfo.hiwat) / ma_get_bytes_per_frame(internalFormat, internalChannels); } -#else - /* We need to retrieve the format of the device so we can know the channel count and sample rate. Then we can calculate the buffer size. */ - if (ioctl(fd, AUDIO_GETPAR, &fdPar) < 0) { - close(fd); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to retrieve initial device parameters.", MA_FORMAT_NOT_SUPPORTED); + + if (pDevice->usingDefaultSampleRate) { + sampleRate = ma_find_best_sample_rate_from_sio_cap__sndio(&caps, pConfig->deviceType, format, channels); } - internalFormat = ma_format_from_swpar__audio4(&fdPar); - internalChannels = (deviceType == ma_device_type_capture) ? fdPar.rchan : fdPar.pchan; - internalSampleRate = fdPar.rate; - if (internalFormat == ma_format_unknown) { - close(fd); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] The device's internal device format is not supported by miniaudio. The device is unusable.", MA_FORMAT_NOT_SUPPORTED); + ((ma_sio_initpar_proc)pDevice->pContext->sndio.sio_initpar)(&par); + par.msb = 0; + par.le = ma_is_little_endian(); + + switch (format) { + case ma_format_u8: + { + par.bits = 8; + par.bps = 1; + par.sig = 0; + } break; + + case ma_format_s24: + { + par.bits = 24; + par.bps = 3; + par.sig = 1; + } break; + + case ma_format_s32: + { + par.bits = 32; + par.bps = 4; + par.sig = 1; + } break; + + case ma_format_s16: + case ma_format_f32: + default: + { + par.bits = 16; + par.bps = 2; + par.sig = 1; + } break; + } + + if (deviceType == ma_device_type_capture) { + par.rchan = channels; + } else { + par.pchan = channels; } - /* Buffer. */ - { - ma_uint32 internalBufferSizeInBytes; + par.rate = sampleRate; - internalBufferSizeInFrames = pConfig->bufferSizeInFrames; - if (internalBufferSizeInFrames == 0) { - internalBufferSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->bufferSizeInMilliseconds, internalSampleRate); - } + internalPeriodSizeInFrames = pConfig->periodSizeInFrames; + if (internalPeriodSizeInFrames == 0) { + internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->periodSizeInMilliseconds, par.rate); + } - /* What miniaudio calls a fragment, audio4 calls a block. */ - internalBufferSizeInBytes = internalBufferSizeInFrames * ma_get_bytes_per_frame(internalFormat, internalChannels); - if (internalBufferSizeInBytes < 16) { - internalBufferSizeInBytes = 16; - } - - fdPar.nblks = pConfig->periods; - fdPar.round = internalBufferSizeInBytes / fdPar.nblks; + par.round = internalPeriodSizeInFrames; + par.appbufsz = par.round * pConfig->periods; - if (ioctl(fd, AUDIO_SETPAR, &fdPar) < 0) { - close(fd); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to set device parameters.", MA_FORMAT_NOT_SUPPORTED); - } - - if (ioctl(fd, AUDIO_GETPAR, &fdPar) < 0) { - close(fd); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to retrieve actual device parameters.", MA_FORMAT_NOT_SUPPORTED); - } + if (((ma_sio_setpar_proc)pContext->sndio.sio_setpar)((struct ma_sio_hdl*)handle, &par) == 0) { + ((ma_sio_close_proc)pContext->sndio.sio_close)((struct ma_sio_hdl*)handle); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[sndio] Failed to set buffer size.", MA_FORMAT_NOT_SUPPORTED); } - - internalFormat = ma_format_from_swpar__audio4(&fdPar); - internalChannels = (deviceType == ma_device_type_capture) ? fdPar.rchan : fdPar.pchan; - internalSampleRate = fdPar.rate; - internalPeriods = fdPar.nblks; - internalBufferSizeInFrames = (fdPar.nblks * fdPar.round) / ma_get_bytes_per_frame(internalFormat, internalChannels); -#endif - - if (internalFormat == ma_format_unknown) { - close(fd); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] The device's internal device format is not supported by miniaudio. The device is unusable.", MA_FORMAT_NOT_SUPPORTED); + if (((ma_sio_getpar_proc)pContext->sndio.sio_getpar)((struct ma_sio_hdl*)handle, &par) == 0) { + ((ma_sio_close_proc)pContext->sndio.sio_close)((struct ma_sio_hdl*)handle); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[sndio] Failed to retrieve buffer size.", MA_FORMAT_NOT_SUPPORTED); } + internalFormat = ma_format_from_sio_enc__sndio(par.bits, par.bps, par.sig, par.le, par.msb); + internalChannels = (deviceType == ma_device_type_capture) ? par.rchan : par.pchan; + internalSampleRate = par.rate; + internalPeriods = par.appbufsz / par.round; + internalPeriodSizeInFrames = par.round; + if (deviceType == ma_device_type_capture) { - pDevice->audio4.fdCapture = fd; + pDevice->sndio.handleCapture = handle; pDevice->capture.internalFormat = internalFormat; pDevice->capture.internalChannels = internalChannels; pDevice->capture.internalSampleRate = internalSampleRate; - ma_get_standard_channel_map(ma_standard_channel_map_sound4, internalChannels, pDevice->capture.internalChannelMap); - pDevice->capture.internalBufferSizeInFrames = internalBufferSizeInFrames; + ma_get_standard_channel_map(ma_standard_channel_map_sndio, pDevice->capture.internalChannels, pDevice->capture.internalChannelMap); + pDevice->capture.internalPeriodSizeInFrames = internalPeriodSizeInFrames; pDevice->capture.internalPeriods = internalPeriods; } else { - pDevice->audio4.fdPlayback = fd; + pDevice->sndio.handlePlayback = handle; pDevice->playback.internalFormat = internalFormat; pDevice->playback.internalChannels = internalChannels; pDevice->playback.internalSampleRate = internalSampleRate; - ma_get_standard_channel_map(ma_standard_channel_map_sound4, internalChannels, pDevice->playback.internalChannelMap); - pDevice->playback.internalBufferSizeInFrames = internalBufferSizeInFrames; + ma_get_standard_channel_map(ma_standard_channel_map_sndio, pDevice->playback.internalChannels, pDevice->playback.internalChannelMap); + pDevice->playback.internalPeriodSizeInFrames = internalPeriodSizeInFrames; pDevice->playback.internalPeriods = internalPeriods; } +#ifdef MA_DEBUG_OUTPUT + printf("DEVICE INFO\n"); + printf(" Format: %s\n", ma_get_format_name(internalFormat)); + printf(" Channels: %d\n", internalChannels); + printf(" Sample Rate: %d\n", internalSampleRate); + printf(" Period Size: %d\n", internalPeriodSizeInFrames); + printf(" Periods: %d\n", internalPeriods); + printf(" appbufsz: %d\n", par.appbufsz); + printf(" round: %d\n", par.round); +#endif + return MA_SUCCESS; } -ma_result ma_device_init__audio4(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) +static ma_result ma_device_init__sndio(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) { - ma_assert(pDevice != NULL); + MA_ASSERT(pDevice != NULL); - ma_zero_object(&pDevice->audio4); + MA_ZERO_OBJECT(&pDevice->sndio); if (pConfig->deviceType == ma_device_type_loopback) { return MA_DEVICE_TYPE_NOT_SUPPORTED; } - - pDevice->audio4.fdCapture = -1; - pDevice->audio4.fdPlayback = -1; - - /* - The version of the operating system dictates whether or not the device is exclusive or shared. NetBSD - introduced in-kernel mixing which means it's shared. All other BSD flavours are exclusive as far as - I'm aware. - */ -#if defined(__NetBSD_Version__) && __NetBSD_Version__ >= 800000000 - /* NetBSD 8.0+ */ - if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) || - ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) { - return MA_SHARE_MODE_NOT_SUPPORTED; - } -#else - /* All other flavors. */ -#endif if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { - ma_result result = ma_device_init_fd__audio4(pContext, pConfig, ma_device_type_capture, pDevice); + ma_result result = ma_device_init_handle__sndio(pContext, pConfig, ma_device_type_capture, pDevice); if (result != MA_SUCCESS) { return result; } } if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { - ma_result result = ma_device_init_fd__audio4(pContext, pConfig, ma_device_type_playback, pDevice); + ma_result result = ma_device_init_handle__sndio(pContext, pConfig, ma_device_type_playback, pDevice); if (result != MA_SUCCESS) { - if (pConfig->deviceType == ma_device_type_duplex) { - close(pDevice->audio4.fdCapture); - } return result; } } @@ -22245,68 +23418,32 @@ ma_result ma_device_init__audio4(ma_context* pContext, const ma_device_config* p return MA_SUCCESS; } -#if 0 -ma_result ma_device_start__audio4(ma_device* pDevice) -{ - ma_assert(pDevice != NULL); - - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - if (pDevice->audio4.fdCapture == -1) { - return MA_INVALID_ARGS; - } - } - - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - if (pDevice->audio4.fdPlayback == -1) { - return MA_INVALID_ARGS; - } - } - - return MA_SUCCESS; -} -#endif - -ma_result ma_device_stop_fd__audio4(ma_device* pDevice, int fd) +static ma_result ma_device_stop__sndio(ma_device* pDevice) { - if (fd == -1) { - return MA_INVALID_ARGS; - } - -#if !defined(MA_AUDIO4_USE_NEW_API) - if (ioctl(fd, AUDIO_FLUSH, 0) < 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to stop device. AUDIO_FLUSH failed.", MA_FAILED_TO_STOP_BACKEND_DEVICE); - } -#else - if (ioctl(fd, AUDIO_STOP, 0) < 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to stop device. AUDIO_STOP failed.", MA_FAILED_TO_STOP_BACKEND_DEVICE); - } -#endif + MA_ASSERT(pDevice != NULL); - return MA_SUCCESS; -} + /* + From the documentation: -ma_result ma_device_stop__audio4(ma_device* pDevice) -{ - ma_assert(pDevice != NULL); + The sio_stop() function puts the audio subsystem in the same state as before sio_start() is called. It stops recording, drains the play buffer and then + stops playback. If samples to play are queued but playback hasn't started yet then playback is forced immediately; playback will actually stop once the + buffer is drained. In no case are samples in the play buffer discarded. + + Therefore, sio_stop() performs all of the necessary draining for us. + */ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - ma_result result = ma_device_stop_fd__audio4(pDevice, pDevice->audio4.fdCapture); - if (result != MA_SUCCESS) { - return result; - } + ((ma_sio_stop_proc)pDevice->pContext->sndio.sio_stop)((struct ma_sio_hdl*)pDevice->sndio.handleCapture); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - ma_result result = ma_device_stop_fd__audio4(pDevice, pDevice->audio4.fdPlayback); - if (result != MA_SUCCESS) { - return result; - } + ((ma_sio_stop_proc)pDevice->pContext->sndio.sio_stop)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback); } return MA_SUCCESS; } -ma_result ma_device_write__audio4(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) +static ma_result ma_device_write__sndio(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) { int result; @@ -22314,19 +23451,19 @@ ma_result ma_device_write__audio4(ma_device* pDevice, const void* pPCMFrames, ma *pFramesWritten = 0; } - result = write(pDevice->audio4.fdPlayback, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); - if (result < 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to write data to the device.", MA_FAILED_TO_SEND_DATA_TO_DEVICE); + result = ((ma_sio_write_proc)pDevice->pContext->sndio.sio_write)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); + if (result == 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[sndio] Failed to send data from the client to the device.", MA_FAILED_TO_SEND_DATA_TO_DEVICE); } if (pFramesWritten != NULL) { - *pFramesWritten = (ma_uint32)result / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + *pFramesWritten = frameCount; } - + return MA_SUCCESS; } -ma_result ma_device_read__audio4(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead) +static ma_result ma_device_read__sndio(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead) { int result; @@ -22334,24 +23471,30 @@ ma_result ma_device_read__audio4(ma_device* pDevice, void* pPCMFrames, ma_uint32 *pFramesRead = 0; } - result = read(pDevice->audio4.fdCapture, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); - if (result < 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to read data from the device.", MA_FAILED_TO_READ_DATA_FROM_DEVICE); + result = ((ma_sio_read_proc)pDevice->pContext->sndio.sio_read)((struct ma_sio_hdl*)pDevice->sndio.handleCapture, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); + if (result == 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[sndio] Failed to read data from the device to be sent to the device.", MA_FAILED_TO_SEND_DATA_TO_DEVICE); } if (pFramesRead != NULL) { - *pFramesRead = (ma_uint32)result / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + *pFramesRead = frameCount; } - + return MA_SUCCESS; } -ma_result ma_device_main_loop__audio4(ma_device* pDevice) +static ma_result ma_device_main_loop__sndio(ma_device* pDevice) { ma_result result = MA_SUCCESS; ma_bool32 exitLoop = MA_FALSE; - /* No need to explicitly start the device like the other backends. */ + /* Devices need to be started here. */ + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + ((ma_sio_start_proc)pDevice->pContext->sndio.sio_start)((struct ma_sio_hdl*)pDevice->sndio.handleCapture); + } + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + ((ma_sio_start_proc)pDevice->pContext->sndio.sio_start)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback); /* <-- Doesn't actually playback until data is written. */ + } while (ma_device__get_state(pDevice) == MA_STATE_STARTED && !exitLoop) { switch (pDevice->type) @@ -22359,77 +23502,88 @@ ma_result ma_device_main_loop__audio4(ma_device* pDevice) case ma_device_type_duplex: { /* The process is: device_read -> convert -> callback -> convert -> device_write */ - ma_uint8 capturedDeviceData[8192]; - ma_uint8 playbackDeviceData[8192]; - ma_uint32 capturedDeviceDataCapInFrames = sizeof(capturedDeviceData) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 playbackDeviceDataCapInFrames = sizeof(playbackDeviceData) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - - ma_uint32 totalFramesProcessed = 0; - ma_uint32 periodSizeInFrames = ma_min(pDevice->capture.internalBufferSizeInFrames/pDevice->capture.internalPeriods, pDevice->playback.internalBufferSizeInFrames/pDevice->playback.internalPeriods); + ma_uint32 totalCapturedDeviceFramesProcessed = 0; + ma_uint32 capturedDevicePeriodSizeInFrames = ma_min(pDevice->capture.internalPeriodSizeInFrames, pDevice->playback.internalPeriodSizeInFrames); - while (totalFramesProcessed < periodSizeInFrames) { - ma_uint32 framesRemaining = periodSizeInFrames - totalFramesProcessed; - ma_uint32 framesProcessed; - ma_uint32 framesToProcess = framesRemaining; - if (framesToProcess > capturedDeviceDataCapInFrames) { - framesToProcess = capturedDeviceDataCapInFrames; + while (totalCapturedDeviceFramesProcessed < capturedDevicePeriodSizeInFrames) { + ma_uint8 capturedDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint8 playbackDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 capturedDeviceDataCapInFrames = sizeof(capturedDeviceData) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + ma_uint32 playbackDeviceDataCapInFrames = sizeof(playbackDeviceData) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + ma_uint32 capturedDeviceFramesRemaining; + ma_uint32 capturedDeviceFramesProcessed; + ma_uint32 capturedDeviceFramesToProcess; + ma_uint32 capturedDeviceFramesToTryProcessing = capturedDevicePeriodSizeInFrames - totalCapturedDeviceFramesProcessed; + if (capturedDeviceFramesToTryProcessing > capturedDeviceDataCapInFrames) { + capturedDeviceFramesToTryProcessing = capturedDeviceDataCapInFrames; } - result = ma_device_read__audio4(pDevice, capturedDeviceData, framesToProcess, &framesProcessed); + result = ma_device_read__sndio(pDevice, capturedDeviceData, capturedDeviceFramesToTryProcessing, &capturedDeviceFramesToProcess); if (result != MA_SUCCESS) { exitLoop = MA_TRUE; break; } - pDevice->capture._dspFrameCount = framesToProcess; - pDevice->capture._dspFrames = capturedDeviceData; + capturedDeviceFramesRemaining = capturedDeviceFramesToProcess; + capturedDeviceFramesProcessed = 0; for (;;) { - ma_uint8 capturedData[8192]; - ma_uint8 playbackData[8192]; - ma_uint32 capturedDataCapInFrames = sizeof(capturedData) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); - ma_uint32 playbackDataCapInFrames = sizeof(playbackData) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); - - ma_uint32 capturedFramesToTryProcessing = ma_min(capturedDataCapInFrames, playbackDataCapInFrames); - ma_uint32 capturedFramesToProcess = (ma_uint32)ma_pcm_converter_read(&pDevice->capture.converter, capturedData, capturedFramesToTryProcessing); - if (capturedFramesToProcess == 0) { - break; /* Don't fire the data callback with zero frames. */ + ma_uint8 capturedClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint8 playbackClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 capturedClientDataCapInFrames = sizeof(capturedClientData) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); + ma_uint32 playbackClientDataCapInFrames = sizeof(playbackClientData) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); + ma_uint64 capturedClientFramesToProcessThisIteration = ma_min(capturedClientDataCapInFrames, playbackClientDataCapInFrames); + ma_uint64 capturedDeviceFramesToProcessThisIteration = capturedDeviceFramesRemaining; + ma_uint8* pRunningCapturedDeviceFrames = ma_offset_ptr(capturedDeviceData, capturedDeviceFramesProcessed * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); + + /* Convert capture data from device format to client format. */ + result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningCapturedDeviceFrames, &capturedDeviceFramesToProcessThisIteration, capturedClientData, &capturedClientFramesToProcessThisIteration); + if (result != MA_SUCCESS) { + break; + } + + /* + If we weren't able to generate any output frames it must mean we've exhaused all of our input. The only time this would not be the case is if capturedClientData was too small + which should never be the case when it's of the size MA_DATA_CONVERTER_STACK_BUFFER_SIZE. + */ + if (capturedClientFramesToProcessThisIteration == 0) { + break; } - ma_device__on_data(pDevice, playbackData, capturedData, capturedFramesToProcess); + ma_device__on_data(pDevice, playbackClientData, capturedClientData, (ma_uint32)capturedClientFramesToProcessThisIteration); /* Safe cast .*/ - /* At this point the playbackData buffer should be holding data that needs to be written to the device. */ - pDevice->playback._dspFrameCount = capturedFramesToProcess; - pDevice->playback._dspFrames = playbackData; + capturedDeviceFramesProcessed += (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */ + capturedDeviceFramesRemaining -= (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */ + + /* At this point the playbackClientData buffer should be holding data that needs to be written to the device. */ for (;;) { - ma_uint32 playbackDeviceFramesCount = (ma_uint32)ma_pcm_converter_read(&pDevice->playback.converter, playbackDeviceData, playbackDeviceDataCapInFrames); - if (playbackDeviceFramesCount == 0) { + ma_uint64 convertedClientFrameCount = capturedClientFramesToProcessThisIteration; + ma_uint64 convertedDeviceFrameCount = playbackDeviceDataCapInFrames; + result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, playbackClientData, &convertedClientFrameCount, playbackDeviceData, &convertedDeviceFrameCount); + if (result != MA_SUCCESS) { break; } - result = ma_device_write__audio4(pDevice, playbackDeviceData, playbackDeviceFramesCount, NULL); + result = ma_device_write__sndio(pDevice, playbackDeviceData, (ma_uint32)convertedDeviceFrameCount, NULL); /* Safe cast. */ if (result != MA_SUCCESS) { exitLoop = MA_TRUE; break; } - if (playbackDeviceFramesCount < playbackDeviceDataCapInFrames) { + capturedClientFramesToProcessThisIteration -= (ma_uint32)convertedClientFrameCount; /* Safe cast. */ + if (capturedClientFramesToProcessThisIteration == 0) { break; } } - if (capturedFramesToProcess < capturedFramesToTryProcessing) { - break; - } - - /* In case an error happened from ma_device_write2__alsa()... */ + /* In case an error happened from ma_device_write__sndio()... */ if (result != MA_SUCCESS) { exitLoop = MA_TRUE; break; } } - totalFramesProcessed += framesProcessed; + totalCapturedDeviceFramesProcessed += capturedDeviceFramesProcessed; } } break; @@ -22438,7 +23592,7 @@ ma_result ma_device_main_loop__audio4(ma_device* pDevice) /* We read in chunks of the period size, but use a stack allocated buffer for the intermediary. */ ma_uint8 intermediaryBuffer[8192]; ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 periodSizeInFrames = pDevice->capture.internalBufferSizeInFrames / pDevice->capture.internalPeriods; + ma_uint32 periodSizeInFrames = pDevice->capture.internalPeriodSizeInFrames; ma_uint32 framesReadThisPeriod = 0; while (framesReadThisPeriod < periodSizeInFrames) { ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesReadThisPeriod; @@ -22448,7 +23602,7 @@ ma_result ma_device_main_loop__audio4(ma_device* pDevice) framesToReadThisIteration = intermediaryBufferSizeInFrames; } - result = ma_device_read__audio4(pDevice, intermediaryBuffer, framesToReadThisIteration, &framesProcessed); + result = ma_device_read__sndio(pDevice, intermediaryBuffer, framesToReadThisIteration, &framesProcessed); if (result != MA_SUCCESS) { exitLoop = MA_TRUE; break; @@ -22465,7 +23619,7 @@ ma_result ma_device_main_loop__audio4(ma_device* pDevice) /* We write in chunks of the period size, but use a stack allocated buffer for the intermediary. */ ma_uint8 intermediaryBuffer[8192]; ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - ma_uint32 periodSizeInFrames = pDevice->playback.internalBufferSizeInFrames / pDevice->playback.internalPeriods; + ma_uint32 periodSizeInFrames = pDevice->playback.internalPeriodSizeInFrames; ma_uint32 framesWrittenThisPeriod = 0; while (framesWrittenThisPeriod < periodSizeInFrames) { ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesWrittenThisPeriod; @@ -22477,7 +23631,7 @@ ma_result ma_device_main_loop__audio4(ma_device* pDevice) ma_device__read_frames_from_client(pDevice, framesToWriteThisIteration, intermediaryBuffer); - result = ma_device_write__audio4(pDevice, intermediaryBuffer, framesToWriteThisIteration, &framesProcessed); + result = ma_device_write__sndio(pDevice, intermediaryBuffer, framesToWriteThisIteration, &framesProcessed); if (result != MA_SUCCESS) { exitLoop = MA_TRUE; break; @@ -22495,566 +23649,819 @@ ma_result ma_device_main_loop__audio4(ma_device* pDevice) /* Here is where the device is stopped. */ - ma_device_stop__audio4(pDevice); + ma_device_stop__sndio(pDevice); return result; } -ma_result ma_context_uninit__audio4(ma_context* pContext) +static ma_result ma_context_uninit__sndio(ma_context* pContext) { - ma_assert(pContext != NULL); - ma_assert(pContext->backend == ma_backend_audio4); + MA_ASSERT(pContext != NULL); + MA_ASSERT(pContext->backend == ma_backend_sndio); (void)pContext; return MA_SUCCESS; } -ma_result ma_context_init__audio4(const ma_context_config* pConfig, ma_context* pContext) +static ma_result ma_context_init__sndio(const ma_context_config* pConfig, ma_context* pContext) { - ma_assert(pContext != NULL); +#ifndef MA_NO_RUNTIME_LINKING + const char* libsndioNames[] = { + "libsndio.so" + }; + size_t i; - (void)pConfig; + for (i = 0; i < ma_countof(libsndioNames); ++i) { + pContext->sndio.sndioSO = ma_dlopen(pContext, libsndioNames[i]); + if (pContext->sndio.sndioSO != NULL) { + break; + } + } - pContext->onUninit = ma_context_uninit__audio4; - pContext->onDeviceIDEqual = ma_context_is_device_id_equal__audio4; - pContext->onEnumDevices = ma_context_enumerate_devices__audio4; - pContext->onGetDeviceInfo = ma_context_get_device_info__audio4; - pContext->onDeviceInit = ma_device_init__audio4; - pContext->onDeviceUninit = ma_device_uninit__audio4; + if (pContext->sndio.sndioSO == NULL) { + return MA_NO_BACKEND; + } + + pContext->sndio.sio_open = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_open"); + pContext->sndio.sio_close = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_close"); + pContext->sndio.sio_setpar = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_setpar"); + pContext->sndio.sio_getpar = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_getpar"); + pContext->sndio.sio_getcap = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_getcap"); + pContext->sndio.sio_write = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_write"); + pContext->sndio.sio_read = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_read"); + pContext->sndio.sio_start = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_start"); + pContext->sndio.sio_stop = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_stop"); + pContext->sndio.sio_initpar = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_initpar"); +#else + pContext->sndio.sio_open = sio_open; + pContext->sndio.sio_close = sio_close; + pContext->sndio.sio_setpar = sio_setpar; + pContext->sndio.sio_getpar = sio_getpar; + pContext->sndio.sio_getcap = sio_getcap; + pContext->sndio.sio_write = sio_write; + pContext->sndio.sio_read = sio_read; + pContext->sndio.sio_start = sio_start; + pContext->sndio.sio_stop = sio_stop; + pContext->sndio.sio_initpar = sio_initpar; +#endif + + pContext->onUninit = ma_context_uninit__sndio; + pContext->onDeviceIDEqual = ma_context_is_device_id_equal__sndio; + pContext->onEnumDevices = ma_context_enumerate_devices__sndio; + pContext->onGetDeviceInfo = ma_context_get_device_info__sndio; + pContext->onDeviceInit = ma_device_init__sndio; + pContext->onDeviceUninit = ma_device_uninit__sndio; pContext->onDeviceStart = NULL; /* Not required for synchronous backends. */ pContext->onDeviceStop = NULL; /* Not required for synchronous backends. */ - pContext->onDeviceMainLoop = ma_device_main_loop__audio4; + pContext->onDeviceMainLoop = ma_device_main_loop__sndio; + (void)pConfig; return MA_SUCCESS; } -#endif /* audio4 */ +#endif /* sndio */ + /****************************************************************************** -OSS Backend +audio(4) Backend ******************************************************************************/ -#ifdef MA_HAS_OSS -#include -#include +#ifdef MA_HAS_AUDIO4 #include -#include +#include +#include +#include +#include +#include +#include -#ifndef SNDCTL_DSP_HALT -#define SNDCTL_DSP_HALT SNDCTL_DSP_RESET +#if defined(__OpenBSD__) + #include + #if defined(OpenBSD) && OpenBSD >= 201709 + #define MA_AUDIO4_USE_NEW_API + #endif #endif -int ma_open_temp_device__oss() +static void ma_construct_device_id__audio4(char* id, size_t idSize, const char* base, int deviceIndex) { - /* The OSS sample code uses "/dev/mixer" as the device for getting system properties so I'm going to do the same. */ - int fd = open("/dev/mixer", O_RDONLY, 0); - if (fd >= 0) { - return fd; - } + size_t baseLen; - return -1; + MA_ASSERT(id != NULL); + MA_ASSERT(idSize > 0); + MA_ASSERT(deviceIndex >= 0); + + baseLen = strlen(base); + MA_ASSERT(idSize > baseLen); + + ma_strcpy_s(id, idSize, base); + ma_itoa_s(deviceIndex, id+baseLen, idSize-baseLen, 10); } -ma_result ma_context_open_device__oss(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, int* pfd) +static ma_result ma_extract_device_index_from_id__audio4(const char* id, const char* base, int* pIndexOut) { - const char* deviceName; - int flags; - - ma_assert(pContext != NULL); - ma_assert(pfd != NULL); - (void)pContext; - - *pfd = -1; + size_t idLen; + size_t baseLen; + const char* deviceIndexStr; - /* This function should only be called for playback or capture, not duplex. */ - if (deviceType == ma_device_type_duplex) { - return MA_INVALID_ARGS; + MA_ASSERT(id != NULL); + MA_ASSERT(base != NULL); + MA_ASSERT(pIndexOut != NULL); + + idLen = strlen(id); + baseLen = strlen(base); + if (idLen <= baseLen) { + return MA_ERROR; /* Doesn't look like the id starts with the base. */ } - - deviceName = "/dev/dsp"; - if (pDeviceID != NULL) { - deviceName = pDeviceID->oss; + + if (strncmp(id, base, baseLen) != 0) { + return MA_ERROR; /* ID does not begin with base. */ } - - flags = (deviceType == ma_device_type_playback) ? O_WRONLY : O_RDONLY; - if (shareMode == ma_share_mode_exclusive) { - flags |= O_EXCL; + + deviceIndexStr = id + baseLen; + if (deviceIndexStr[0] == '\0') { + return MA_ERROR; /* No index specified in the ID. */ } - - *pfd = open(deviceName, flags, 0); - if (*pfd == -1) { - return MA_FAILED_TO_OPEN_BACKEND_DEVICE; + + if (pIndexOut) { + *pIndexOut = atoi(deviceIndexStr); } - + return MA_SUCCESS; } -ma_bool32 ma_context_is_device_id_equal__oss(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) +static ma_bool32 ma_context_is_device_id_equal__audio4(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) { - ma_assert(pContext != NULL); - ma_assert(pID0 != NULL); - ma_assert(pID1 != NULL); + MA_ASSERT(pContext != NULL); + MA_ASSERT(pID0 != NULL); + MA_ASSERT(pID1 != NULL); (void)pContext; - return ma_strcmp(pID0->oss, pID1->oss) == 0; + return ma_strcmp(pID0->audio4, pID1->audio4) == 0; } -ma_result ma_context_enumerate_devices__oss(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +#if !defined(MA_AUDIO4_USE_NEW_API) /* Old API */ +static ma_format ma_format_from_encoding__audio4(unsigned int encoding, unsigned int precision) { - int fd; - oss_sysinfo si; - int result; - - ma_assert(pContext != NULL); - ma_assert(callback != NULL); - - fd = ma_open_temp_device__oss(); - if (fd == -1) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[OSS] Failed to open a temporary device for retrieving system information used for device enumeration.", MA_NO_BACKEND); - } - - result = ioctl(fd, SNDCTL_SYSINFO, &si); - if (result != -1) { - int iAudioDevice; - for (iAudioDevice = 0; iAudioDevice < si.numaudios; ++iAudioDevice) { - oss_audioinfo ai; - ai.dev = iAudioDevice; - result = ioctl(fd, SNDCTL_AUDIOINFO, &ai); - if (result != -1) { - if (ai.devnode[0] != '\0') { /* <-- Can be blank, according to documentation. */ - ma_device_info deviceInfo; - ma_bool32 isTerminating = MA_FALSE; - - ma_zero_object(&deviceInfo); - - /* ID */ - ma_strncpy_s(deviceInfo.id.oss, sizeof(deviceInfo.id.oss), ai.devnode, (size_t)-1); - - /* - The human readable device name should be in the "ai.handle" variable, but it can - sometimes be empty in which case we just fall back to "ai.name" which is less user - friendly, but usually has a value. - */ - if (ai.handle[0] != '\0') { - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), ai.handle, (size_t)-1); - } else { - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), ai.name, (size_t)-1); - } - - /* The device can be both playback and capture. */ - if (!isTerminating && (ai.caps & PCM_CAP_OUTPUT) != 0) { - isTerminating = !callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); - } - if (!isTerminating && (ai.caps & PCM_CAP_INPUT) != 0) { - isTerminating = !callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); - } - - if (isTerminating) { - break; - } - } + if (precision == 8 && (encoding == AUDIO_ENCODING_ULINEAR || encoding == AUDIO_ENCODING_ULINEAR || encoding == AUDIO_ENCODING_ULINEAR_LE || encoding == AUDIO_ENCODING_ULINEAR_BE)) { + return ma_format_u8; + } else { + if (ma_is_little_endian() && encoding == AUDIO_ENCODING_SLINEAR_LE) { + if (precision == 16) { + return ma_format_s16; + } else if (precision == 24) { + return ma_format_s24; + } else if (precision == 32) { + return ma_format_s32; + } + } else if (ma_is_big_endian() && encoding == AUDIO_ENCODING_SLINEAR_BE) { + if (precision == 16) { + return ma_format_s16; + } else if (precision == 24) { + return ma_format_s24; + } else if (precision == 32) { + return ma_format_s32; } } - } else { - close(fd); - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[OSS] Failed to retrieve system information for device enumeration.", MA_NO_BACKEND); } - close(fd); - return MA_SUCCESS; + return ma_format_unknown; /* Encoding not supported. */ } -ma_result ma_context_get_device_info__oss(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) +static void ma_encoding_from_format__audio4(ma_format format, unsigned int* pEncoding, unsigned int* pPrecision) { - ma_bool32 foundDevice; - int fdTemp; - oss_sysinfo si; - int result; + MA_ASSERT(format != ma_format_unknown); + MA_ASSERT(pEncoding != NULL); + MA_ASSERT(pPrecision != NULL); - ma_assert(pContext != NULL); - (void)shareMode; + switch (format) + { + case ma_format_u8: + { + *pEncoding = AUDIO_ENCODING_ULINEAR; + *pPrecision = 8; + } break; - /* Handle the default device a little differently. */ - if (pDeviceID == NULL) { - if (deviceType == ma_device_type_playback) { - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); - } else { - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); - } + case ma_format_s24: + { + *pEncoding = (ma_is_little_endian()) ? AUDIO_ENCODING_SLINEAR_LE : AUDIO_ENCODING_SLINEAR_BE; + *pPrecision = 24; + } break; - return MA_SUCCESS; + case ma_format_s32: + { + *pEncoding = (ma_is_little_endian()) ? AUDIO_ENCODING_SLINEAR_LE : AUDIO_ENCODING_SLINEAR_BE; + *pPrecision = 32; + } break; + + case ma_format_s16: + case ma_format_f32: + default: + { + *pEncoding = (ma_is_little_endian()) ? AUDIO_ENCODING_SLINEAR_LE : AUDIO_ENCODING_SLINEAR_BE; + *pPrecision = 16; + } break; + } +} + +static ma_format ma_format_from_prinfo__audio4(struct audio_prinfo* prinfo) +{ + return ma_format_from_encoding__audio4(prinfo->encoding, prinfo->precision); +} +#else +static ma_format ma_format_from_swpar__audio4(struct audio_swpar* par) +{ + if (par->bits == 8 && par->bps == 1 && par->sig == 0) { + return ma_format_u8; + } + if (par->bits == 16 && par->bps == 2 && par->sig == 1 && par->le == ma_is_little_endian()) { + return ma_format_s16; + } + if (par->bits == 24 && par->bps == 3 && par->sig == 1 && par->le == ma_is_little_endian()) { + return ma_format_s24; + } + if (par->bits == 32 && par->bps == 4 && par->sig == 1 && par->le == ma_is_little_endian()) { + return ma_format_f32; } + /* Format not supported. */ + return ma_format_unknown; +} +#endif - /* If we get here it means we are _not_ using the default device. */ - foundDevice = MA_FALSE; +static ma_result ma_context_get_device_info_from_fd__audio4(ma_context* pContext, ma_device_type deviceType, int fd, ma_device_info* pInfoOut) +{ + audio_device_t fdDevice; +#if !defined(MA_AUDIO4_USE_NEW_API) + int counter = 0; + audio_info_t fdInfo; +#else + struct audio_swpar fdPar; + ma_format format; +#endif - fdTemp = ma_open_temp_device__oss(); - if (fdTemp == -1) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[OSS] Failed to open a temporary device for retrieving system information used for device enumeration.", MA_NO_BACKEND); + MA_ASSERT(pContext != NULL); + MA_ASSERT(fd >= 0); + MA_ASSERT(pInfoOut != NULL); + + (void)pContext; + (void)deviceType; + + if (ioctl(fd, AUDIO_GETDEV, &fdDevice) < 0) { + return MA_ERROR; /* Failed to retrieve device info. */ } - result = ioctl(fdTemp, SNDCTL_SYSINFO, &si); - if (result != -1) { - int iAudioDevice; - for (iAudioDevice = 0; iAudioDevice < si.numaudios; ++iAudioDevice) { - oss_audioinfo ai; - ai.dev = iAudioDevice; - result = ioctl(fdTemp, SNDCTL_AUDIOINFO, &ai); - if (result != -1) { - if (ma_strcmp(ai.devnode, pDeviceID->oss) == 0) { - /* It has the same name, so now just confirm the type. */ - if ((deviceType == ma_device_type_playback && ((ai.caps & PCM_CAP_OUTPUT) != 0)) || - (deviceType == ma_device_type_capture && ((ai.caps & PCM_CAP_INPUT) != 0))) { - unsigned int formatMask; + /* Name. */ + ma_strcpy_s(pInfoOut->name, sizeof(pInfoOut->name), fdDevice.name); - /* ID */ - ma_strncpy_s(pDeviceInfo->id.oss, sizeof(pDeviceInfo->id.oss), ai.devnode, (size_t)-1); +#if !defined(MA_AUDIO4_USE_NEW_API) + /* Supported formats. We get this by looking at the encodings. */ + for (;;) { + audio_encoding_t encoding; + ma_format format; - /* - The human readable device name should be in the "ai.handle" variable, but it can - sometimes be empty in which case we just fall back to "ai.name" which is less user - friendly, but usually has a value. - */ - if (ai.handle[0] != '\0') { - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), ai.handle, (size_t)-1); - } else { - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), ai.name, (size_t)-1); - } + MA_ZERO_OBJECT(&encoding); + encoding.index = counter; + if (ioctl(fd, AUDIO_GETENC, &encoding) < 0) { + break; + } - pDeviceInfo->minChannels = ai.min_channels; - pDeviceInfo->maxChannels = ai.max_channels; - pDeviceInfo->minSampleRate = ai.min_rate; - pDeviceInfo->maxSampleRate = ai.max_rate; - pDeviceInfo->formatCount = 0; + format = ma_format_from_encoding__audio4(encoding.encoding, encoding.precision); + if (format != ma_format_unknown) { + pInfoOut->formats[pInfoOut->formatCount++] = format; + } - if (deviceType == ma_device_type_playback) { - formatMask = ai.oformats; - } else { - formatMask = ai.iformats; - } + counter += 1; + } - if ((formatMask & AFMT_U8) != 0) { - pDeviceInfo->formats[pDeviceInfo->formatCount++] = ma_format_u8; - } - if (((formatMask & AFMT_S16_LE) != 0 && ma_is_little_endian()) || (AFMT_S16_BE && ma_is_big_endian())) { - pDeviceInfo->formats[pDeviceInfo->formatCount++] = ma_format_s16; - } - if (((formatMask & AFMT_S32_LE) != 0 && ma_is_little_endian()) || (AFMT_S32_BE && ma_is_big_endian())) { - pDeviceInfo->formats[pDeviceInfo->formatCount++] = ma_format_s32; - } + if (ioctl(fd, AUDIO_GETINFO, &fdInfo) < 0) { + return MA_ERROR; + } - foundDevice = MA_TRUE; - break; - } - } - } - } + if (deviceType == ma_device_type_playback) { + pInfoOut->minChannels = fdInfo.play.channels; + pInfoOut->maxChannels = fdInfo.play.channels; + pInfoOut->minSampleRate = fdInfo.play.sample_rate; + pInfoOut->maxSampleRate = fdInfo.play.sample_rate; } else { - close(fdTemp); - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[OSS] Failed to retrieve system information for device enumeration.", MA_NO_BACKEND); + pInfoOut->minChannels = fdInfo.record.channels; + pInfoOut->maxChannels = fdInfo.record.channels; + pInfoOut->minSampleRate = fdInfo.record.sample_rate; + pInfoOut->maxSampleRate = fdInfo.record.sample_rate; + } +#else + if (ioctl(fd, AUDIO_GETPAR, &fdPar) < 0) { + return MA_ERROR; + } + + format = ma_format_from_swpar__audio4(&fdPar); + if (format == ma_format_unknown) { + return MA_FORMAT_NOT_SUPPORTED; + } + pInfoOut->formats[pInfoOut->formatCount++] = format; + + if (deviceType == ma_device_type_playback) { + pInfoOut->minChannels = fdPar.pchan; + pInfoOut->maxChannels = fdPar.pchan; + } else { + pInfoOut->minChannels = fdPar.rchan; + pInfoOut->maxChannels = fdPar.rchan; } + + pInfoOut->minSampleRate = fdPar.rate; + pInfoOut->maxSampleRate = fdPar.rate; +#endif + + return MA_SUCCESS; +} +static ma_result ma_context_enumerate_devices__audio4(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +{ + const int maxDevices = 64; + char devpath[256]; + int iDevice; - close(fdTemp); + MA_ASSERT(pContext != NULL); + MA_ASSERT(callback != NULL); + + /* + Every device will be named "/dev/audioN", with a "/dev/audioctlN" equivalent. We use the "/dev/audioctlN" + version here since we can open it even when another process has control of the "/dev/audioN" device. + */ + for (iDevice = 0; iDevice < maxDevices; ++iDevice) { + struct stat st; + int fd; + ma_bool32 isTerminating = MA_FALSE; - if (!foundDevice) { - return MA_NO_DEVICE; + ma_strcpy_s(devpath, sizeof(devpath), "/dev/audioctl"); + ma_itoa_s(iDevice, devpath+strlen(devpath), sizeof(devpath)-strlen(devpath), 10); + + if (stat(devpath, &st) < 0) { + break; + } + + /* The device exists, but we need to check if it's usable as playback and/or capture. */ + + /* Playback. */ + if (!isTerminating) { + fd = open(devpath, O_RDONLY, 0); + if (fd >= 0) { + /* Supports playback. */ + ma_device_info deviceInfo; + MA_ZERO_OBJECT(&deviceInfo); + ma_construct_device_id__audio4(deviceInfo.id.audio4, sizeof(deviceInfo.id.audio4), "/dev/audio", iDevice); + if (ma_context_get_device_info_from_fd__audio4(pContext, ma_device_type_playback, fd, &deviceInfo) == MA_SUCCESS) { + isTerminating = !callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); + } + + close(fd); + } + } + + /* Capture. */ + if (!isTerminating) { + fd = open(devpath, O_WRONLY, 0); + if (fd >= 0) { + /* Supports capture. */ + ma_device_info deviceInfo; + MA_ZERO_OBJECT(&deviceInfo); + ma_construct_device_id__audio4(deviceInfo.id.audio4, sizeof(deviceInfo.id.audio4), "/dev/audio", iDevice); + if (ma_context_get_device_info_from_fd__audio4(pContext, ma_device_type_capture, fd, &deviceInfo) == MA_SUCCESS) { + isTerminating = !callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); + } + + close(fd); + } + } + + if (isTerminating) { + break; + } } - + return MA_SUCCESS; } -void ma_device_uninit__oss(ma_device* pDevice) +static ma_result ma_context_get_device_info__audio4(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) { - ma_assert(pDevice != NULL); + int fd = -1; + int deviceIndex = -1; + char ctlid[256]; + ma_result result; - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - close(pDevice->oss.fdCapture); + MA_ASSERT(pContext != NULL); + (void)shareMode; + + /* + We need to open the "/dev/audioctlN" device to get the info. To do this we need to extract the number + from the device ID which will be in "/dev/audioN" format. + */ + if (pDeviceID == NULL) { + /* Default device. */ + ma_strcpy_s(ctlid, sizeof(ctlid), "/dev/audioctl"); + } else { + /* Specific device. We need to convert from "/dev/audioN" to "/dev/audioctlN". */ + result = ma_extract_device_index_from_id__audio4(pDeviceID->audio4, "/dev/audio", &deviceIndex); + if (result != MA_SUCCESS) { + return result; + } + + ma_construct_device_id__audio4(ctlid, sizeof(ctlid), "/dev/audioctl", deviceIndex); } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - close(pDevice->oss.fdPlayback); + fd = open(ctlid, (deviceType == ma_device_type_playback) ? O_WRONLY : O_RDONLY, 0); + if (fd == -1) { + return MA_NO_DEVICE; } -} - -int ma_format_to_oss(ma_format format) -{ - int ossFormat = AFMT_U8; - switch (format) { - case ma_format_s16: ossFormat = (ma_is_little_endian()) ? AFMT_S16_LE : AFMT_S16_BE; break; - case ma_format_s24: ossFormat = (ma_is_little_endian()) ? AFMT_S32_LE : AFMT_S32_BE; break; - case ma_format_s32: ossFormat = (ma_is_little_endian()) ? AFMT_S32_LE : AFMT_S32_BE; break; - case ma_format_f32: ossFormat = (ma_is_little_endian()) ? AFMT_S16_LE : AFMT_S16_BE; break; - case ma_format_u8: - default: ossFormat = AFMT_U8; break; + + if (deviceIndex == -1) { + ma_strcpy_s(pDeviceInfo->id.audio4, sizeof(pDeviceInfo->id.audio4), "/dev/audio"); + } else { + ma_construct_device_id__audio4(pDeviceInfo->id.audio4, sizeof(pDeviceInfo->id.audio4), "/dev/audio", deviceIndex); } - - return ossFormat; + + result = ma_context_get_device_info_from_fd__audio4(pContext, deviceType, fd, pDeviceInfo); + + close(fd); + return result; } -ma_format ma_format_from_oss(int ossFormat) +static void ma_device_uninit__audio4(ma_device* pDevice) { - if (ossFormat == AFMT_U8) { - return ma_format_u8; - } else { - if (ma_is_little_endian()) { - switch (ossFormat) { - case AFMT_S16_LE: return ma_format_s16; - case AFMT_S32_LE: return ma_format_s32; - default: return ma_format_unknown; - } - } else { - switch (ossFormat) { - case AFMT_S16_BE: return ma_format_s16; - case AFMT_S32_BE: return ma_format_s32; - default: return ma_format_unknown; - } - } + MA_ASSERT(pDevice != NULL); + + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + close(pDevice->audio4.fdCapture); } - return ma_format_unknown; + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + close(pDevice->audio4.fdPlayback); + } } -ma_result ma_device_init_fd__oss(ma_context* pContext, const ma_device_config* pConfig, ma_device_type deviceType, ma_device* pDevice) +static ma_result ma_device_init_fd__audio4(ma_context* pContext, const ma_device_config* pConfig, ma_device_type deviceType, ma_device* pDevice) { - ma_result result; - int ossResult; + const char* pDefaultDeviceNames[] = { + "/dev/audio", + "/dev/audio0" + }; int fd; - const ma_device_id* pDeviceID = NULL; - ma_share_mode shareMode; - int ossFormat; - int ossChannels; - int ossSampleRate; - int ossFragment; + int fdFlags = 0; +#if !defined(MA_AUDIO4_USE_NEW_API) /* Old API */ + audio_info_t fdInfo; +#else + struct audio_swpar fdPar; +#endif + ma_format internalFormat; + ma_uint32 internalChannels; + ma_uint32 internalSampleRate; + ma_uint32 internalPeriodSizeInFrames; + ma_uint32 internalPeriods; - ma_assert(pContext != NULL); - ma_assert(pConfig != NULL); - ma_assert(deviceType != ma_device_type_duplex); - ma_assert(pDevice != NULL); + MA_ASSERT(pContext != NULL); + MA_ASSERT(pConfig != NULL); + MA_ASSERT(deviceType != ma_device_type_duplex); + MA_ASSERT(pDevice != NULL); (void)pContext; + /* The first thing to do is open the file. */ if (deviceType == ma_device_type_capture) { - pDeviceID = pConfig->capture.pDeviceID; - shareMode = pConfig->capture.shareMode; - ossFormat = ma_format_to_oss(pConfig->capture.format); - ossChannels = (int)pConfig->capture.channels; - ossSampleRate = (int)pConfig->sampleRate; + fdFlags = O_RDONLY; } else { - pDeviceID = pConfig->playback.pDeviceID; - shareMode = pConfig->playback.shareMode; - ossFormat = ma_format_to_oss(pConfig->playback.format); - ossChannels = (int)pConfig->playback.channels; - ossSampleRate = (int)pConfig->sampleRate; + fdFlags = O_WRONLY; } + /*fdFlags |= O_NONBLOCK;*/ - result = ma_context_open_device__oss(pContext, deviceType, pDeviceID, shareMode, &fd); - if (result != MA_SUCCESS) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to open device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + if ((deviceType == ma_device_type_capture && pConfig->capture.pDeviceID == NULL) || (deviceType == ma_device_type_playback && pConfig->playback.pDeviceID == NULL)) { + /* Default device. */ + size_t iDevice; + for (iDevice = 0; iDevice < ma_countof(pDefaultDeviceNames); ++iDevice) { + fd = open(pDefaultDeviceNames[iDevice], fdFlags, 0); + if (fd != -1) { + break; + } + } + } else { + /* Specific device. */ + fd = open((deviceType == ma_device_type_capture) ? pConfig->capture.pDeviceID->audio4 : pConfig->playback.pDeviceID->audio4, fdFlags, 0); } - /* - The OSS documantation is very clear about the order we should be initializing the device's properties: - 1) Format - 2) Channels - 3) Sample rate. - */ + if (fd == -1) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to open device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - /* Format. */ - ossResult = ioctl(fd, SNDCTL_DSP_SETFMT, &ossFormat); - if (ossResult == -1) { +#if !defined(MA_AUDIO4_USE_NEW_API) /* Old API */ + AUDIO_INITINFO(&fdInfo); + + /* We get the driver to do as much of the data conversion as possible. */ + if (deviceType == ma_device_type_capture) { + fdInfo.mode = AUMODE_RECORD; + ma_encoding_from_format__audio4(pConfig->capture.format, &fdInfo.record.encoding, &fdInfo.record.precision); + fdInfo.record.channels = pConfig->capture.channels; + fdInfo.record.sample_rate = pConfig->sampleRate; + } else { + fdInfo.mode = AUMODE_PLAY; + ma_encoding_from_format__audio4(pConfig->playback.format, &fdInfo.play.encoding, &fdInfo.play.precision); + fdInfo.play.channels = pConfig->playback.channels; + fdInfo.play.sample_rate = pConfig->sampleRate; + } + + if (ioctl(fd, AUDIO_SETINFO, &fdInfo) < 0) { close(fd); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to set format.", MA_FORMAT_NOT_SUPPORTED); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to set device format. AUDIO_SETINFO failed.", MA_FORMAT_NOT_SUPPORTED); + } + + if (ioctl(fd, AUDIO_GETINFO, &fdInfo) < 0) { + close(fd); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] AUDIO_GETINFO failed.", MA_FORMAT_NOT_SUPPORTED); } - /* Channels. */ - ossResult = ioctl(fd, SNDCTL_DSP_CHANNELS, &ossChannels); - if (ossResult == -1) { + if (deviceType == ma_device_type_capture) { + internalFormat = ma_format_from_prinfo__audio4(&fdInfo.record); + internalChannels = fdInfo.record.channels; + internalSampleRate = fdInfo.record.sample_rate; + } else { + internalFormat = ma_format_from_prinfo__audio4(&fdInfo.play); + internalChannels = fdInfo.play.channels; + internalSampleRate = fdInfo.play.sample_rate; + } + + if (internalFormat == ma_format_unknown) { close(fd); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to set channel count.", MA_FORMAT_NOT_SUPPORTED); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] The device's internal device format is not supported by miniaudio. The device is unusable.", MA_FORMAT_NOT_SUPPORTED); } - /* Sample Rate. */ - ossResult = ioctl(fd, SNDCTL_DSP_SPEED, &ossSampleRate); - if (ossResult == -1) { + /* Buffer. */ + { + ma_uint32 internalPeriodSizeInBytes; + + internalPeriodSizeInFrames = pConfig->periodSizeInFrames; + if (internalPeriodSizeInFrames == 0) { + internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->periodSizeInMilliseconds, internalSampleRate); + } + + internalPeriodSizeInBytes = internalPeriodSizeInFrames * ma_get_bytes_per_frame(internalFormat, internalChannels); + if (internalPeriodSizeInBytes < 16) { + internalPeriodSizeInBytes = 16; + } + + internalPeriods = pConfig->periods; + if (internalPeriods < 2) { + internalPeriods = 2; + } + + /* What miniaudio calls a period, audio4 calls a block. */ + AUDIO_INITINFO(&fdInfo); + fdInfo.hiwat = internalPeriods; + fdInfo.lowat = internalPeriods-1; + fdInfo.blocksize = internalPeriodSizeInBytes; + if (ioctl(fd, AUDIO_SETINFO, &fdInfo) < 0) { + close(fd); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to set internal buffer size. AUDIO_SETINFO failed.", MA_FORMAT_NOT_SUPPORTED); + } + + internalPeriods = fdInfo.hiwat; + internalPeriodSizeInFrames = fdInfo.blocksize / ma_get_bytes_per_frame(internalFormat, internalChannels); + } +#else + /* We need to retrieve the format of the device so we can know the channel count and sample rate. Then we can calculate the buffer size. */ + if (ioctl(fd, AUDIO_GETPAR, &fdPar) < 0) { close(fd); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to set sample rate.", MA_FORMAT_NOT_SUPPORTED); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to retrieve initial device parameters.", MA_FORMAT_NOT_SUPPORTED); } - /* - Buffer. + internalFormat = ma_format_from_swpar__audio4(&fdPar); + internalChannels = (deviceType == ma_device_type_capture) ? fdPar.rchan : fdPar.pchan; + internalSampleRate = fdPar.rate; - The documentation says that the fragment settings should be set as soon as possible, but I'm not sure if - it should be done before or after format/channels/rate. - - OSS wants the fragment size in bytes and a power of 2. When setting, we specify the power, not the actual - value. - */ + if (internalFormat == ma_format_unknown) { + close(fd); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] The device's internal device format is not supported by miniaudio. The device is unusable.", MA_FORMAT_NOT_SUPPORTED); + } + + /* Buffer. */ { - ma_uint32 fragmentSizeInBytes; - ma_uint32 bufferSizeInFrames; - ma_uint32 ossFragmentSizePower; - - bufferSizeInFrames = pConfig->bufferSizeInFrames; - if (bufferSizeInFrames == 0) { - bufferSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->bufferSizeInMilliseconds, (ma_uint32)ossSampleRate); - } + ma_uint32 internalPeriodSizeInBytes; - fragmentSizeInBytes = ma_round_to_power_of_2((bufferSizeInFrames / pConfig->periods) * ma_get_bytes_per_frame(ma_format_from_oss(ossFormat), ossChannels)); - if (fragmentSizeInBytes < 16) { - fragmentSizeInBytes = 16; + internalPeriodSizeInFrames = pConfig->periodSizeInFrames; + if (internalPeriodSizeInFrames == 0) { + internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->periodSizeInMilliseconds, internalSampleRate); } - ossFragmentSizePower = 4; - fragmentSizeInBytes >>= 4; - while (fragmentSizeInBytes >>= 1) { - ossFragmentSizePower += 1; + /* What miniaudio calls a period, audio4 calls a block. */ + internalPeriodSizeInBytes = internalPeriodSizeInFrames * ma_get_bytes_per_frame(internalFormat, internalChannels); + if (internalPeriodSizeInBytes < 16) { + internalPeriodSizeInBytes = 16; + } + + fdPar.nblks = pConfig->periods; + fdPar.round = internalPeriodSizeInBytes; + + if (ioctl(fd, AUDIO_SETPAR, &fdPar) < 0) { + close(fd); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to set device parameters.", MA_FORMAT_NOT_SUPPORTED); } - ossFragment = (int)((pConfig->periods << 16) | ossFragmentSizePower); - ossResult = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &ossFragment); - if (ossResult == -1) { + if (ioctl(fd, AUDIO_GETPAR, &fdPar) < 0) { close(fd); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to set fragment size and period count.", MA_FORMAT_NOT_SUPPORTED); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to retrieve actual device parameters.", MA_FORMAT_NOT_SUPPORTED); } } - /* Internal settings. */ + internalFormat = ma_format_from_swpar__audio4(&fdPar); + internalChannels = (deviceType == ma_device_type_capture) ? fdPar.rchan : fdPar.pchan; + internalSampleRate = fdPar.rate; + internalPeriods = fdPar.nblks; + internalPeriodSizeInFrames = fdPar.round / ma_get_bytes_per_frame(internalFormat, internalChannels); +#endif + + if (internalFormat == ma_format_unknown) { + close(fd); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] The device's internal device format is not supported by miniaudio. The device is unusable.", MA_FORMAT_NOT_SUPPORTED); + } + if (deviceType == ma_device_type_capture) { - pDevice->oss.fdCapture = fd; - pDevice->capture.internalFormat = ma_format_from_oss(ossFormat); - pDevice->capture.internalChannels = ossChannels; - pDevice->capture.internalSampleRate = ossSampleRate; - ma_get_standard_channel_map(ma_standard_channel_map_sound4, pDevice->capture.internalChannels, pDevice->capture.internalChannelMap); - pDevice->capture.internalPeriods = (ma_uint32)(ossFragment >> 16); - pDevice->capture.internalBufferSizeInFrames = (((ma_uint32)(1 << (ossFragment & 0xFFFF))) * pDevice->capture.internalPeriods) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - - if (pDevice->capture.internalFormat == ma_format_unknown) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] The device's internal format is not supported by miniaudio.", MA_FORMAT_NOT_SUPPORTED); - } + pDevice->audio4.fdCapture = fd; + pDevice->capture.internalFormat = internalFormat; + pDevice->capture.internalChannels = internalChannels; + pDevice->capture.internalSampleRate = internalSampleRate; + ma_get_standard_channel_map(ma_standard_channel_map_sound4, internalChannels, pDevice->capture.internalChannelMap); + pDevice->capture.internalPeriodSizeInFrames = internalPeriodSizeInFrames; + pDevice->capture.internalPeriods = internalPeriods; } else { - pDevice->oss.fdPlayback = fd; - pDevice->playback.internalFormat = ma_format_from_oss(ossFormat); - pDevice->playback.internalChannels = ossChannels; - pDevice->playback.internalSampleRate = ossSampleRate; - ma_get_standard_channel_map(ma_standard_channel_map_sound4, pDevice->playback.internalChannels, pDevice->playback.internalChannelMap); - pDevice->playback.internalPeriods = (ma_uint32)(ossFragment >> 16); - pDevice->playback.internalBufferSizeInFrames = (((ma_uint32)(1 << (ossFragment & 0xFFFF))) * pDevice->playback.internalPeriods) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - - if (pDevice->playback.internalFormat == ma_format_unknown) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] The device's internal format is not supported by miniaudio.", MA_FORMAT_NOT_SUPPORTED); - } + pDevice->audio4.fdPlayback = fd; + pDevice->playback.internalFormat = internalFormat; + pDevice->playback.internalChannels = internalChannels; + pDevice->playback.internalSampleRate = internalSampleRate; + ma_get_standard_channel_map(ma_standard_channel_map_sound4, internalChannels, pDevice->playback.internalChannelMap); + pDevice->playback.internalPeriodSizeInFrames = internalPeriodSizeInFrames; + pDevice->playback.internalPeriods = internalPeriods; } return MA_SUCCESS; } -ma_result ma_device_init__oss(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) +static ma_result ma_device_init__audio4(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) { - ma_assert(pContext != NULL); - ma_assert(pConfig != NULL); - ma_assert(pDevice != NULL); + MA_ASSERT(pDevice != NULL); - ma_zero_object(&pDevice->oss); + MA_ZERO_OBJECT(&pDevice->audio4); if (pConfig->deviceType == ma_device_type_loopback) { return MA_DEVICE_TYPE_NOT_SUPPORTED; } + + pDevice->audio4.fdCapture = -1; + pDevice->audio4.fdPlayback = -1; + + /* + The version of the operating system dictates whether or not the device is exclusive or shared. NetBSD + introduced in-kernel mixing which means it's shared. All other BSD flavours are exclusive as far as + I'm aware. + */ +#if defined(__NetBSD_Version__) && __NetBSD_Version__ >= 800000000 + /* NetBSD 8.0+ */ + if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) || + ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) { + return MA_SHARE_MODE_NOT_SUPPORTED; + } +#else + /* All other flavors. */ +#endif if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { - ma_result result = ma_device_init_fd__oss(pContext, pConfig, ma_device_type_capture, pDevice); + ma_result result = ma_device_init_fd__audio4(pContext, pConfig, ma_device_type_capture, pDevice); if (result != MA_SUCCESS) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to open device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + return result; } } if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { - ma_result result = ma_device_init_fd__oss(pContext, pConfig, ma_device_type_playback, pDevice); + ma_result result = ma_device_init_fd__audio4(pContext, pConfig, ma_device_type_playback, pDevice); if (result != MA_SUCCESS) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to open device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + if (pConfig->deviceType == ma_device_type_duplex) { + close(pDevice->audio4.fdCapture); + } + return result; } } return MA_SUCCESS; } -ma_result ma_device_stop__oss(ma_device* pDevice) +#if 0 +static ma_result ma_device_start__audio4(ma_device* pDevice) { - ma_assert(pDevice != NULL); + MA_ASSERT(pDevice != NULL); - /* - We want to use SNDCTL_DSP_HALT. From the documentation: - - In multithreaded applications SNDCTL_DSP_HALT (SNDCTL_DSP_RESET) must only be called by the thread - that actually reads/writes the audio device. It must not be called by some master thread to kill the - audio thread. The audio thread will not stop or get any kind of notification that the device was - stopped by the master thread. The device gets stopped but the next read or write call will silently - restart the device. - - This is actually safe in our case, because this function is only ever called from within our worker - thread anyway. Just keep this in mind, though... - */ + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + if (pDevice->audio4.fdCapture == -1) { + return MA_INVALID_ARGS; + } + } + + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + if (pDevice->audio4.fdPlayback == -1) { + return MA_INVALID_ARGS; + } + } + + return MA_SUCCESS; +} +#endif + +static ma_result ma_device_stop_fd__audio4(ma_device* pDevice, int fd) +{ + if (fd == -1) { + return MA_INVALID_ARGS; + } + +#if !defined(MA_AUDIO4_USE_NEW_API) + if (ioctl(fd, AUDIO_FLUSH, 0) < 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to stop device. AUDIO_FLUSH failed.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + } +#else + if (ioctl(fd, AUDIO_STOP, 0) < 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to stop device. AUDIO_STOP failed.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + } +#endif + + return MA_SUCCESS; +} + +static ma_result ma_device_stop__audio4(ma_device* pDevice) +{ + MA_ASSERT(pDevice != NULL); if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - int result = ioctl(pDevice->oss.fdCapture, SNDCTL_DSP_HALT, 0); - if (result == -1) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to stop device. SNDCTL_DSP_HALT failed.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + ma_result result; + + result = ma_device_stop_fd__audio4(pDevice, pDevice->audio4.fdCapture); + if (result != MA_SUCCESS) { + return result; } } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - int result = ioctl(pDevice->oss.fdPlayback, SNDCTL_DSP_HALT, 0); - if (result == -1) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to stop device. SNDCTL_DSP_HALT failed.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + ma_result result; + + /* Drain the device first. If this fails we'll just need to flush without draining. Unfortunately draining isn't available on newer version of OpenBSD. */ + #if !defined(MA_AUDIO4_USE_NEW_API) + ioctl(pDevice->audio4.fdPlayback, AUDIO_DRAIN, 0); + #endif + + /* Here is where the device is stopped immediately. */ + result = ma_device_stop_fd__audio4(pDevice, pDevice->audio4.fdPlayback); + if (result != MA_SUCCESS) { + return result; } } return MA_SUCCESS; } -ma_result ma_device_write__oss(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) +static ma_result ma_device_write__audio4(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) { - int resultOSS; + int result; if (pFramesWritten != NULL) { *pFramesWritten = 0; } - resultOSS = write(pDevice->oss.fdPlayback, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); - if (resultOSS < 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to send data from the client to the device.", MA_FAILED_TO_SEND_DATA_TO_DEVICE); + result = write(pDevice->audio4.fdPlayback, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); + if (result < 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to write data to the device.", MA_FAILED_TO_SEND_DATA_TO_DEVICE); } if (pFramesWritten != NULL) { - *pFramesWritten = (ma_uint32)resultOSS / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + *pFramesWritten = (ma_uint32)result / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); } - + return MA_SUCCESS; } -ma_result ma_device_read__oss(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead) +static ma_result ma_device_read__audio4(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead) { - int resultOSS; + int result; if (pFramesRead != NULL) { *pFramesRead = 0; } - resultOSS = read(pDevice->oss.fdCapture, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); - if (resultOSS < 0) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to read data from the device to be sent to the client.", MA_FAILED_TO_READ_DATA_FROM_DEVICE); + result = read(pDevice->audio4.fdCapture, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); + if (result < 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to read data from the device.", MA_FAILED_TO_READ_DATA_FROM_DEVICE); } - + if (pFramesRead != NULL) { - *pFramesRead = (ma_uint32)resultOSS / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + *pFramesRead = (ma_uint32)result / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); } return MA_SUCCESS; } -ma_result ma_device_main_loop__oss(ma_device* pDevice) +static ma_result ma_device_main_loop__audio4(ma_device* pDevice) { ma_result result = MA_SUCCESS; ma_bool32 exitLoop = MA_FALSE; @@ -23067,77 +24474,88 @@ ma_result ma_device_main_loop__oss(ma_device* pDevice) case ma_device_type_duplex: { /* The process is: device_read -> convert -> callback -> convert -> device_write */ - ma_uint8 capturedDeviceData[8192]; - ma_uint8 playbackDeviceData[8192]; - ma_uint32 capturedDeviceDataCapInFrames = sizeof(capturedDeviceData) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 playbackDeviceDataCapInFrames = sizeof(playbackDeviceData) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - - ma_uint32 totalFramesProcessed = 0; - ma_uint32 periodSizeInFrames = ma_min(pDevice->capture.internalBufferSizeInFrames/pDevice->capture.internalPeriods, pDevice->playback.internalBufferSizeInFrames/pDevice->playback.internalPeriods); + ma_uint32 totalCapturedDeviceFramesProcessed = 0; + ma_uint32 capturedDevicePeriodSizeInFrames = ma_min(pDevice->capture.internalPeriodSizeInFrames, pDevice->playback.internalPeriodSizeInFrames); - while (totalFramesProcessed < periodSizeInFrames) { - ma_uint32 framesRemaining = periodSizeInFrames - totalFramesProcessed; - ma_uint32 framesProcessed; - ma_uint32 framesToProcess = framesRemaining; - if (framesToProcess > capturedDeviceDataCapInFrames) { - framesToProcess = capturedDeviceDataCapInFrames; + while (totalCapturedDeviceFramesProcessed < capturedDevicePeriodSizeInFrames) { + ma_uint8 capturedDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint8 playbackDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 capturedDeviceDataCapInFrames = sizeof(capturedDeviceData) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + ma_uint32 playbackDeviceDataCapInFrames = sizeof(playbackDeviceData) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + ma_uint32 capturedDeviceFramesRemaining; + ma_uint32 capturedDeviceFramesProcessed; + ma_uint32 capturedDeviceFramesToProcess; + ma_uint32 capturedDeviceFramesToTryProcessing = capturedDevicePeriodSizeInFrames - totalCapturedDeviceFramesProcessed; + if (capturedDeviceFramesToTryProcessing > capturedDeviceDataCapInFrames) { + capturedDeviceFramesToTryProcessing = capturedDeviceDataCapInFrames; } - result = ma_device_read__oss(pDevice, capturedDeviceData, framesToProcess, &framesProcessed); + result = ma_device_read__audio4(pDevice, capturedDeviceData, capturedDeviceFramesToTryProcessing, &capturedDeviceFramesToProcess); if (result != MA_SUCCESS) { exitLoop = MA_TRUE; break; } - pDevice->capture._dspFrameCount = framesToProcess; - pDevice->capture._dspFrames = capturedDeviceData; + capturedDeviceFramesRemaining = capturedDeviceFramesToProcess; + capturedDeviceFramesProcessed = 0; for (;;) { - ma_uint8 capturedData[8192]; - ma_uint8 playbackData[8192]; - ma_uint32 capturedDataCapInFrames = sizeof(capturedData) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); - ma_uint32 playbackDataCapInFrames = sizeof(playbackData) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); - - ma_uint32 capturedFramesToTryProcessing = ma_min(capturedDataCapInFrames, playbackDataCapInFrames); - ma_uint32 capturedFramesToProcess = (ma_uint32)ma_pcm_converter_read(&pDevice->capture.converter, capturedData, capturedFramesToTryProcessing); - if (capturedFramesToProcess == 0) { - break; /* Don't fire the data callback with zero frames. */ + ma_uint8 capturedClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint8 playbackClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 capturedClientDataCapInFrames = sizeof(capturedClientData) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); + ma_uint32 playbackClientDataCapInFrames = sizeof(playbackClientData) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); + ma_uint64 capturedClientFramesToProcessThisIteration = ma_min(capturedClientDataCapInFrames, playbackClientDataCapInFrames); + ma_uint64 capturedDeviceFramesToProcessThisIteration = capturedDeviceFramesRemaining; + ma_uint8* pRunningCapturedDeviceFrames = ma_offset_ptr(capturedDeviceData, capturedDeviceFramesProcessed * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); + + /* Convert capture data from device format to client format. */ + result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningCapturedDeviceFrames, &capturedDeviceFramesToProcessThisIteration, capturedClientData, &capturedClientFramesToProcessThisIteration); + if (result != MA_SUCCESS) { + break; + } + + /* + If we weren't able to generate any output frames it must mean we've exhaused all of our input. The only time this would not be the case is if capturedClientData was too small + which should never be the case when it's of the size MA_DATA_CONVERTER_STACK_BUFFER_SIZE. + */ + if (capturedClientFramesToProcessThisIteration == 0) { + break; } - ma_device__on_data(pDevice, playbackData, capturedData, capturedFramesToProcess); + ma_device__on_data(pDevice, playbackClientData, capturedClientData, (ma_uint32)capturedClientFramesToProcessThisIteration); /* Safe cast .*/ + + capturedDeviceFramesProcessed += (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */ + capturedDeviceFramesRemaining -= (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */ - /* At this point the playbackData buffer should be holding data that needs to be written to the device. */ - pDevice->playback._dspFrameCount = capturedFramesToProcess; - pDevice->playback._dspFrames = playbackData; + /* At this point the playbackClientData buffer should be holding data that needs to be written to the device. */ for (;;) { - ma_uint32 playbackDeviceFramesCount = (ma_uint32)ma_pcm_converter_read(&pDevice->playback.converter, playbackDeviceData, playbackDeviceDataCapInFrames); - if (playbackDeviceFramesCount == 0) { + ma_uint64 convertedClientFrameCount = capturedClientFramesToProcessThisIteration; + ma_uint64 convertedDeviceFrameCount = playbackDeviceDataCapInFrames; + result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, playbackClientData, &convertedClientFrameCount, playbackDeviceData, &convertedDeviceFrameCount); + if (result != MA_SUCCESS) { break; } - result = ma_device_write__oss(pDevice, playbackDeviceData, playbackDeviceFramesCount, NULL); + result = ma_device_write__audio4(pDevice, playbackDeviceData, (ma_uint32)convertedDeviceFrameCount, NULL); /* Safe cast. */ if (result != MA_SUCCESS) { exitLoop = MA_TRUE; break; } - if (playbackDeviceFramesCount < playbackDeviceDataCapInFrames) { + capturedClientFramesToProcessThisIteration -= (ma_uint32)convertedClientFrameCount; /* Safe cast. */ + if (capturedClientFramesToProcessThisIteration == 0) { break; } } - if (capturedFramesToProcess < capturedFramesToTryProcessing) { - break; - } - - /* In case an error happened from ma_device_write2__alsa()... */ + /* In case an error happened from ma_device_write__audio4()... */ if (result != MA_SUCCESS) { exitLoop = MA_TRUE; break; } } - totalFramesProcessed += framesProcessed; + totalCapturedDeviceFramesProcessed += capturedDeviceFramesProcessed; } } break; @@ -23146,7 +24564,7 @@ ma_result ma_device_main_loop__oss(ma_device* pDevice) /* We read in chunks of the period size, but use a stack allocated buffer for the intermediary. */ ma_uint8 intermediaryBuffer[8192]; ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - ma_uint32 periodSizeInFrames = pDevice->capture.internalBufferSizeInFrames / pDevice->capture.internalPeriods; + ma_uint32 periodSizeInFrames = pDevice->capture.internalPeriodSizeInFrames; ma_uint32 framesReadThisPeriod = 0; while (framesReadThisPeriod < periodSizeInFrames) { ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesReadThisPeriod; @@ -23156,7 +24574,7 @@ ma_result ma_device_main_loop__oss(ma_device* pDevice) framesToReadThisIteration = intermediaryBufferSizeInFrames; } - result = ma_device_read__oss(pDevice, intermediaryBuffer, framesToReadThisIteration, &framesProcessed); + result = ma_device_read__audio4(pDevice, intermediaryBuffer, framesToReadThisIteration, &framesProcessed); if (result != MA_SUCCESS) { exitLoop = MA_TRUE; break; @@ -23168,2201 +24586,2448 @@ ma_result ma_device_main_loop__oss(ma_device* pDevice) } } break; - case ma_device_type_playback: - { - /* We write in chunks of the period size, but use a stack allocated buffer for the intermediary. */ - ma_uint8 intermediaryBuffer[8192]; - ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - ma_uint32 periodSizeInFrames = pDevice->playback.internalBufferSizeInFrames / pDevice->playback.internalPeriods; - ma_uint32 framesWrittenThisPeriod = 0; - while (framesWrittenThisPeriod < periodSizeInFrames) { - ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesWrittenThisPeriod; - ma_uint32 framesProcessed; - ma_uint32 framesToWriteThisIteration = framesRemainingInPeriod; - if (framesToWriteThisIteration > intermediaryBufferSizeInFrames) { - framesToWriteThisIteration = intermediaryBufferSizeInFrames; - } + case ma_device_type_playback: + { + /* We write in chunks of the period size, but use a stack allocated buffer for the intermediary. */ + ma_uint8 intermediaryBuffer[8192]; + ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + ma_uint32 periodSizeInFrames = pDevice->playback.internalPeriodSizeInFrames; + ma_uint32 framesWrittenThisPeriod = 0; + while (framesWrittenThisPeriod < periodSizeInFrames) { + ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesWrittenThisPeriod; + ma_uint32 framesProcessed; + ma_uint32 framesToWriteThisIteration = framesRemainingInPeriod; + if (framesToWriteThisIteration > intermediaryBufferSizeInFrames) { + framesToWriteThisIteration = intermediaryBufferSizeInFrames; + } + + ma_device__read_frames_from_client(pDevice, framesToWriteThisIteration, intermediaryBuffer); + + result = ma_device_write__audio4(pDevice, intermediaryBuffer, framesToWriteThisIteration, &framesProcessed); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } + + framesWrittenThisPeriod += framesProcessed; + } + } break; + + /* To silence a warning. Will never hit this. */ + case ma_device_type_loopback: + default: break; + } + } + + + /* Here is where the device is stopped. */ + ma_device_stop__audio4(pDevice); + + return result; +} + +static ma_result ma_context_uninit__audio4(ma_context* pContext) +{ + MA_ASSERT(pContext != NULL); + MA_ASSERT(pContext->backend == ma_backend_audio4); + + (void)pContext; + return MA_SUCCESS; +} + +static ma_result ma_context_init__audio4(const ma_context_config* pConfig, ma_context* pContext) +{ + MA_ASSERT(pContext != NULL); + + (void)pConfig; + + pContext->onUninit = ma_context_uninit__audio4; + pContext->onDeviceIDEqual = ma_context_is_device_id_equal__audio4; + pContext->onEnumDevices = ma_context_enumerate_devices__audio4; + pContext->onGetDeviceInfo = ma_context_get_device_info__audio4; + pContext->onDeviceInit = ma_device_init__audio4; + pContext->onDeviceUninit = ma_device_uninit__audio4; + pContext->onDeviceStart = NULL; /* Not required for synchronous backends. */ + pContext->onDeviceStop = NULL; /* Not required for synchronous backends. */ + pContext->onDeviceMainLoop = ma_device_main_loop__audio4; + + return MA_SUCCESS; +} +#endif /* audio4 */ + + +/****************************************************************************** + +OSS Backend + +******************************************************************************/ +#ifdef MA_HAS_OSS +#include +#include +#include +#include + +#ifndef SNDCTL_DSP_HALT +#define SNDCTL_DSP_HALT SNDCTL_DSP_RESET +#endif + +static int ma_open_temp_device__oss() +{ + /* The OSS sample code uses "/dev/mixer" as the device for getting system properties so I'm going to do the same. */ + int fd = open("/dev/mixer", O_RDONLY, 0); + if (fd >= 0) { + return fd; + } + + return -1; +} + +static ma_result ma_context_open_device__oss(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, int* pfd) +{ + const char* deviceName; + int flags; - ma_device__read_frames_from_client(pDevice, framesToWriteThisIteration, intermediaryBuffer); + MA_ASSERT(pContext != NULL); + MA_ASSERT(pfd != NULL); + (void)pContext; - result = ma_device_write__oss(pDevice, intermediaryBuffer, framesToWriteThisIteration, &framesProcessed); - if (result != MA_SUCCESS) { - exitLoop = MA_TRUE; - break; - } + *pfd = -1; - framesWrittenThisPeriod += framesProcessed; - } - } break; + /* This function should only be called for playback or capture, not duplex. */ + if (deviceType == ma_device_type_duplex) { + return MA_INVALID_ARGS; + } - /* To silence a warning. Will never hit this. */ - case ma_device_type_loopback: - default: break; - } + deviceName = "/dev/dsp"; + if (pDeviceID != NULL) { + deviceName = pDeviceID->oss; } + flags = (deviceType == ma_device_type_playback) ? O_WRONLY : O_RDONLY; + if (shareMode == ma_share_mode_exclusive) { + flags |= O_EXCL; + } - /* Here is where the device is stopped. */ - ma_device_stop__oss(pDevice); + *pfd = open(deviceName, flags, 0); + if (*pfd == -1) { + return MA_FAILED_TO_OPEN_BACKEND_DEVICE; + } - return result; + return MA_SUCCESS; } -ma_result ma_context_uninit__oss(ma_context* pContext) +static ma_bool32 ma_context_is_device_id_equal__oss(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) { - ma_assert(pContext != NULL); - ma_assert(pContext->backend == ma_backend_oss); - + MA_ASSERT(pContext != NULL); + MA_ASSERT(pID0 != NULL); + MA_ASSERT(pID1 != NULL); (void)pContext; - return MA_SUCCESS; + + return ma_strcmp(pID0->oss, pID1->oss) == 0; } -ma_result ma_context_init__oss(const ma_context_config* pConfig, ma_context* pContext) +static ma_result ma_context_enumerate_devices__oss(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { int fd; - int ossVersion; + oss_sysinfo si; int result; - ma_assert(pContext != NULL); - - (void)pConfig; + MA_ASSERT(pContext != NULL); + MA_ASSERT(callback != NULL); - /* Try opening a temporary device first so we can get version information. This is closed at the end. */ fd = ma_open_temp_device__oss(); if (fd == -1) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[OSS] Failed to open temporary device for retrieving system properties.", MA_NO_BACKEND); /* Looks liks OSS isn't installed, or there are no available devices. */ - } - - /* Grab the OSS version. */ - ossVersion = 0; - result = ioctl(fd, OSS_GETVERSION, &ossVersion); - if (result == -1) { - close(fd); - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[OSS] Failed to retrieve OSS version.", MA_NO_BACKEND); + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[OSS] Failed to open a temporary device for retrieving system information used for device enumeration.", MA_NO_BACKEND); } - pContext->oss.versionMajor = ((ossVersion & 0xFF0000) >> 16); - pContext->oss.versionMinor = ((ossVersion & 0x00FF00) >> 8); - - pContext->onUninit = ma_context_uninit__oss; - pContext->onDeviceIDEqual = ma_context_is_device_id_equal__oss; - pContext->onEnumDevices = ma_context_enumerate_devices__oss; - pContext->onGetDeviceInfo = ma_context_get_device_info__oss; - pContext->onDeviceInit = ma_device_init__oss; - pContext->onDeviceUninit = ma_device_uninit__oss; - pContext->onDeviceStart = NULL; /* Not required for synchronous backends. */ - pContext->onDeviceStop = NULL; /* Not required for synchronous backends. */ - pContext->onDeviceMainLoop = ma_device_main_loop__oss; - - close(fd); - return MA_SUCCESS; -} -#endif /* OSS */ - + result = ioctl(fd, SNDCTL_SYSINFO, &si); + if (result != -1) { + int iAudioDevice; + for (iAudioDevice = 0; iAudioDevice < si.numaudios; ++iAudioDevice) { + oss_audioinfo ai; + ai.dev = iAudioDevice; + result = ioctl(fd, SNDCTL_AUDIOINFO, &ai); + if (result != -1) { + if (ai.devnode[0] != '\0') { /* <-- Can be blank, according to documentation. */ + ma_device_info deviceInfo; + ma_bool32 isTerminating = MA_FALSE; -/****************************************************************************** + MA_ZERO_OBJECT(&deviceInfo); -AAudio Backend + /* ID */ + ma_strncpy_s(deviceInfo.id.oss, sizeof(deviceInfo.id.oss), ai.devnode, (size_t)-1); -******************************************************************************/ -#ifdef MA_HAS_AAUDIO -/*#include */ + /* + The human readable device name should be in the "ai.handle" variable, but it can + sometimes be empty in which case we just fall back to "ai.name" which is less user + friendly, but usually has a value. + */ + if (ai.handle[0] != '\0') { + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), ai.handle, (size_t)-1); + } else { + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), ai.name, (size_t)-1); + } -#define MA_AAUDIO_UNSPECIFIED 0 + /* The device can be both playback and capture. */ + if (!isTerminating && (ai.caps & PCM_CAP_OUTPUT) != 0) { + isTerminating = !callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); + } + if (!isTerminating && (ai.caps & PCM_CAP_INPUT) != 0) { + isTerminating = !callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); + } -typedef int32_t ma_aaudio_result_t; -typedef int32_t ma_aaudio_direction_t; -typedef int32_t ma_aaudio_sharing_mode_t; -typedef int32_t ma_aaudio_format_t; -typedef int32_t ma_aaudio_stream_state_t; -typedef int32_t ma_aaudio_performance_mode_t; -typedef int32_t ma_aaudio_data_callback_result_t; + if (isTerminating) { + break; + } + } + } + } + } else { + close(fd); + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[OSS] Failed to retrieve system information for device enumeration.", MA_NO_BACKEND); + } -/* Result codes. miniaudio only cares about the success code. */ -#define MA_AAUDIO_OK 0 + close(fd); + return MA_SUCCESS; +} -/* Directions. */ -#define MA_AAUDIO_DIRECTION_OUTPUT 0 -#define MA_AAUDIO_DIRECTION_INPUT 1 +static ma_result ma_context_get_device_info__oss(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) +{ + ma_bool32 foundDevice; + int fdTemp; + oss_sysinfo si; + int result; -/* Sharing modes. */ -#define MA_AAUDIO_SHARING_MODE_EXCLUSIVE 0 -#define MA_AAUDIO_SHARING_MODE_SHARED 1 + MA_ASSERT(pContext != NULL); + (void)shareMode; -/* Formats. */ -#define MA_AAUDIO_FORMAT_PCM_I16 1 -#define MA_AAUDIO_FORMAT_PCM_FLOAT 2 + /* Handle the default device a little differently. */ + if (pDeviceID == NULL) { + if (deviceType == ma_device_type_playback) { + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); + } else { + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); + } -/* Stream states. */ -#define MA_AAUDIO_STREAM_STATE_UNINITIALIZED 0 -#define MA_AAUDIO_STREAM_STATE_UNKNOWN 1 -#define MA_AAUDIO_STREAM_STATE_OPEN 2 -#define MA_AAUDIO_STREAM_STATE_STARTING 3 -#define MA_AAUDIO_STREAM_STATE_STARTED 4 -#define MA_AAUDIO_STREAM_STATE_PAUSING 5 -#define MA_AAUDIO_STREAM_STATE_PAUSED 6 -#define MA_AAUDIO_STREAM_STATE_FLUSHING 7 -#define MA_AAUDIO_STREAM_STATE_FLUSHED 8 -#define MA_AAUDIO_STREAM_STATE_STOPPING 9 -#define MA_AAUDIO_STREAM_STATE_STOPPED 10 -#define MA_AAUDIO_STREAM_STATE_CLOSING 11 -#define MA_AAUDIO_STREAM_STATE_CLOSED 12 -#define MA_AAUDIO_STREAM_STATE_DISCONNECTED 13 + return MA_SUCCESS; + } -/* Performance modes. */ -#define MA_AAUDIO_PERFORMANCE_MODE_NONE 10 -#define MA_AAUDIO_PERFORMANCE_MODE_POWER_SAVING 11 -#define MA_AAUDIO_PERFORMANCE_MODE_LOW_LATENCY 12 -/* Callback results. */ -#define MA_AAUDIO_CALLBACK_RESULT_CONTINUE 0 -#define MA_AAUDIO_CALLBACK_RESULT_STOP 1 + /* If we get here it means we are _not_ using the default device. */ + foundDevice = MA_FALSE; -/* Objects. */ -typedef struct ma_AAudioStreamBuilder_t* ma_AAudioStreamBuilder; -typedef struct ma_AAudioStream_t* ma_AAudioStream; + fdTemp = ma_open_temp_device__oss(); + if (fdTemp == -1) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[OSS] Failed to open a temporary device for retrieving system information used for device enumeration.", MA_NO_BACKEND); + } -typedef ma_aaudio_data_callback_result_t (*ma_AAudioStream_dataCallback)(ma_AAudioStream* pStream, void* pUserData, void* pAudioData, int32_t numFrames); + result = ioctl(fdTemp, SNDCTL_SYSINFO, &si); + if (result != -1) { + int iAudioDevice; + for (iAudioDevice = 0; iAudioDevice < si.numaudios; ++iAudioDevice) { + oss_audioinfo ai; + ai.dev = iAudioDevice; + result = ioctl(fdTemp, SNDCTL_AUDIOINFO, &ai); + if (result != -1) { + if (ma_strcmp(ai.devnode, pDeviceID->oss) == 0) { + /* It has the same name, so now just confirm the type. */ + if ((deviceType == ma_device_type_playback && ((ai.caps & PCM_CAP_OUTPUT) != 0)) || + (deviceType == ma_device_type_capture && ((ai.caps & PCM_CAP_INPUT) != 0))) { + unsigned int formatMask; -typedef ma_aaudio_result_t (* MA_PFN_AAudio_createStreamBuilder) (ma_AAudioStreamBuilder** ppBuilder); -typedef ma_aaudio_result_t (* MA_PFN_AAudioStreamBuilder_delete) (ma_AAudioStreamBuilder* pBuilder); -typedef void (* MA_PFN_AAudioStreamBuilder_setDeviceId) (ma_AAudioStreamBuilder* pBuilder, int32_t deviceId); -typedef void (* MA_PFN_AAudioStreamBuilder_setDirection) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_direction_t direction); -typedef void (* MA_PFN_AAudioStreamBuilder_setSharingMode) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_sharing_mode_t sharingMode); -typedef void (* MA_PFN_AAudioStreamBuilder_setFormat) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_format_t format); -typedef void (* MA_PFN_AAudioStreamBuilder_setChannelCount) (ma_AAudioStreamBuilder* pBuilder, int32_t channelCount); -typedef void (* MA_PFN_AAudioStreamBuilder_setSampleRate) (ma_AAudioStreamBuilder* pBuilder, int32_t sampleRate); -typedef void (* MA_PFN_AAudioStreamBuilder_setBufferCapacityInFrames)(ma_AAudioStreamBuilder* pBuilder, int32_t numFrames); -typedef void (* MA_PFN_AAudioStreamBuilder_setFramesPerDataCallback) (ma_AAudioStreamBuilder* pBuilder, int32_t numFrames); -typedef void (* MA_PFN_AAudioStreamBuilder_setDataCallback) (ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream_dataCallback callback, void* pUserData); -typedef void (* MA_PFN_AAudioStreamBuilder_setPerformanceMode) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_performance_mode_t mode); -typedef ma_aaudio_result_t (* MA_PFN_AAudioStreamBuilder_openStream) (ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream** ppStream); -typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_close) (ma_AAudioStream* pStream); -typedef ma_aaudio_stream_state_t (* MA_PFN_AAudioStream_getState) (ma_AAudioStream* pStream); -typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_waitForStateChange) (ma_AAudioStream* pStream, ma_aaudio_stream_state_t inputState, ma_aaudio_stream_state_t* pNextState, int64_t timeoutInNanoseconds); -typedef ma_aaudio_format_t (* MA_PFN_AAudioStream_getFormat) (ma_AAudioStream* pStream); -typedef int32_t (* MA_PFN_AAudioStream_getChannelCount) (ma_AAudioStream* pStream); -typedef int32_t (* MA_PFN_AAudioStream_getSampleRate) (ma_AAudioStream* pStream); -typedef int32_t (* MA_PFN_AAudioStream_getBufferCapacityInFrames) (ma_AAudioStream* pStream); -typedef int32_t (* MA_PFN_AAudioStream_getFramesPerDataCallback) (ma_AAudioStream* pStream); -typedef int32_t (* MA_PFN_AAudioStream_getFramesPerBurst) (ma_AAudioStream* pStream); -typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_requestStart) (ma_AAudioStream* pStream); -typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_requestStop) (ma_AAudioStream* pStream); + /* ID */ + ma_strncpy_s(pDeviceInfo->id.oss, sizeof(pDeviceInfo->id.oss), ai.devnode, (size_t)-1); -ma_result ma_result_from_aaudio(ma_aaudio_result_t resultAA) -{ - switch (resultAA) - { - case MA_AAUDIO_OK: return MA_SUCCESS; - default: break; - } + /* + The human readable device name should be in the "ai.handle" variable, but it can + sometimes be empty in which case we just fall back to "ai.name" which is less user + friendly, but usually has a value. + */ + if (ai.handle[0] != '\0') { + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), ai.handle, (size_t)-1); + } else { + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), ai.name, (size_t)-1); + } - return MA_ERROR; -} + pDeviceInfo->minChannels = ai.min_channels; + pDeviceInfo->maxChannels = ai.max_channels; + pDeviceInfo->minSampleRate = ai.min_rate; + pDeviceInfo->maxSampleRate = ai.max_rate; + pDeviceInfo->formatCount = 0; -ma_aaudio_data_callback_result_t ma_stream_data_callback_capture__aaudio(ma_AAudioStream* pStream, void* pUserData, void* pAudioData, int32_t frameCount) -{ - ma_device* pDevice = (ma_device*)pUserData; - ma_assert(pDevice != NULL); + if (deviceType == ma_device_type_playback) { + formatMask = ai.oformats; + } else { + formatMask = ai.iformats; + } - if (pDevice->type == ma_device_type_duplex) { - ma_device__handle_duplex_callback_capture(pDevice, frameCount, pAudioData, &pDevice->aaudio.duplexRB); + if ((formatMask & AFMT_U8) != 0) { + pDeviceInfo->formats[pDeviceInfo->formatCount++] = ma_format_u8; + } + if (((formatMask & AFMT_S16_LE) != 0 && ma_is_little_endian()) || (AFMT_S16_BE && ma_is_big_endian())) { + pDeviceInfo->formats[pDeviceInfo->formatCount++] = ma_format_s16; + } + if (((formatMask & AFMT_S32_LE) != 0 && ma_is_little_endian()) || (AFMT_S32_BE && ma_is_big_endian())) { + pDeviceInfo->formats[pDeviceInfo->formatCount++] = ma_format_s32; + } + + foundDevice = MA_TRUE; + break; + } + } + } + } } else { - ma_device__send_frames_to_client(pDevice, frameCount, pAudioData); /* Send directly to the client. */ + close(fdTemp); + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[OSS] Failed to retrieve system information for device enumeration.", MA_NO_BACKEND); } - (void)pStream; - return MA_AAUDIO_CALLBACK_RESULT_CONTINUE; -} -ma_aaudio_data_callback_result_t ma_stream_data_callback_playback__aaudio(ma_AAudioStream* pStream, void* pUserData, void* pAudioData, int32_t frameCount) -{ - ma_device* pDevice = (ma_device*)pUserData; - ma_assert(pDevice != NULL); + close(fdTemp); - if (pDevice->type == ma_device_type_duplex) { - ma_device__handle_duplex_callback_playback(pDevice, frameCount, pAudioData, &pDevice->aaudio.duplexRB); - } else { - ma_device__read_frames_from_client(pDevice, frameCount, pAudioData); /* Read directly from the client. */ + if (!foundDevice) { + return MA_NO_DEVICE; } - (void)pStream; - return MA_AAUDIO_CALLBACK_RESULT_CONTINUE; + return MA_SUCCESS; } -ma_result ma_open_stream__aaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, const ma_device_config* pConfig, const ma_device* pDevice, ma_AAudioStream** ppStream) +static void ma_device_uninit__oss(ma_device* pDevice) { - ma_AAudioStreamBuilder* pBuilder; - ma_aaudio_result_t resultAA; + MA_ASSERT(pDevice != NULL); - ma_assert(deviceType != ma_device_type_duplex); /* This function should not be called for a full-duplex device type. */ - - *ppStream = NULL; - - resultAA = ((MA_PFN_AAudio_createStreamBuilder)pContext->aaudio.AAudio_createStreamBuilder)(&pBuilder); - if (resultAA != MA_AAUDIO_OK) { - return ma_result_from_aaudio(resultAA); + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + close(pDevice->oss.fdCapture); } - - if (pDeviceID != NULL) { - ((MA_PFN_AAudioStreamBuilder_setDeviceId)pContext->aaudio.AAudioStreamBuilder_setDeviceId)(pBuilder, pDeviceID->aaudio); + + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + close(pDevice->oss.fdPlayback); } +} - ((MA_PFN_AAudioStreamBuilder_setDirection)pContext->aaudio.AAudioStreamBuilder_setDirection)(pBuilder, (deviceType == ma_device_type_playback) ? MA_AAUDIO_DIRECTION_OUTPUT : MA_AAUDIO_DIRECTION_INPUT); - ((MA_PFN_AAudioStreamBuilder_setSharingMode)pContext->aaudio.AAudioStreamBuilder_setSharingMode)(pBuilder, (shareMode == ma_share_mode_shared) ? MA_AAUDIO_SHARING_MODE_SHARED : MA_AAUDIO_SHARING_MODE_EXCLUSIVE); - - if (pConfig != NULL) { - ma_uint32 bufferCapacityInFrames; +static int ma_format_to_oss(ma_format format) +{ + int ossFormat = AFMT_U8; + switch (format) { + case ma_format_s16: ossFormat = (ma_is_little_endian()) ? AFMT_S16_LE : AFMT_S16_BE; break; + case ma_format_s24: ossFormat = (ma_is_little_endian()) ? AFMT_S32_LE : AFMT_S32_BE; break; + case ma_format_s32: ossFormat = (ma_is_little_endian()) ? AFMT_S32_LE : AFMT_S32_BE; break; + case ma_format_f32: ossFormat = (ma_is_little_endian()) ? AFMT_S16_LE : AFMT_S16_BE; break; + case ma_format_u8: + default: ossFormat = AFMT_U8; break; + } - if (pDevice == NULL || !pDevice->usingDefaultSampleRate) { - ((MA_PFN_AAudioStreamBuilder_setSampleRate)pContext->aaudio.AAudioStreamBuilder_setSampleRate)(pBuilder, pConfig->sampleRate); - } + return ossFormat; +} - if (deviceType == ma_device_type_capture) { - if (pDevice == NULL || !pDevice->capture.usingDefaultChannels) { - ((MA_PFN_AAudioStreamBuilder_setChannelCount)pContext->aaudio.AAudioStreamBuilder_setChannelCount)(pBuilder, pConfig->capture.channels); - } - if (pDevice == NULL || !pDevice->capture.usingDefaultFormat) { - ((MA_PFN_AAudioStreamBuilder_setFormat)pContext->aaudio.AAudioStreamBuilder_setFormat)(pBuilder, (pConfig->capture.format == ma_format_s16) ? MA_AAUDIO_FORMAT_PCM_I16 : MA_AAUDIO_FORMAT_PCM_FLOAT); +static ma_format ma_format_from_oss(int ossFormat) +{ + if (ossFormat == AFMT_U8) { + return ma_format_u8; + } else { + if (ma_is_little_endian()) { + switch (ossFormat) { + case AFMT_S16_LE: return ma_format_s16; + case AFMT_S32_LE: return ma_format_s32; + default: return ma_format_unknown; } } else { - if (pDevice == NULL || !pDevice->playback.usingDefaultChannels) { - ((MA_PFN_AAudioStreamBuilder_setChannelCount)pContext->aaudio.AAudioStreamBuilder_setChannelCount)(pBuilder, pConfig->playback.channels); - } - if (pDevice == NULL || !pDevice->playback.usingDefaultFormat) { - ((MA_PFN_AAudioStreamBuilder_setFormat)pContext->aaudio.AAudioStreamBuilder_setFormat)(pBuilder, (pConfig->playback.format == ma_format_s16) ? MA_AAUDIO_FORMAT_PCM_I16 : MA_AAUDIO_FORMAT_PCM_FLOAT); + switch (ossFormat) { + case AFMT_S16_BE: return ma_format_s16; + case AFMT_S32_BE: return ma_format_s32; + default: return ma_format_unknown; } } - - bufferCapacityInFrames = pConfig->bufferSizeInFrames; - if (bufferCapacityInFrames == 0) { - bufferCapacityInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->bufferSizeInMilliseconds, pConfig->sampleRate); - } - bufferCapacityInFrames = (bufferCapacityInFrames / pConfig->periods) * pConfig->periods; /* <-- Make sure the buffer capacity is an even multiple of a period. */ - ((MA_PFN_AAudioStreamBuilder_setBufferCapacityInFrames)pContext->aaudio.AAudioStreamBuilder_setBufferCapacityInFrames)(pBuilder, bufferCapacityInFrames); - - ((MA_PFN_AAudioStreamBuilder_setFramesPerDataCallback)pContext->aaudio.AAudioStreamBuilder_setFramesPerDataCallback)(pBuilder, bufferCapacityInFrames / pConfig->periods); - - if (deviceType == ma_device_type_capture) { - ((MA_PFN_AAudioStreamBuilder_setDataCallback)pContext->aaudio.AAudioStreamBuilder_setDataCallback)(pBuilder, ma_stream_data_callback_capture__aaudio, (void*)pDevice); - } else { - ((MA_PFN_AAudioStreamBuilder_setDataCallback)pContext->aaudio.AAudioStreamBuilder_setDataCallback)(pBuilder, ma_stream_data_callback_playback__aaudio, (void*)pDevice); - } - - /* Not sure how this affects things, but since there's a mapping between miniaudio's performance profiles and AAudio's performance modes, let go ahead and set it. */ - ((MA_PFN_AAudioStreamBuilder_setPerformanceMode)pContext->aaudio.AAudioStreamBuilder_setPerformanceMode)(pBuilder, (pConfig->performanceProfile == ma_performance_profile_low_latency) ? MA_AAUDIO_PERFORMANCE_MODE_LOW_LATENCY : MA_AAUDIO_PERFORMANCE_MODE_NONE); - } - - resultAA = ((MA_PFN_AAudioStreamBuilder_openStream)pContext->aaudio.AAudioStreamBuilder_openStream)(pBuilder, ppStream); - if (resultAA != MA_AAUDIO_OK) { - *ppStream = NULL; - ((MA_PFN_AAudioStreamBuilder_delete)pContext->aaudio.AAudioStreamBuilder_delete)(pBuilder); - return ma_result_from_aaudio(resultAA); } - ((MA_PFN_AAudioStreamBuilder_delete)pContext->aaudio.AAudioStreamBuilder_delete)(pBuilder); - return MA_SUCCESS; + return ma_format_unknown; } -ma_result ma_close_stream__aaudio(ma_context* pContext, ma_AAudioStream* pStream) +static ma_result ma_device_init_fd__oss(ma_context* pContext, const ma_device_config* pConfig, ma_device_type deviceType, ma_device* pDevice) { - return ma_result_from_aaudio(((MA_PFN_AAudioStream_close)pContext->aaudio.AAudioStream_close)(pStream)); -} + ma_result result; + int ossResult; + int fd; + const ma_device_id* pDeviceID = NULL; + ma_share_mode shareMode; + int ossFormat; + int ossChannels; + int ossSampleRate; + int ossFragment; -ma_bool32 ma_has_default_device__aaudio(ma_context* pContext, ma_device_type deviceType) -{ - /* The only way to know this is to try creating a stream. */ - ma_AAudioStream* pStream; - ma_result result = ma_open_stream__aaudio(pContext, deviceType, NULL, ma_share_mode_shared, NULL, NULL, &pStream); - if (result != MA_SUCCESS) { - return MA_FALSE; - } + MA_ASSERT(pContext != NULL); + MA_ASSERT(pConfig != NULL); + MA_ASSERT(deviceType != ma_device_type_duplex); + MA_ASSERT(pDevice != NULL); - ma_close_stream__aaudio(pContext, pStream); - return MA_TRUE; -} + (void)pContext; -ma_result ma_wait_for_simple_state_transition__aaudio(ma_context* pContext, ma_AAudioStream* pStream, ma_aaudio_stream_state_t oldState, ma_aaudio_stream_state_t newState) -{ - ma_aaudio_stream_state_t actualNewState; - ma_aaudio_result_t resultAA = ((MA_PFN_AAudioStream_waitForStateChange)pContext->aaudio.AAudioStream_waitForStateChange)(pStream, oldState, &actualNewState, 5000000000); /* 5 second timeout. */ - if (resultAA != MA_AAUDIO_OK) { - return ma_result_from_aaudio(resultAA); + if (deviceType == ma_device_type_capture) { + pDeviceID = pConfig->capture.pDeviceID; + shareMode = pConfig->capture.shareMode; + ossFormat = ma_format_to_oss(pConfig->capture.format); + ossChannels = (int)pConfig->capture.channels; + ossSampleRate = (int)pConfig->sampleRate; + } else { + pDeviceID = pConfig->playback.pDeviceID; + shareMode = pConfig->playback.shareMode; + ossFormat = ma_format_to_oss(pConfig->playback.format); + ossChannels = (int)pConfig->playback.channels; + ossSampleRate = (int)pConfig->sampleRate; } - if (newState != actualNewState) { - return MA_ERROR; /* Failed to transition into the expected state. */ + result = ma_context_open_device__oss(pContext, deviceType, pDeviceID, shareMode, &fd); + if (result != MA_SUCCESS) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to open device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); } - return MA_SUCCESS; -} + /* + The OSS documantation is very clear about the order we should be initializing the device's properties: + 1) Format + 2) Channels + 3) Sample rate. + */ + /* Format. */ + ossResult = ioctl(fd, SNDCTL_DSP_SETFMT, &ossFormat); + if (ossResult == -1) { + close(fd); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to set format.", MA_FORMAT_NOT_SUPPORTED); + } -ma_bool32 ma_context_is_device_id_equal__aaudio(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) -{ - ma_assert(pContext != NULL); - ma_assert(pID0 != NULL); - ma_assert(pID1 != NULL); - (void)pContext; + /* Channels. */ + ossResult = ioctl(fd, SNDCTL_DSP_CHANNELS, &ossChannels); + if (ossResult == -1) { + close(fd); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to set channel count.", MA_FORMAT_NOT_SUPPORTED); + } - return pID0->aaudio == pID1->aaudio; -} + /* Sample Rate. */ + ossResult = ioctl(fd, SNDCTL_DSP_SPEED, &ossSampleRate); + if (ossResult == -1) { + close(fd); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to set sample rate.", MA_FORMAT_NOT_SUPPORTED); + } -ma_result ma_context_enumerate_devices__aaudio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) -{ - ma_bool32 cbResult = MA_TRUE; + /* + Buffer. - ma_assert(pContext != NULL); - ma_assert(callback != NULL); + The documentation says that the fragment settings should be set as soon as possible, but I'm not sure if + it should be done before or after format/channels/rate. + + OSS wants the fragment size in bytes and a power of 2. When setting, we specify the power, not the actual + value. + */ + { + ma_uint32 periodSizeInFrames; + ma_uint32 periodSizeInBytes; + ma_uint32 ossFragmentSizePower; + + periodSizeInFrames = pConfig->periodSizeInFrames; + if (periodSizeInFrames == 0) { + periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->periodSizeInMilliseconds, (ma_uint32)ossSampleRate); + } - /* Unfortunately AAudio does not have an enumeration API. Therefore I'm only going to report default devices, but only if it can instantiate a stream. */ + periodSizeInBytes = ma_round_to_power_of_2(periodSizeInFrames * ma_get_bytes_per_frame(ma_format_from_oss(ossFormat), ossChannels)); + if (periodSizeInBytes < 16) { + periodSizeInBytes = 16; + } - /* Playback. */ - if (cbResult) { - ma_device_info deviceInfo; - ma_zero_object(&deviceInfo); - deviceInfo.id.aaudio = MA_AAUDIO_UNSPECIFIED; - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); + ossFragmentSizePower = 4; + periodSizeInBytes >>= 4; + while (periodSizeInBytes >>= 1) { + ossFragmentSizePower += 1; + } - if (ma_has_default_device__aaudio(pContext, ma_device_type_playback)) { - cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); + ossFragment = (int)((pConfig->periods << 16) | ossFragmentSizePower); + ossResult = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &ossFragment); + if (ossResult == -1) { + close(fd); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to set fragment size and period count.", MA_FORMAT_NOT_SUPPORTED); } } - /* Capture. */ - if (cbResult) { - ma_device_info deviceInfo; - ma_zero_object(&deviceInfo); - deviceInfo.id.aaudio = MA_AAUDIO_UNSPECIFIED; - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); + /* Internal settings. */ + if (deviceType == ma_device_type_capture) { + pDevice->oss.fdCapture = fd; + pDevice->capture.internalFormat = ma_format_from_oss(ossFormat); + pDevice->capture.internalChannels = ossChannels; + pDevice->capture.internalSampleRate = ossSampleRate; + ma_get_standard_channel_map(ma_standard_channel_map_sound4, pDevice->capture.internalChannels, pDevice->capture.internalChannelMap); + pDevice->capture.internalPeriods = (ma_uint32)(ossFragment >> 16); + pDevice->capture.internalPeriodSizeInFrames = (ma_uint32)(1 << (ossFragment & 0xFFFF)) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + + if (pDevice->capture.internalFormat == ma_format_unknown) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] The device's internal format is not supported by miniaudio.", MA_FORMAT_NOT_SUPPORTED); + } + } else { + pDevice->oss.fdPlayback = fd; + pDevice->playback.internalFormat = ma_format_from_oss(ossFormat); + pDevice->playback.internalChannels = ossChannels; + pDevice->playback.internalSampleRate = ossSampleRate; + ma_get_standard_channel_map(ma_standard_channel_map_sound4, pDevice->playback.internalChannels, pDevice->playback.internalChannelMap); + pDevice->playback.internalPeriods = (ma_uint32)(ossFragment >> 16); + pDevice->playback.internalPeriodSizeInFrames = (ma_uint32)(1 << (ossFragment & 0xFFFF)) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - if (ma_has_default_device__aaudio(pContext, ma_device_type_capture)) { - cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); + if (pDevice->playback.internalFormat == ma_format_unknown) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] The device's internal format is not supported by miniaudio.", MA_FORMAT_NOT_SUPPORTED); } } return MA_SUCCESS; } -ma_result ma_context_get_device_info__aaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) +static ma_result ma_device_init__oss(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) { - ma_AAudioStream* pStream; - ma_result result; + MA_ASSERT(pContext != NULL); + MA_ASSERT(pConfig != NULL); + MA_ASSERT(pDevice != NULL); - ma_assert(pContext != NULL); + MA_ZERO_OBJECT(&pDevice->oss); - /* No exclusive mode with AAudio. */ - if (shareMode == ma_share_mode_exclusive) { - return MA_SHARE_MODE_NOT_SUPPORTED; + if (pConfig->deviceType == ma_device_type_loopback) { + return MA_DEVICE_TYPE_NOT_SUPPORTED; } - /* ID */ - if (pDeviceID != NULL) { - pDeviceInfo->id.aaudio = pDeviceID->aaudio; - } else { - pDeviceInfo->id.aaudio = MA_AAUDIO_UNSPECIFIED; - } - - /* Name */ - if (deviceType == ma_device_type_playback) { - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); - } else { - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); + if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { + ma_result result = ma_device_init_fd__oss(pContext, pConfig, ma_device_type_capture, pDevice); + if (result != MA_SUCCESS) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to open device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } } - - /* We'll need to open the device to get accurate sample rate and channel count information. */ - result = ma_open_stream__aaudio(pContext, deviceType, pDeviceID, shareMode, NULL, NULL, &pStream); - if (result != MA_SUCCESS) { - return result; + if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { + ma_result result = ma_device_init_fd__oss(pContext, pConfig, ma_device_type_playback, pDevice); + if (result != MA_SUCCESS) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to open device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } } - pDeviceInfo->minChannels = ((MA_PFN_AAudioStream_getChannelCount)pContext->aaudio.AAudioStream_getChannelCount)(pStream); - pDeviceInfo->maxChannels = pDeviceInfo->minChannels; - pDeviceInfo->minSampleRate = ((MA_PFN_AAudioStream_getSampleRate)pContext->aaudio.AAudioStream_getSampleRate)(pStream); - pDeviceInfo->maxSampleRate = pDeviceInfo->minSampleRate; - - ma_close_stream__aaudio(pContext, pStream); - pStream = NULL; - - - /* AAudio supports s16 and f32. */ - pDeviceInfo->formatCount = 2; - pDeviceInfo->formats[0] = ma_format_s16; - pDeviceInfo->formats[1] = ma_format_f32; - return MA_SUCCESS; } - -void ma_device_uninit__aaudio(ma_device* pDevice) +static ma_result ma_device_stop__oss(ma_device* pDevice) { - ma_assert(pDevice != NULL); + MA_ASSERT(pDevice != NULL); + + /* + We want to use SNDCTL_DSP_HALT. From the documentation: + + In multithreaded applications SNDCTL_DSP_HALT (SNDCTL_DSP_RESET) must only be called by the thread + that actually reads/writes the audio device. It must not be called by some master thread to kill the + audio thread. The audio thread will not stop or get any kind of notification that the device was + stopped by the master thread. The device gets stopped but the next read or write call will silently + restart the device. + + This is actually safe in our case, because this function is only ever called from within our worker + thread anyway. Just keep this in mind, though... + */ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture); - pDevice->aaudio.pStreamCapture = NULL; + int result = ioctl(pDevice->oss.fdCapture, SNDCTL_DSP_HALT, 0); + if (result == -1) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to stop device. SNDCTL_DSP_HALT failed.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + } } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); - pDevice->aaudio.pStreamPlayback = NULL; + int result = ioctl(pDevice->oss.fdPlayback, SNDCTL_DSP_HALT, 0); + if (result == -1) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to stop device. SNDCTL_DSP_HALT failed.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + } } - if (pDevice->type == ma_device_type_duplex) { - ma_pcm_rb_uninit(&pDevice->aaudio.duplexRB); - } + return MA_SUCCESS; } -ma_result ma_device_init__aaudio(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) +static ma_result ma_device_write__oss(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten) { - ma_result result; - - ma_assert(pDevice != NULL); + int resultOSS; - if (pConfig->deviceType == ma_device_type_loopback) { - return MA_DEVICE_TYPE_NOT_SUPPORTED; + if (pFramesWritten != NULL) { + *pFramesWritten = 0; } - /* No exclusive mode with AAudio. */ - if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) || - ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) { - return MA_SHARE_MODE_NOT_SUPPORTED; + resultOSS = write(pDevice->oss.fdPlayback, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); + if (resultOSS < 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to send data from the client to the device.", MA_FAILED_TO_SEND_DATA_TO_DEVICE); } - /* We first need to try opening the stream. */ - if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { - int32_t framesPerPeriod; - - result = ma_open_stream__aaudio(pContext, ma_device_type_capture, pConfig->capture.pDeviceID, pConfig->capture.shareMode, pConfig, pDevice, (ma_AAudioStream**)&pDevice->aaudio.pStreamCapture); - if (result != MA_SUCCESS) { - return result; /* Failed to open the AAudio stream. */ - } + if (pFramesWritten != NULL) { + *pFramesWritten = (ma_uint32)resultOSS / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + } + + return MA_SUCCESS; +} - pDevice->capture.internalFormat = (((MA_PFN_AAudioStream_getFormat)pContext->aaudio.AAudioStream_getFormat)((ma_AAudioStream*)pDevice->aaudio.pStreamCapture) == MA_AAUDIO_FORMAT_PCM_I16) ? ma_format_s16 : ma_format_f32; - pDevice->capture.internalChannels = ((MA_PFN_AAudioStream_getChannelCount)pContext->aaudio.AAudioStream_getChannelCount)((ma_AAudioStream*)pDevice->aaudio.pStreamCapture); - pDevice->capture.internalSampleRate = ((MA_PFN_AAudioStream_getSampleRate)pContext->aaudio.AAudioStream_getSampleRate)((ma_AAudioStream*)pDevice->aaudio.pStreamCapture); - ma_get_standard_channel_map(ma_standard_channel_map_default, pDevice->capture.internalChannels, pDevice->capture.internalChannelMap); /* <-- Cannot find info on channel order, so assuming a default. */ - pDevice->capture.internalBufferSizeInFrames = ((MA_PFN_AAudioStream_getBufferCapacityInFrames)pContext->aaudio.AAudioStream_getBufferCapacityInFrames)((ma_AAudioStream*)pDevice->aaudio.pStreamCapture); +static ma_result ma_device_read__oss(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead) +{ + int resultOSS; - framesPerPeriod = ((MA_PFN_AAudioStream_getFramesPerDataCallback)pContext->aaudio.AAudioStream_getFramesPerDataCallback)((ma_AAudioStream*)pDevice->aaudio.pStreamCapture); - if (framesPerPeriod > 0) { - pDevice->capture.internalPeriods = pDevice->capture.internalBufferSizeInFrames / framesPerPeriod; - } else { - pDevice->capture.internalPeriods = 1; - } + if (pFramesRead != NULL) { + *pFramesRead = 0; } - if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { - int32_t framesPerPeriod; - - result = ma_open_stream__aaudio(pContext, ma_device_type_playback, pConfig->playback.pDeviceID, pConfig->playback.shareMode, pConfig, pDevice, (ma_AAudioStream**)&pDevice->aaudio.pStreamPlayback); - if (result != MA_SUCCESS) { - return result; /* Failed to open the AAudio stream. */ - } + resultOSS = read(pDevice->oss.fdCapture, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); + if (resultOSS < 0) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to read data from the device to be sent to the client.", MA_FAILED_TO_READ_DATA_FROM_DEVICE); + } + + if (pFramesRead != NULL) { + *pFramesRead = (ma_uint32)resultOSS / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + } - pDevice->playback.internalFormat = (((MA_PFN_AAudioStream_getFormat)pContext->aaudio.AAudioStream_getFormat)((ma_AAudioStream*)pDevice->aaudio.pStreamPlayback) == MA_AAUDIO_FORMAT_PCM_I16) ? ma_format_s16 : ma_format_f32; - pDevice->playback.internalChannels = ((MA_PFN_AAudioStream_getChannelCount)pContext->aaudio.AAudioStream_getChannelCount)((ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); - pDevice->playback.internalSampleRate = ((MA_PFN_AAudioStream_getSampleRate)pContext->aaudio.AAudioStream_getSampleRate)((ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); - ma_get_standard_channel_map(ma_standard_channel_map_default, pDevice->playback.internalChannels, pDevice->playback.internalChannelMap); /* <-- Cannot find info on channel order, so assuming a default. */ - pDevice->playback.internalBufferSizeInFrames = ((MA_PFN_AAudioStream_getBufferCapacityInFrames)pContext->aaudio.AAudioStream_getBufferCapacityInFrames)((ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); + return MA_SUCCESS; +} - framesPerPeriod = ((MA_PFN_AAudioStream_getFramesPerDataCallback)pContext->aaudio.AAudioStream_getFramesPerDataCallback)((ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); - if (framesPerPeriod > 0) { - pDevice->playback.internalPeriods = pDevice->playback.internalBufferSizeInFrames / framesPerPeriod; - } else { - pDevice->playback.internalPeriods = 1; - } - } +static ma_result ma_device_main_loop__oss(ma_device* pDevice) +{ + ma_result result = MA_SUCCESS; + ma_bool32 exitLoop = MA_FALSE; - if (pConfig->deviceType == ma_device_type_duplex) { - ma_uint32 rbSizeInFrames = (ma_uint32)ma_calculate_frame_count_after_src(pDevice->sampleRate, pDevice->capture.internalSampleRate, pDevice->capture.internalBufferSizeInFrames); - ma_result result = ma_pcm_rb_init(pDevice->capture.format, pDevice->capture.channels, rbSizeInFrames, NULL, &pDevice->aaudio.duplexRB); - if (result != MA_SUCCESS) { - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture); - } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); - } - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[AAudio] Failed to initialize ring buffer.", result); - } + /* No need to explicitly start the device like the other backends. */ - /* We need a period to act as a buffer for cases where the playback and capture device's end up desyncing. */ + while (ma_device__get_state(pDevice) == MA_STATE_STARTED && !exitLoop) { + switch (pDevice->type) { - ma_uint32 marginSizeInFrames = rbSizeInFrames / pDevice->capture.internalPeriods; - void* pMarginData; - ma_pcm_rb_acquire_write(&pDevice->aaudio.duplexRB, &marginSizeInFrames, &pMarginData); + case ma_device_type_duplex: { - ma_zero_memory(pMarginData, marginSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels)); - } - ma_pcm_rb_commit_write(&pDevice->aaudio.duplexRB, marginSizeInFrames, pMarginData); - } - } + /* The process is: device_read -> convert -> callback -> convert -> device_write */ + ma_uint32 totalCapturedDeviceFramesProcessed = 0; + ma_uint32 capturedDevicePeriodSizeInFrames = ma_min(pDevice->capture.internalPeriodSizeInFrames, pDevice->playback.internalPeriodSizeInFrames); + + while (totalCapturedDeviceFramesProcessed < capturedDevicePeriodSizeInFrames) { + ma_uint8 capturedDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint8 playbackDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 capturedDeviceDataCapInFrames = sizeof(capturedDeviceData) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + ma_uint32 playbackDeviceDataCapInFrames = sizeof(playbackDeviceData) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + ma_uint32 capturedDeviceFramesRemaining; + ma_uint32 capturedDeviceFramesProcessed; + ma_uint32 capturedDeviceFramesToProcess; + ma_uint32 capturedDeviceFramesToTryProcessing = capturedDevicePeriodSizeInFrames - totalCapturedDeviceFramesProcessed; + if (capturedDeviceFramesToTryProcessing > capturedDeviceDataCapInFrames) { + capturedDeviceFramesToTryProcessing = capturedDeviceDataCapInFrames; + } - return MA_SUCCESS; -} + result = ma_device_read__oss(pDevice, capturedDeviceData, capturedDeviceFramesToTryProcessing, &capturedDeviceFramesToProcess); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } -ma_result ma_device_start_stream__aaudio(ma_device* pDevice, ma_AAudioStream* pStream) -{ - ma_aaudio_result_t resultAA; - ma_aaudio_stream_state_t currentState; + capturedDeviceFramesRemaining = capturedDeviceFramesToProcess; + capturedDeviceFramesProcessed = 0; + + for (;;) { + ma_uint8 capturedClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint8 playbackClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 capturedClientDataCapInFrames = sizeof(capturedClientData) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels); + ma_uint32 playbackClientDataCapInFrames = sizeof(playbackClientData) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels); + ma_uint64 capturedClientFramesToProcessThisIteration = ma_min(capturedClientDataCapInFrames, playbackClientDataCapInFrames); + ma_uint64 capturedDeviceFramesToProcessThisIteration = capturedDeviceFramesRemaining; + ma_uint8* pRunningCapturedDeviceFrames = ma_offset_ptr(capturedDeviceData, capturedDeviceFramesProcessed * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels)); + + /* Convert capture data from device format to client format. */ + result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningCapturedDeviceFrames, &capturedDeviceFramesToProcessThisIteration, capturedClientData, &capturedClientFramesToProcessThisIteration); + if (result != MA_SUCCESS) { + break; + } - ma_assert(pDevice != NULL); + /* + If we weren't able to generate any output frames it must mean we've exhaused all of our input. The only time this would not be the case is if capturedClientData was too small + which should never be the case when it's of the size MA_DATA_CONVERTER_STACK_BUFFER_SIZE. + */ + if (capturedClientFramesToProcessThisIteration == 0) { + break; + } - resultAA = ((MA_PFN_AAudioStream_requestStart)pDevice->pContext->aaudio.AAudioStream_requestStart)(pStream); - if (resultAA != MA_AAUDIO_OK) { - return ma_result_from_aaudio(resultAA); - } + ma_device__on_data(pDevice, playbackClientData, capturedClientData, (ma_uint32)capturedClientFramesToProcessThisIteration); /* Safe cast .*/ - /* Do we actually need to wait for the device to transition into it's started state? */ + capturedDeviceFramesProcessed += (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */ + capturedDeviceFramesRemaining -= (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */ - /* The device should be in either a starting or started state. If it's not set to started we need to wait for it to transition. It should go from starting to started. */ - currentState = ((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream); - if (currentState != MA_AAUDIO_STREAM_STATE_STARTED) { - ma_result result; + /* At this point the playbackClientData buffer should be holding data that needs to be written to the device. */ + for (;;) { + ma_uint64 convertedClientFrameCount = capturedClientFramesToProcessThisIteration; + ma_uint64 convertedDeviceFrameCount = playbackDeviceDataCapInFrames; + result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, playbackClientData, &convertedClientFrameCount, playbackDeviceData, &convertedDeviceFrameCount); + if (result != MA_SUCCESS) { + break; + } - if (currentState != MA_AAUDIO_STREAM_STATE_STARTING) { - return MA_ERROR; /* Expecting the stream to be a starting or started state. */ - } + result = ma_device_write__oss(pDevice, playbackDeviceData, (ma_uint32)convertedDeviceFrameCount, NULL); /* Safe cast. */ + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } - result = ma_wait_for_simple_state_transition__aaudio(pDevice->pContext, pStream, currentState, MA_AAUDIO_STREAM_STATE_STARTED); - if (result != MA_SUCCESS) { - return result; - } - } + capturedClientFramesToProcessThisIteration -= (ma_uint32)convertedClientFrameCount; /* Safe cast. */ + if (capturedClientFramesToProcessThisIteration == 0) { + break; + } + } - return MA_SUCCESS; -} + /* In case an error happened from ma_device_write__oss()... */ + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } + } + + totalCapturedDeviceFramesProcessed += capturedDeviceFramesProcessed; + } + } break; + + case ma_device_type_capture: + { + /* We read in chunks of the period size, but use a stack allocated buffer for the intermediary. */ + ma_uint8 intermediaryBuffer[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + ma_uint32 periodSizeInFrames = pDevice->capture.internalPeriodSizeInFrames; + ma_uint32 framesReadThisPeriod = 0; + while (framesReadThisPeriod < periodSizeInFrames) { + ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesReadThisPeriod; + ma_uint32 framesProcessed; + ma_uint32 framesToReadThisIteration = framesRemainingInPeriod; + if (framesToReadThisIteration > intermediaryBufferSizeInFrames) { + framesToReadThisIteration = intermediaryBufferSizeInFrames; + } + + result = ma_device_read__oss(pDevice, intermediaryBuffer, framesToReadThisIteration, &framesProcessed); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } -ma_result ma_device_stop_stream__aaudio(ma_device* pDevice, ma_AAudioStream* pStream) -{ - ma_aaudio_result_t resultAA; - ma_aaudio_stream_state_t currentState; + ma_device__send_frames_to_client(pDevice, framesProcessed, intermediaryBuffer); - ma_assert(pDevice != NULL); + framesReadThisPeriod += framesProcessed; + } + } break; - resultAA = ((MA_PFN_AAudioStream_requestStop)pDevice->pContext->aaudio.AAudioStream_requestStop)(pStream); - if (resultAA != MA_AAUDIO_OK) { - return ma_result_from_aaudio(resultAA); - } + case ma_device_type_playback: + { + /* We write in chunks of the period size, but use a stack allocated buffer for the intermediary. */ + ma_uint8 intermediaryBuffer[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + ma_uint32 intermediaryBufferSizeInFrames = sizeof(intermediaryBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + ma_uint32 periodSizeInFrames = pDevice->playback.internalPeriodSizeInFrames; + ma_uint32 framesWrittenThisPeriod = 0; + while (framesWrittenThisPeriod < periodSizeInFrames) { + ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesWrittenThisPeriod; + ma_uint32 framesProcessed; + ma_uint32 framesToWriteThisIteration = framesRemainingInPeriod; + if (framesToWriteThisIteration > intermediaryBufferSizeInFrames) { + framesToWriteThisIteration = intermediaryBufferSizeInFrames; + } - /* The device should be in either a stopping or stopped state. If it's not set to started we need to wait for it to transition. It should go from stopping to stopped. */ - currentState = ((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream); - if (currentState != MA_AAUDIO_STREAM_STATE_STOPPED) { - ma_result result; + ma_device__read_frames_from_client(pDevice, framesToWriteThisIteration, intermediaryBuffer); - if (currentState != MA_AAUDIO_STREAM_STATE_STOPPING) { - return MA_ERROR; /* Expecting the stream to be a stopping or stopped state. */ - } + result = ma_device_write__oss(pDevice, intermediaryBuffer, framesToWriteThisIteration, &framesProcessed); + if (result != MA_SUCCESS) { + exitLoop = MA_TRUE; + break; + } - result = ma_wait_for_simple_state_transition__aaudio(pDevice->pContext, pStream, currentState, MA_AAUDIO_STREAM_STATE_STOPPED); - if (result != MA_SUCCESS) { - return result; + framesWrittenThisPeriod += framesProcessed; + } + } break; + + /* To silence a warning. Will never hit this. */ + case ma_device_type_loopback: + default: break; } } - return MA_SUCCESS; -} -ma_result ma_device_start__aaudio(ma_device* pDevice) -{ - ma_assert(pDevice != NULL); + /* Here is where the device is stopped. */ + ma_device_stop__oss(pDevice); - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - ma_result result = ma_device_start_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture); - if (result != MA_SUCCESS) { - return result; - } - } + return result; +} - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - ma_result result = ma_device_start_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); - if (result != MA_SUCCESS) { - if (pDevice->type == ma_device_type_duplex) { - ma_device_stop_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture); - } - return result; - } - } +static ma_result ma_context_uninit__oss(ma_context* pContext) +{ + MA_ASSERT(pContext != NULL); + MA_ASSERT(pContext->backend == ma_backend_oss); + (void)pContext; return MA_SUCCESS; } -ma_result ma_device_stop__aaudio(ma_device* pDevice) +static ma_result ma_context_init__oss(const ma_context_config* pConfig, ma_context* pContext) { - ma_stop_proc onStop; + int fd; + int ossVersion; + int result; - ma_assert(pDevice != NULL); + MA_ASSERT(pContext != NULL); - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - ma_result result = ma_device_stop_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture); - if (result != MA_SUCCESS) { - return result; - } - } + (void)pConfig; - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - ma_result result = ma_device_stop_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); - if (result != MA_SUCCESS) { - return result; - } + /* Try opening a temporary device first so we can get version information. This is closed at the end. */ + fd = ma_open_temp_device__oss(); + if (fd == -1) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[OSS] Failed to open temporary device for retrieving system properties.", MA_NO_BACKEND); /* Looks liks OSS isn't installed, or there are no available devices. */ } - onStop = pDevice->onStop; - if (onStop) { - onStop(pDevice); + /* Grab the OSS version. */ + ossVersion = 0; + result = ioctl(fd, OSS_GETVERSION, &ossVersion); + if (result == -1) { + close(fd); + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[OSS] Failed to retrieve OSS version.", MA_NO_BACKEND); } - return MA_SUCCESS; -} - + pContext->oss.versionMajor = ((ossVersion & 0xFF0000) >> 16); + pContext->oss.versionMinor = ((ossVersion & 0x00FF00) >> 8); -ma_result ma_context_uninit__aaudio(ma_context* pContext) -{ - ma_assert(pContext != NULL); - ma_assert(pContext->backend == ma_backend_aaudio); - - ma_dlclose(pContext, pContext->aaudio.hAAudio); - pContext->aaudio.hAAudio = NULL; + pContext->onUninit = ma_context_uninit__oss; + pContext->onDeviceIDEqual = ma_context_is_device_id_equal__oss; + pContext->onEnumDevices = ma_context_enumerate_devices__oss; + pContext->onGetDeviceInfo = ma_context_get_device_info__oss; + pContext->onDeviceInit = ma_device_init__oss; + pContext->onDeviceUninit = ma_device_uninit__oss; + pContext->onDeviceStart = NULL; /* Not required for synchronous backends. */ + pContext->onDeviceStop = NULL; /* Not required for synchronous backends. */ + pContext->onDeviceMainLoop = ma_device_main_loop__oss; + close(fd); return MA_SUCCESS; } +#endif /* OSS */ -ma_result ma_context_init__aaudio(const ma_context_config* pConfig, ma_context* pContext) -{ - const char* libNames[] = { - "libaaudio.so" - }; - size_t i; - for (i = 0; i < ma_countof(libNames); ++i) { - pContext->aaudio.hAAudio = ma_dlopen(pContext, libNames[i]); - if (pContext->aaudio.hAAudio != NULL) { - break; - } - } +/****************************************************************************** - if (pContext->aaudio.hAAudio == NULL) { - return MA_FAILED_TO_INIT_BACKEND; - } +AAudio Backend - pContext->aaudio.AAudio_createStreamBuilder = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudio_createStreamBuilder"); - pContext->aaudio.AAudioStreamBuilder_delete = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_delete"); - pContext->aaudio.AAudioStreamBuilder_setDeviceId = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setDeviceId"); - pContext->aaudio.AAudioStreamBuilder_setDirection = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setDirection"); - pContext->aaudio.AAudioStreamBuilder_setSharingMode = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setSharingMode"); - pContext->aaudio.AAudioStreamBuilder_setFormat = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setFormat"); - pContext->aaudio.AAudioStreamBuilder_setChannelCount = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setChannelCount"); - pContext->aaudio.AAudioStreamBuilder_setSampleRate = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setSampleRate"); - pContext->aaudio.AAudioStreamBuilder_setBufferCapacityInFrames = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setBufferCapacityInFrames"); - pContext->aaudio.AAudioStreamBuilder_setFramesPerDataCallback = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setFramesPerDataCallback"); - pContext->aaudio.AAudioStreamBuilder_setDataCallback = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setDataCallback"); - pContext->aaudio.AAudioStreamBuilder_setPerformanceMode = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setPerformanceMode"); - pContext->aaudio.AAudioStreamBuilder_openStream = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_openStream"); - pContext->aaudio.AAudioStream_close = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_close"); - pContext->aaudio.AAudioStream_getState = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getState"); - pContext->aaudio.AAudioStream_waitForStateChange = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_waitForStateChange"); - pContext->aaudio.AAudioStream_getFormat = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getFormat"); - pContext->aaudio.AAudioStream_getChannelCount = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getChannelCount"); - pContext->aaudio.AAudioStream_getSampleRate = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getSampleRate"); - pContext->aaudio.AAudioStream_getBufferCapacityInFrames = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getBufferCapacityInFrames"); - pContext->aaudio.AAudioStream_getFramesPerDataCallback = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getFramesPerDataCallback"); - pContext->aaudio.AAudioStream_getFramesPerBurst = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getFramesPerBurst"); - pContext->aaudio.AAudioStream_requestStart = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_requestStart"); - pContext->aaudio.AAudioStream_requestStop = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_requestStop"); +******************************************************************************/ +#ifdef MA_HAS_AAUDIO +/*#include */ - pContext->isBackendAsynchronous = MA_TRUE; +#define MA_AAUDIO_UNSPECIFIED 0 - pContext->onUninit = ma_context_uninit__aaudio; - pContext->onDeviceIDEqual = ma_context_is_device_id_equal__aaudio; - pContext->onEnumDevices = ma_context_enumerate_devices__aaudio; - pContext->onGetDeviceInfo = ma_context_get_device_info__aaudio; - pContext->onDeviceInit = ma_device_init__aaudio; - pContext->onDeviceUninit = ma_device_uninit__aaudio; - pContext->onDeviceStart = ma_device_start__aaudio; - pContext->onDeviceStop = ma_device_stop__aaudio; +typedef int32_t ma_aaudio_result_t; +typedef int32_t ma_aaudio_direction_t; +typedef int32_t ma_aaudio_sharing_mode_t; +typedef int32_t ma_aaudio_format_t; +typedef int32_t ma_aaudio_stream_state_t; +typedef int32_t ma_aaudio_performance_mode_t; +typedef int32_t ma_aaudio_data_callback_result_t; - (void)pConfig; - return MA_SUCCESS; -} -#endif /* AAudio */ +/* Result codes. miniaudio only cares about the success code. */ +#define MA_AAUDIO_OK 0 +/* Directions. */ +#define MA_AAUDIO_DIRECTION_OUTPUT 0 +#define MA_AAUDIO_DIRECTION_INPUT 1 -/****************************************************************************** +/* Sharing modes. */ +#define MA_AAUDIO_SHARING_MODE_EXCLUSIVE 0 +#define MA_AAUDIO_SHARING_MODE_SHARED 1 -OpenSL|ES Backend +/* Formats. */ +#define MA_AAUDIO_FORMAT_PCM_I16 1 +#define MA_AAUDIO_FORMAT_PCM_FLOAT 2 -******************************************************************************/ -#ifdef MA_HAS_OPENSL -#include -#ifdef MA_ANDROID -#include -#endif +/* Stream states. */ +#define MA_AAUDIO_STREAM_STATE_UNINITIALIZED 0 +#define MA_AAUDIO_STREAM_STATE_UNKNOWN 1 +#define MA_AAUDIO_STREAM_STATE_OPEN 2 +#define MA_AAUDIO_STREAM_STATE_STARTING 3 +#define MA_AAUDIO_STREAM_STATE_STARTED 4 +#define MA_AAUDIO_STREAM_STATE_PAUSING 5 +#define MA_AAUDIO_STREAM_STATE_PAUSED 6 +#define MA_AAUDIO_STREAM_STATE_FLUSHING 7 +#define MA_AAUDIO_STREAM_STATE_FLUSHED 8 +#define MA_AAUDIO_STREAM_STATE_STOPPING 9 +#define MA_AAUDIO_STREAM_STATE_STOPPED 10 +#define MA_AAUDIO_STREAM_STATE_CLOSING 11 +#define MA_AAUDIO_STREAM_STATE_CLOSED 12 +#define MA_AAUDIO_STREAM_STATE_DISCONNECTED 13 -/* OpenSL|ES has one-per-application objects :( */ -SLObjectItf g_maEngineObjectSL = NULL; -SLEngineItf g_maEngineSL = NULL; -ma_uint32 g_maOpenSLInitCounter = 0; +/* Performance modes. */ +#define MA_AAUDIO_PERFORMANCE_MODE_NONE 10 +#define MA_AAUDIO_PERFORMANCE_MODE_POWER_SAVING 11 +#define MA_AAUDIO_PERFORMANCE_MODE_LOW_LATENCY 12 -#define MA_OPENSL_OBJ(p) (*((SLObjectItf)(p))) -#define MA_OPENSL_OUTPUTMIX(p) (*((SLOutputMixItf)(p))) -#define MA_OPENSL_PLAY(p) (*((SLPlayItf)(p))) -#define MA_OPENSL_RECORD(p) (*((SLRecordItf)(p))) +/* Callback results. */ +#define MA_AAUDIO_CALLBACK_RESULT_CONTINUE 0 +#define MA_AAUDIO_CALLBACK_RESULT_STOP 1 -#ifdef MA_ANDROID -#define MA_OPENSL_BUFFERQUEUE(p) (*((SLAndroidSimpleBufferQueueItf)(p))) -#else -#define MA_OPENSL_BUFFERQUEUE(p) (*((SLBufferQueueItf)(p))) -#endif +/* Objects. */ +typedef struct ma_AAudioStreamBuilder_t* ma_AAudioStreamBuilder; +typedef struct ma_AAudioStream_t* ma_AAudioStream; + +typedef ma_aaudio_data_callback_result_t (* ma_AAudioStream_dataCallback) (ma_AAudioStream* pStream, void* pUserData, void* pAudioData, int32_t numFrames); +typedef void (* ma_AAudioStream_errorCallback)(ma_AAudioStream *pStream, void *pUserData, ma_aaudio_result_t error); + +typedef ma_aaudio_result_t (* MA_PFN_AAudio_createStreamBuilder) (ma_AAudioStreamBuilder** ppBuilder); +typedef ma_aaudio_result_t (* MA_PFN_AAudioStreamBuilder_delete) (ma_AAudioStreamBuilder* pBuilder); +typedef void (* MA_PFN_AAudioStreamBuilder_setDeviceId) (ma_AAudioStreamBuilder* pBuilder, int32_t deviceId); +typedef void (* MA_PFN_AAudioStreamBuilder_setDirection) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_direction_t direction); +typedef void (* MA_PFN_AAudioStreamBuilder_setSharingMode) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_sharing_mode_t sharingMode); +typedef void (* MA_PFN_AAudioStreamBuilder_setFormat) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_format_t format); +typedef void (* MA_PFN_AAudioStreamBuilder_setChannelCount) (ma_AAudioStreamBuilder* pBuilder, int32_t channelCount); +typedef void (* MA_PFN_AAudioStreamBuilder_setSampleRate) (ma_AAudioStreamBuilder* pBuilder, int32_t sampleRate); +typedef void (* MA_PFN_AAudioStreamBuilder_setBufferCapacityInFrames)(ma_AAudioStreamBuilder* pBuilder, int32_t numFrames); +typedef void (* MA_PFN_AAudioStreamBuilder_setFramesPerDataCallback) (ma_AAudioStreamBuilder* pBuilder, int32_t numFrames); +typedef void (* MA_PFN_AAudioStreamBuilder_setDataCallback) (ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream_dataCallback callback, void* pUserData); +typedef void (* MA_PFN_AAudioStreamBuilder_setErrorCallback) (ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream_errorCallback callback, void* pUserData); +typedef void (* MA_PFN_AAudioStreamBuilder_setPerformanceMode) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_performance_mode_t mode); +typedef ma_aaudio_result_t (* MA_PFN_AAudioStreamBuilder_openStream) (ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream** ppStream); +typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_close) (ma_AAudioStream* pStream); +typedef ma_aaudio_stream_state_t (* MA_PFN_AAudioStream_getState) (ma_AAudioStream* pStream); +typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_waitForStateChange) (ma_AAudioStream* pStream, ma_aaudio_stream_state_t inputState, ma_aaudio_stream_state_t* pNextState, int64_t timeoutInNanoseconds); +typedef ma_aaudio_format_t (* MA_PFN_AAudioStream_getFormat) (ma_AAudioStream* pStream); +typedef int32_t (* MA_PFN_AAudioStream_getChannelCount) (ma_AAudioStream* pStream); +typedef int32_t (* MA_PFN_AAudioStream_getSampleRate) (ma_AAudioStream* pStream); +typedef int32_t (* MA_PFN_AAudioStream_getBufferCapacityInFrames) (ma_AAudioStream* pStream); +typedef int32_t (* MA_PFN_AAudioStream_getFramesPerDataCallback) (ma_AAudioStream* pStream); +typedef int32_t (* MA_PFN_AAudioStream_getFramesPerBurst) (ma_AAudioStream* pStream); +typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_requestStart) (ma_AAudioStream* pStream); +typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_requestStop) (ma_AAudioStream* pStream); -/* Converts an individual OpenSL-style channel identifier (SL_SPEAKER_FRONT_LEFT, etc.) to miniaudio. */ -ma_uint8 ma_channel_id_to_ma__opensl(SLuint32 id) +static ma_result ma_result_from_aaudio(ma_aaudio_result_t resultAA) { - switch (id) + switch (resultAA) { - case SL_SPEAKER_FRONT_LEFT: return MA_CHANNEL_FRONT_LEFT; - case SL_SPEAKER_FRONT_RIGHT: return MA_CHANNEL_FRONT_RIGHT; - case SL_SPEAKER_FRONT_CENTER: return MA_CHANNEL_FRONT_CENTER; - case SL_SPEAKER_LOW_FREQUENCY: return MA_CHANNEL_LFE; - case SL_SPEAKER_BACK_LEFT: return MA_CHANNEL_BACK_LEFT; - case SL_SPEAKER_BACK_RIGHT: return MA_CHANNEL_BACK_RIGHT; - case SL_SPEAKER_FRONT_LEFT_OF_CENTER: return MA_CHANNEL_FRONT_LEFT_CENTER; - case SL_SPEAKER_FRONT_RIGHT_OF_CENTER: return MA_CHANNEL_FRONT_RIGHT_CENTER; - case SL_SPEAKER_BACK_CENTER: return MA_CHANNEL_BACK_CENTER; - case SL_SPEAKER_SIDE_LEFT: return MA_CHANNEL_SIDE_LEFT; - case SL_SPEAKER_SIDE_RIGHT: return MA_CHANNEL_SIDE_RIGHT; - case SL_SPEAKER_TOP_CENTER: return MA_CHANNEL_TOP_CENTER; - case SL_SPEAKER_TOP_FRONT_LEFT: return MA_CHANNEL_TOP_FRONT_LEFT; - case SL_SPEAKER_TOP_FRONT_CENTER: return MA_CHANNEL_TOP_FRONT_CENTER; - case SL_SPEAKER_TOP_FRONT_RIGHT: return MA_CHANNEL_TOP_FRONT_RIGHT; - case SL_SPEAKER_TOP_BACK_LEFT: return MA_CHANNEL_TOP_BACK_LEFT; - case SL_SPEAKER_TOP_BACK_CENTER: return MA_CHANNEL_TOP_BACK_CENTER; - case SL_SPEAKER_TOP_BACK_RIGHT: return MA_CHANNEL_TOP_BACK_RIGHT; - default: return 0; + case MA_AAUDIO_OK: return MA_SUCCESS; + default: break; } + + return MA_ERROR; } -/* Converts an individual miniaudio channel identifier (MA_CHANNEL_FRONT_LEFT, etc.) to OpenSL-style. */ -SLuint32 ma_channel_id_to_opensl(ma_uint8 id) +static void ma_stream_error_callback__aaudio(ma_AAudioStream* pStream, void* pUserData, ma_aaudio_result_t error) { - switch (id) - { - case MA_CHANNEL_MONO: return SL_SPEAKER_FRONT_CENTER; - case MA_CHANNEL_FRONT_LEFT: return SL_SPEAKER_FRONT_LEFT; - case MA_CHANNEL_FRONT_RIGHT: return SL_SPEAKER_FRONT_RIGHT; - case MA_CHANNEL_FRONT_CENTER: return SL_SPEAKER_FRONT_CENTER; - case MA_CHANNEL_LFE: return SL_SPEAKER_LOW_FREQUENCY; - case MA_CHANNEL_BACK_LEFT: return SL_SPEAKER_BACK_LEFT; - case MA_CHANNEL_BACK_RIGHT: return SL_SPEAKER_BACK_RIGHT; - case MA_CHANNEL_FRONT_LEFT_CENTER: return SL_SPEAKER_FRONT_LEFT_OF_CENTER; - case MA_CHANNEL_FRONT_RIGHT_CENTER: return SL_SPEAKER_FRONT_RIGHT_OF_CENTER; - case MA_CHANNEL_BACK_CENTER: return SL_SPEAKER_BACK_CENTER; - case MA_CHANNEL_SIDE_LEFT: return SL_SPEAKER_SIDE_LEFT; - case MA_CHANNEL_SIDE_RIGHT: return SL_SPEAKER_SIDE_RIGHT; - case MA_CHANNEL_TOP_CENTER: return SL_SPEAKER_TOP_CENTER; - case MA_CHANNEL_TOP_FRONT_LEFT: return SL_SPEAKER_TOP_FRONT_LEFT; - case MA_CHANNEL_TOP_FRONT_CENTER: return SL_SPEAKER_TOP_FRONT_CENTER; - case MA_CHANNEL_TOP_FRONT_RIGHT: return SL_SPEAKER_TOP_FRONT_RIGHT; - case MA_CHANNEL_TOP_BACK_LEFT: return SL_SPEAKER_TOP_BACK_LEFT; - case MA_CHANNEL_TOP_BACK_CENTER: return SL_SPEAKER_TOP_BACK_CENTER; - case MA_CHANNEL_TOP_BACK_RIGHT: return SL_SPEAKER_TOP_BACK_RIGHT; - default: return 0; + ma_device* pDevice = (ma_device*)pUserData; + MA_ASSERT(pDevice != NULL); + + (void)error; + +#if defined(MA_DEBUG_OUTPUT) + printf("[AAudio] ERROR CALLBACK: error=%d, AAudioStream_getState()=%d\n", error, ((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream)); +#endif + + /* + From the documentation for AAudio, when a device is disconnected all we can do is stop it. However, we cannot stop it from the callback - we need + to do it from another thread. Therefore we are going to use an event thread for the AAudio backend to do this cleanly and safely. + */ + if (((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream) == MA_AAUDIO_STREAM_STATE_DISCONNECTED) { +#if defined(MA_DEBUG_OUTPUT) + printf("[AAudio] Device Disconnected.\n"); +#endif } } -/* Converts a channel mapping to an OpenSL-style channel mask. */ -SLuint32 ma_channel_map_to_channel_mask__opensl(const ma_channel channelMap[MA_MAX_CHANNELS], ma_uint32 channels) +static ma_aaudio_data_callback_result_t ma_stream_data_callback_capture__aaudio(ma_AAudioStream* pStream, void* pUserData, void* pAudioData, int32_t frameCount) { - SLuint32 channelMask = 0; - ma_uint32 iChannel; - for (iChannel = 0; iChannel < channels; ++iChannel) { - channelMask |= ma_channel_id_to_opensl(channelMap[iChannel]); + ma_device* pDevice = (ma_device*)pUserData; + MA_ASSERT(pDevice != NULL); + + if (pDevice->type == ma_device_type_duplex) { + ma_device__handle_duplex_callback_capture(pDevice, frameCount, pAudioData, &pDevice->aaudio.duplexRB); + } else { + ma_device__send_frames_to_client(pDevice, frameCount, pAudioData); /* Send directly to the client. */ } - return channelMask; + (void)pStream; + return MA_AAUDIO_CALLBACK_RESULT_CONTINUE; } -/* Converts an OpenSL-style channel mask to a miniaudio channel map. */ -void ma_channel_mask_to_channel_map__opensl(SLuint32 channelMask, ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) +static ma_aaudio_data_callback_result_t ma_stream_data_callback_playback__aaudio(ma_AAudioStream* pStream, void* pUserData, void* pAudioData, int32_t frameCount) { - if (channels == 1 && channelMask == 0) { - channelMap[0] = MA_CHANNEL_MONO; - } else if (channels == 2 && channelMask == 0) { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + ma_device* pDevice = (ma_device*)pUserData; + MA_ASSERT(pDevice != NULL); + + if (pDevice->type == ma_device_type_duplex) { + ma_device__handle_duplex_callback_playback(pDevice, frameCount, pAudioData, &pDevice->aaudio.duplexRB); } else { - if (channels == 1 && (channelMask & SL_SPEAKER_FRONT_CENTER) != 0) { - channelMap[0] = MA_CHANNEL_MONO; - } else { - /* Just iterate over each bit. */ - ma_uint32 iChannel = 0; - ma_uint32 iBit; - for (iBit = 0; iBit < 32; ++iBit) { - SLuint32 bitValue = (channelMask & (1UL << iBit)); - if (bitValue != 0) { - /* The bit is set. */ - channelMap[iChannel] = ma_channel_id_to_ma__opensl(bitValue); - iChannel += 1; - } - } - } + ma_device__read_frames_from_client(pDevice, frameCount, pAudioData); /* Read directly from the client. */ } + + (void)pStream; + return MA_AAUDIO_CALLBACK_RESULT_CONTINUE; } -SLuint32 ma_round_to_standard_sample_rate__opensl(SLuint32 samplesPerSec) +static ma_result ma_open_stream__aaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, const ma_device_config* pConfig, const ma_device* pDevice, ma_AAudioStream** ppStream) { - if (samplesPerSec <= SL_SAMPLINGRATE_8) { - return SL_SAMPLINGRATE_8; - } - if (samplesPerSec <= SL_SAMPLINGRATE_11_025) { - return SL_SAMPLINGRATE_11_025; - } - if (samplesPerSec <= SL_SAMPLINGRATE_12) { - return SL_SAMPLINGRATE_12; - } - if (samplesPerSec <= SL_SAMPLINGRATE_16) { - return SL_SAMPLINGRATE_16; - } - if (samplesPerSec <= SL_SAMPLINGRATE_22_05) { - return SL_SAMPLINGRATE_22_05; - } - if (samplesPerSec <= SL_SAMPLINGRATE_24) { - return SL_SAMPLINGRATE_24; - } - if (samplesPerSec <= SL_SAMPLINGRATE_32) { - return SL_SAMPLINGRATE_32; - } - if (samplesPerSec <= SL_SAMPLINGRATE_44_1) { - return SL_SAMPLINGRATE_44_1; - } - if (samplesPerSec <= SL_SAMPLINGRATE_48) { - return SL_SAMPLINGRATE_48; - } + ma_AAudioStreamBuilder* pBuilder; + ma_aaudio_result_t resultAA; - /* Android doesn't support more than 48000. */ -#ifndef MA_ANDROID - if (samplesPerSec <= SL_SAMPLINGRATE_64) { - return SL_SAMPLINGRATE_64; - } - if (samplesPerSec <= SL_SAMPLINGRATE_88_2) { - return SL_SAMPLINGRATE_88_2; - } - if (samplesPerSec <= SL_SAMPLINGRATE_96) { - return SL_SAMPLINGRATE_96; + MA_ASSERT(deviceType != ma_device_type_duplex); /* This function should not be called for a full-duplex device type. */ + + *ppStream = NULL; + + resultAA = ((MA_PFN_AAudio_createStreamBuilder)pContext->aaudio.AAudio_createStreamBuilder)(&pBuilder); + if (resultAA != MA_AAUDIO_OK) { + return ma_result_from_aaudio(resultAA); } - if (samplesPerSec <= SL_SAMPLINGRATE_192) { - return SL_SAMPLINGRATE_192; + + if (pDeviceID != NULL) { + ((MA_PFN_AAudioStreamBuilder_setDeviceId)pContext->aaudio.AAudioStreamBuilder_setDeviceId)(pBuilder, pDeviceID->aaudio); } -#endif - return SL_SAMPLINGRATE_16; -} + ((MA_PFN_AAudioStreamBuilder_setDirection)pContext->aaudio.AAudioStreamBuilder_setDirection)(pBuilder, (deviceType == ma_device_type_playback) ? MA_AAUDIO_DIRECTION_OUTPUT : MA_AAUDIO_DIRECTION_INPUT); + ((MA_PFN_AAudioStreamBuilder_setSharingMode)pContext->aaudio.AAudioStreamBuilder_setSharingMode)(pBuilder, (shareMode == ma_share_mode_shared) ? MA_AAUDIO_SHARING_MODE_SHARED : MA_AAUDIO_SHARING_MODE_EXCLUSIVE); + if (pConfig != NULL) { + ma_uint32 bufferCapacityInFrames; -ma_bool32 ma_context_is_device_id_equal__opensl(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) -{ - ma_assert(pContext != NULL); - ma_assert(pID0 != NULL); - ma_assert(pID1 != NULL); - (void)pContext; + if (pDevice == NULL || !pDevice->usingDefaultSampleRate) { + ((MA_PFN_AAudioStreamBuilder_setSampleRate)pContext->aaudio.AAudioStreamBuilder_setSampleRate)(pBuilder, pConfig->sampleRate); + } - return pID0->opensl == pID1->opensl; -} + if (deviceType == ma_device_type_capture) { + if (pDevice == NULL || !pDevice->capture.usingDefaultChannels) { + ((MA_PFN_AAudioStreamBuilder_setChannelCount)pContext->aaudio.AAudioStreamBuilder_setChannelCount)(pBuilder, pConfig->capture.channels); + } + if (pDevice == NULL || !pDevice->capture.usingDefaultFormat) { + ((MA_PFN_AAudioStreamBuilder_setFormat)pContext->aaudio.AAudioStreamBuilder_setFormat)(pBuilder, (pConfig->capture.format == ma_format_s16) ? MA_AAUDIO_FORMAT_PCM_I16 : MA_AAUDIO_FORMAT_PCM_FLOAT); + } + } else { + if (pDevice == NULL || !pDevice->playback.usingDefaultChannels) { + ((MA_PFN_AAudioStreamBuilder_setChannelCount)pContext->aaudio.AAudioStreamBuilder_setChannelCount)(pBuilder, pConfig->playback.channels); + } + if (pDevice == NULL || !pDevice->playback.usingDefaultFormat) { + ((MA_PFN_AAudioStreamBuilder_setFormat)pContext->aaudio.AAudioStreamBuilder_setFormat)(pBuilder, (pConfig->playback.format == ma_format_s16) ? MA_AAUDIO_FORMAT_PCM_I16 : MA_AAUDIO_FORMAT_PCM_FLOAT); + } + } -ma_result ma_context_enumerate_devices__opensl(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) -{ - ma_bool32 cbResult; + bufferCapacityInFrames = pConfig->periodSizeInFrames * pConfig->periods; + if (bufferCapacityInFrames == 0) { + bufferCapacityInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->periodSizeInMilliseconds, pConfig->sampleRate) * pConfig->periods; + } - ma_assert(pContext != NULL); - ma_assert(callback != NULL); + ((MA_PFN_AAudioStreamBuilder_setBufferCapacityInFrames)pContext->aaudio.AAudioStreamBuilder_setBufferCapacityInFrames)(pBuilder, bufferCapacityInFrames); + ((MA_PFN_AAudioStreamBuilder_setFramesPerDataCallback)pContext->aaudio.AAudioStreamBuilder_setFramesPerDataCallback)(pBuilder, bufferCapacityInFrames / pConfig->periods); - ma_assert(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to enumerate devices. */ - if (g_maOpenSLInitCounter == 0) { - return MA_INVALID_OPERATION; - } + if (deviceType == ma_device_type_capture) { + ((MA_PFN_AAudioStreamBuilder_setDataCallback)pContext->aaudio.AAudioStreamBuilder_setDataCallback)(pBuilder, ma_stream_data_callback_capture__aaudio, (void*)pDevice); + } else { + ((MA_PFN_AAudioStreamBuilder_setDataCallback)pContext->aaudio.AAudioStreamBuilder_setDataCallback)(pBuilder, ma_stream_data_callback_playback__aaudio, (void*)pDevice); + } - /* - TODO: Test Me. - - This is currently untested, so for now we are just returning default devices. - */ -#if 0 && !defined(MA_ANDROID) - ma_bool32 isTerminated = MA_FALSE; + /* Not sure how this affects things, but since there's a mapping between miniaudio's performance profiles and AAudio's performance modes, let go ahead and set it. */ + ((MA_PFN_AAudioStreamBuilder_setPerformanceMode)pContext->aaudio.AAudioStreamBuilder_setPerformanceMode)(pBuilder, (pConfig->performanceProfile == ma_performance_profile_low_latency) ? MA_AAUDIO_PERFORMANCE_MODE_LOW_LATENCY : MA_AAUDIO_PERFORMANCE_MODE_NONE); + } - SLuint32 pDeviceIDs[128]; - SLint32 deviceCount = sizeof(pDeviceIDs) / sizeof(pDeviceIDs[0]); + ((MA_PFN_AAudioStreamBuilder_setErrorCallback)pContext->aaudio.AAudioStreamBuilder_setErrorCallback)(pBuilder, ma_stream_error_callback__aaudio, (void*)pDevice); - SLAudioIODeviceCapabilitiesItf deviceCaps; - SLresult resultSL = (*g_maEngineObjectSL)->GetInterface(g_maEngineObjectSL, SL_IID_AUDIOIODEVICECAPABILITIES, &deviceCaps); - if (resultSL != SL_RESULT_SUCCESS) { - /* The interface may not be supported so just report a default device. */ - goto return_default_device; + resultAA = ((MA_PFN_AAudioStreamBuilder_openStream)pContext->aaudio.AAudioStreamBuilder_openStream)(pBuilder, ppStream); + if (resultAA != MA_AAUDIO_OK) { + *ppStream = NULL; + ((MA_PFN_AAudioStreamBuilder_delete)pContext->aaudio.AAudioStreamBuilder_delete)(pBuilder); + return ma_result_from_aaudio(resultAA); } - /* Playback */ - if (!isTerminated) { - resultSL = (*deviceCaps)->GetAvailableAudioOutputs(deviceCaps, &deviceCount, pDeviceIDs); - if (resultSL != SL_RESULT_SUCCESS) { - return MA_NO_DEVICE; - } - - for (SLint32 iDevice = 0; iDevice < deviceCount; ++iDevice) { - ma_device_info deviceInfo; - ma_zero_object(&deviceInfo); - deviceInfo.id.opensl = pDeviceIDs[iDevice]; + ((MA_PFN_AAudioStreamBuilder_delete)pContext->aaudio.AAudioStreamBuilder_delete)(pBuilder); + return MA_SUCCESS; +} - SLAudioOutputDescriptor desc; - resultSL = (*deviceCaps)->QueryAudioOutputCapabilities(deviceCaps, deviceInfo.id.opensl, &desc); - if (resultSL == SL_RESULT_SUCCESS) { - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), (const char*)desc.pDeviceName, (size_t)-1); +static ma_result ma_close_stream__aaudio(ma_context* pContext, ma_AAudioStream* pStream) +{ + return ma_result_from_aaudio(((MA_PFN_AAudioStream_close)pContext->aaudio.AAudioStream_close)(pStream)); +} - ma_bool32 cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); - if (cbResult == MA_FALSE) { - isTerminated = MA_TRUE; - break; - } - } - } +static ma_bool32 ma_has_default_device__aaudio(ma_context* pContext, ma_device_type deviceType) +{ + /* The only way to know this is to try creating a stream. */ + ma_AAudioStream* pStream; + ma_result result = ma_open_stream__aaudio(pContext, deviceType, NULL, ma_share_mode_shared, NULL, NULL, &pStream); + if (result != MA_SUCCESS) { + return MA_FALSE; } - /* Capture */ - if (!isTerminated) { - resultSL = (*deviceCaps)->GetAvailableAudioInputs(deviceCaps, &deviceCount, pDeviceIDs); - if (resultSL != SL_RESULT_SUCCESS) { - return MA_NO_DEVICE; - } - - for (SLint32 iDevice = 0; iDevice < deviceCount; ++iDevice) { - ma_device_info deviceInfo; - ma_zero_object(&deviceInfo); - deviceInfo.id.opensl = pDeviceIDs[iDevice]; + ma_close_stream__aaudio(pContext, pStream); + return MA_TRUE; +} - SLAudioInputDescriptor desc; - resultSL = (*deviceCaps)->QueryAudioInputCapabilities(deviceCaps, deviceInfo.id.opensl, &desc); - if (resultSL == SL_RESULT_SUCCESS) { - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), (const char*)desc.deviceName, (size_t)-1); +static ma_result ma_wait_for_simple_state_transition__aaudio(ma_context* pContext, ma_AAudioStream* pStream, ma_aaudio_stream_state_t oldState, ma_aaudio_stream_state_t newState) +{ + ma_aaudio_stream_state_t actualNewState; + ma_aaudio_result_t resultAA = ((MA_PFN_AAudioStream_waitForStateChange)pContext->aaudio.AAudioStream_waitForStateChange)(pStream, oldState, &actualNewState, 5000000000); /* 5 second timeout. */ + if (resultAA != MA_AAUDIO_OK) { + return ma_result_from_aaudio(resultAA); + } - ma_bool32 cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); - if (cbResult == MA_FALSE) { - isTerminated = MA_TRUE; - break; - } - } - } + if (newState != actualNewState) { + return MA_ERROR; /* Failed to transition into the expected state. */ } return MA_SUCCESS; -#else - goto return_default_device; -#endif +} -return_default_device:; - cbResult = MA_TRUE; + +static ma_bool32 ma_context_is_device_id_equal__aaudio(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) +{ + MA_ASSERT(pContext != NULL); + MA_ASSERT(pID0 != NULL); + MA_ASSERT(pID1 != NULL); + (void)pContext; + + return pID0->aaudio == pID1->aaudio; +} + +static ma_result ma_context_enumerate_devices__aaudio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +{ + ma_bool32 cbResult = MA_TRUE; + + MA_ASSERT(pContext != NULL); + MA_ASSERT(callback != NULL); + + /* Unfortunately AAudio does not have an enumeration API. Therefore I'm only going to report default devices, but only if it can instantiate a stream. */ /* Playback. */ if (cbResult) { ma_device_info deviceInfo; - ma_zero_object(&deviceInfo); + MA_ZERO_OBJECT(&deviceInfo); + deviceInfo.id.aaudio = MA_AAUDIO_UNSPECIFIED; ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); - cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); + + if (ma_has_default_device__aaudio(pContext, ma_device_type_playback)) { + cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); + } } /* Capture. */ if (cbResult) { ma_device_info deviceInfo; - ma_zero_object(&deviceInfo); + MA_ZERO_OBJECT(&deviceInfo); + deviceInfo.id.aaudio = MA_AAUDIO_UNSPECIFIED; ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); - cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); + + if (ma_has_default_device__aaudio(pContext, ma_device_type_capture)) { + cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); + } } return MA_SUCCESS; } -ma_result ma_context_get_device_info__opensl(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) +static ma_result ma_context_get_device_info__aaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) { - ma_assert(pContext != NULL); + ma_AAudioStream* pStream; + ma_result result; - ma_assert(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to get device info. */ - if (g_maOpenSLInitCounter == 0) { - return MA_INVALID_OPERATION; - } + MA_ASSERT(pContext != NULL); - /* No exclusive mode with OpenSL|ES. */ + /* No exclusive mode with AAudio. */ if (shareMode == ma_share_mode_exclusive) { return MA_SHARE_MODE_NOT_SUPPORTED; } - /* - TODO: Test Me. - - This is currently untested, so for now we are just returning default devices. - */ -#if 0 && !defined(MA_ANDROID) - SLAudioIODeviceCapabilitiesItf deviceCaps; - SLresult resultSL = (*g_maEngineObjectSL)->GetInterface(g_maEngineObjectSL, SL_IID_AUDIOIODEVICECAPABILITIES, &deviceCaps); - if (resultSL != SL_RESULT_SUCCESS) { - /* The interface may not be supported so just report a default device. */ - goto return_default_device; - } - - if (deviceType == ma_device_type_playback) { - SLAudioOutputDescriptor desc; - resultSL = (*deviceCaps)->QueryAudioOutputCapabilities(deviceCaps, pDeviceID->opensl, &desc); - if (resultSL != SL_RESULT_SUCCESS) { - return MA_NO_DEVICE; - } - - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), (const char*)desc.pDeviceName, (size_t)-1); - } else { - SLAudioInputDescriptor desc; - resultSL = (*deviceCaps)->QueryAudioInputCapabilities(deviceCaps, pDeviceID->opensl, &desc); - if (resultSL != SL_RESULT_SUCCESS) { - return MA_NO_DEVICE; - } - - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), (const char*)desc.deviceName, (size_t)-1); - } - - goto return_detailed_info; -#else - goto return_default_device; -#endif - -return_default_device: + /* ID */ if (pDeviceID != NULL) { - if ((deviceType == ma_device_type_playback && pDeviceID->opensl != SL_DEFAULTDEVICEID_AUDIOOUTPUT) || - (deviceType == ma_device_type_capture && pDeviceID->opensl != SL_DEFAULTDEVICEID_AUDIOINPUT)) { - return MA_NO_DEVICE; /* Don't know the device. */ - } + pDeviceInfo->id.aaudio = pDeviceID->aaudio; + } else { + pDeviceInfo->id.aaudio = MA_AAUDIO_UNSPECIFIED; } - - /* Name / Description */ + + /* Name */ if (deviceType == ma_device_type_playback) { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); } else { ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); } - goto return_detailed_info; + /* We'll need to open the device to get accurate sample rate and channel count information. */ + result = ma_open_stream__aaudio(pContext, deviceType, pDeviceID, shareMode, NULL, NULL, &pStream); + if (result != MA_SUCCESS) { + return result; + } + + pDeviceInfo->minChannels = ((MA_PFN_AAudioStream_getChannelCount)pContext->aaudio.AAudioStream_getChannelCount)(pStream); + pDeviceInfo->maxChannels = pDeviceInfo->minChannels; + pDeviceInfo->minSampleRate = ((MA_PFN_AAudioStream_getSampleRate)pContext->aaudio.AAudioStream_getSampleRate)(pStream); + pDeviceInfo->maxSampleRate = pDeviceInfo->minSampleRate; -return_detailed_info: + ma_close_stream__aaudio(pContext, pStream); + pStream = NULL; - /* - For now we're just outputting a set of values that are supported by the API but not necessarily supported - by the device natively. Later on we should work on this so that it more closely reflects the device's - actual native format. - */ - pDeviceInfo->minChannels = 1; - pDeviceInfo->maxChannels = 2; - pDeviceInfo->minSampleRate = 8000; - pDeviceInfo->maxSampleRate = 48000; + + /* AAudio supports s16 and f32. */ pDeviceInfo->formatCount = 2; - pDeviceInfo->formats[0] = ma_format_u8; - pDeviceInfo->formats[1] = ma_format_s16; -#if defined(MA_ANDROID) && __ANDROID_API__ >= 21 - pDeviceInfo->formats[pDeviceInfo->formatCount] = ma_format_f32; - pDeviceInfo->formatCount += 1; -#endif + pDeviceInfo->formats[0] = ma_format_s16; + pDeviceInfo->formats[1] = ma_format_f32; return MA_SUCCESS; } -#ifdef MA_ANDROID -/*void ma_buffer_queue_callback_capture__opensl_android(SLAndroidSimpleBufferQueueItf pBufferQueue, SLuint32 eventFlags, const void* pBuffer, SLuint32 bufferSize, SLuint32 dataUsed, void* pContext)*/ -void ma_buffer_queue_callback_capture__opensl_android(SLAndroidSimpleBufferQueueItf pBufferQueue, void* pUserData) +static void ma_device_uninit__aaudio(ma_device* pDevice) { - ma_device* pDevice = (ma_device*)pUserData; - size_t periodSizeInBytes; - ma_uint8* pBuffer; - SLresult resultSL; - - ma_assert(pDevice != NULL); + MA_ASSERT(pDevice != NULL); - (void)pBufferQueue; + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture); + pDevice->aaudio.pStreamCapture = NULL; + } - /* - For now, don't do anything unless the buffer was fully processed. From what I can tell, it looks like - OpenSL|ES 1.1 improves on buffer queues to the point that we could much more intelligently handle this, - but unfortunately it looks like Android is only supporting OpenSL|ES 1.0.1 for now :( - */ + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); + pDevice->aaudio.pStreamPlayback = NULL; + } - /* Don't do anything if the device is not started. */ - if (pDevice->state != MA_STATE_STARTED) { - return; + if (pDevice->type == ma_device_type_duplex) { + ma_pcm_rb_uninit(&pDevice->aaudio.duplexRB); } +} - periodSizeInBytes = (pDevice->capture.internalBufferSizeInFrames / pDevice->capture.internalPeriods) * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - pBuffer = pDevice->opensl.pBufferCapture + (pDevice->opensl.currentBufferIndexCapture * periodSizeInBytes); +static ma_result ma_device_init__aaudio(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) +{ + ma_result result; - if (pDevice->type == ma_device_type_duplex) { - ma_device__handle_duplex_callback_capture(pDevice, (pDevice->capture.internalBufferSizeInFrames / pDevice->capture.internalPeriods), pBuffer, &pDevice->opensl.duplexRB); - } else { - ma_device__send_frames_to_client(pDevice, (pDevice->capture.internalBufferSizeInFrames / pDevice->capture.internalPeriods), pBuffer); + MA_ASSERT(pDevice != NULL); + + if (pConfig->deviceType == ma_device_type_loopback) { + return MA_DEVICE_TYPE_NOT_SUPPORTED; } - resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture, pBuffer, periodSizeInBytes); - if (resultSL != SL_RESULT_SUCCESS) { - return; + /* No exclusive mode with AAudio. */ + if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) || + ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) { + return MA_SHARE_MODE_NOT_SUPPORTED; } - pDevice->opensl.currentBufferIndexCapture = (pDevice->opensl.currentBufferIndexCapture + 1) % pDevice->capture.internalPeriods; -} + /* We first need to try opening the stream. */ + if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { + int32_t bufferCapacityInFrames; + int32_t framesPerDataCallback; -void ma_buffer_queue_callback_playback__opensl_android(SLAndroidSimpleBufferQueueItf pBufferQueue, void* pUserData) -{ - ma_device* pDevice = (ma_device*)pUserData; - size_t periodSizeInBytes; - ma_uint8* pBuffer; - SLresult resultSL; + result = ma_open_stream__aaudio(pContext, ma_device_type_capture, pConfig->capture.pDeviceID, pConfig->capture.shareMode, pConfig, pDevice, (ma_AAudioStream**)&pDevice->aaudio.pStreamCapture); + if (result != MA_SUCCESS) { + return result; /* Failed to open the AAudio stream. */ + } - ma_assert(pDevice != NULL); + pDevice->capture.internalFormat = (((MA_PFN_AAudioStream_getFormat)pContext->aaudio.AAudioStream_getFormat)((ma_AAudioStream*)pDevice->aaudio.pStreamCapture) == MA_AAUDIO_FORMAT_PCM_I16) ? ma_format_s16 : ma_format_f32; + pDevice->capture.internalChannels = ((MA_PFN_AAudioStream_getChannelCount)pContext->aaudio.AAudioStream_getChannelCount)((ma_AAudioStream*)pDevice->aaudio.pStreamCapture); + pDevice->capture.internalSampleRate = ((MA_PFN_AAudioStream_getSampleRate)pContext->aaudio.AAudioStream_getSampleRate)((ma_AAudioStream*)pDevice->aaudio.pStreamCapture); + ma_get_standard_channel_map(ma_standard_channel_map_default, pDevice->capture.internalChannels, pDevice->capture.internalChannelMap); /* <-- Cannot find info on channel order, so assuming a default. */ - (void)pBufferQueue; + bufferCapacityInFrames = ((MA_PFN_AAudioStream_getBufferCapacityInFrames)pContext->aaudio.AAudioStream_getBufferCapacityInFrames)((ma_AAudioStream*)pDevice->aaudio.pStreamCapture); + framesPerDataCallback = ((MA_PFN_AAudioStream_getFramesPerDataCallback)pContext->aaudio.AAudioStream_getFramesPerDataCallback)((ma_AAudioStream*)pDevice->aaudio.pStreamCapture); - /* Don't do anything if the device is not started. */ - if (pDevice->state != MA_STATE_STARTED) { - return; + if (framesPerDataCallback > 0) { + pDevice->capture.internalPeriodSizeInFrames = framesPerDataCallback; + pDevice->capture.internalPeriods = bufferCapacityInFrames / framesPerDataCallback; + } else { + pDevice->capture.internalPeriodSizeInFrames = bufferCapacityInFrames; + pDevice->capture.internalPeriods = 1; + } } - periodSizeInBytes = (pDevice->playback.internalBufferSizeInFrames / pDevice->playback.internalPeriods) * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - pBuffer = pDevice->opensl.pBufferPlayback + (pDevice->opensl.currentBufferIndexPlayback * periodSizeInBytes); + if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { + int32_t bufferCapacityInFrames; + int32_t framesPerDataCallback; - if (pDevice->type == ma_device_type_duplex) { - ma_device__handle_duplex_callback_playback(pDevice, (pDevice->playback.internalBufferSizeInFrames / pDevice->playback.internalPeriods), pBuffer, &pDevice->opensl.duplexRB); - } else { - ma_device__read_frames_from_client(pDevice, (pDevice->playback.internalBufferSizeInFrames / pDevice->playback.internalPeriods), pBuffer); + result = ma_open_stream__aaudio(pContext, ma_device_type_playback, pConfig->playback.pDeviceID, pConfig->playback.shareMode, pConfig, pDevice, (ma_AAudioStream**)&pDevice->aaudio.pStreamPlayback); + if (result != MA_SUCCESS) { + return result; /* Failed to open the AAudio stream. */ + } + + pDevice->playback.internalFormat = (((MA_PFN_AAudioStream_getFormat)pContext->aaudio.AAudioStream_getFormat)((ma_AAudioStream*)pDevice->aaudio.pStreamPlayback) == MA_AAUDIO_FORMAT_PCM_I16) ? ma_format_s16 : ma_format_f32; + pDevice->playback.internalChannels = ((MA_PFN_AAudioStream_getChannelCount)pContext->aaudio.AAudioStream_getChannelCount)((ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); + pDevice->playback.internalSampleRate = ((MA_PFN_AAudioStream_getSampleRate)pContext->aaudio.AAudioStream_getSampleRate)((ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); + ma_get_standard_channel_map(ma_standard_channel_map_default, pDevice->playback.internalChannels, pDevice->playback.internalChannelMap); /* <-- Cannot find info on channel order, so assuming a default. */ + + bufferCapacityInFrames = ((MA_PFN_AAudioStream_getBufferCapacityInFrames)pContext->aaudio.AAudioStream_getBufferCapacityInFrames)((ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); + framesPerDataCallback = ((MA_PFN_AAudioStream_getFramesPerDataCallback)pContext->aaudio.AAudioStream_getFramesPerDataCallback)((ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); + + if (framesPerDataCallback > 0) { + pDevice->playback.internalPeriodSizeInFrames = framesPerDataCallback; + pDevice->playback.internalPeriods = bufferCapacityInFrames / framesPerDataCallback; + } else { + pDevice->playback.internalPeriodSizeInFrames = bufferCapacityInFrames; + pDevice->playback.internalPeriods = 1; + } } - resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback, pBuffer, periodSizeInBytes); - if (resultSL != SL_RESULT_SUCCESS) { - return; + if (pConfig->deviceType == ma_device_type_duplex) { + ma_uint32 rbSizeInFrames = (ma_uint32)ma_calculate_frame_count_after_resampling(pDevice->sampleRate, pDevice->capture.internalSampleRate, pDevice->capture.internalPeriodSizeInFrames) * pDevice->capture.internalPeriods; + ma_result result = ma_pcm_rb_init(pDevice->capture.format, pDevice->capture.channels, rbSizeInFrames, NULL, &pDevice->pContext->allocationCallbacks, &pDevice->aaudio.duplexRB); + if (result != MA_SUCCESS) { + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture); + } + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); + } + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[AAudio] Failed to initialize ring buffer.", result); + } + + /* We need a period to act as a buffer for cases where the playback and capture device's end up desyncing. */ + { + ma_uint32 marginSizeInFrames = rbSizeInFrames / pDevice->capture.internalPeriods; + void* pMarginData; + ma_pcm_rb_acquire_write(&pDevice->aaudio.duplexRB, &marginSizeInFrames, &pMarginData); + { + MA_ZERO_MEMORY(pMarginData, marginSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels)); + } + ma_pcm_rb_commit_write(&pDevice->aaudio.duplexRB, marginSizeInFrames, pMarginData); + } } - pDevice->opensl.currentBufferIndexPlayback = (pDevice->opensl.currentBufferIndexPlayback + 1) % pDevice->playback.internalPeriods; + return MA_SUCCESS; } -#endif -void ma_device_uninit__opensl(ma_device* pDevice) +static ma_result ma_device_start_stream__aaudio(ma_device* pDevice, ma_AAudioStream* pStream) { - ma_assert(pDevice != NULL); + ma_aaudio_result_t resultAA; + ma_aaudio_stream_state_t currentState; - ma_assert(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it before uninitializing the device. */ - if (g_maOpenSLInitCounter == 0) { - return; + MA_ASSERT(pDevice != NULL); + + resultAA = ((MA_PFN_AAudioStream_requestStart)pDevice->pContext->aaudio.AAudioStream_requestStart)(pStream); + if (resultAA != MA_AAUDIO_OK) { + return ma_result_from_aaudio(resultAA); } - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - if (pDevice->opensl.pAudioRecorderObj) { - MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->Destroy((SLObjectItf)pDevice->opensl.pAudioRecorderObj); - } + /* Do we actually need to wait for the device to transition into it's started state? */ - ma_free(pDevice->opensl.pBufferCapture); - } + /* The device should be in either a starting or started state. If it's not set to started we need to wait for it to transition. It should go from starting to started. */ + currentState = ((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream); + if (currentState != MA_AAUDIO_STREAM_STATE_STARTED) { + ma_result result; - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - if (pDevice->opensl.pAudioPlayerObj) { - MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->Destroy((SLObjectItf)pDevice->opensl.pAudioPlayerObj); - } - if (pDevice->opensl.pOutputMixObj) { - MA_OPENSL_OBJ(pDevice->opensl.pOutputMixObj)->Destroy((SLObjectItf)pDevice->opensl.pOutputMixObj); + if (currentState != MA_AAUDIO_STREAM_STATE_STARTING) { + return MA_ERROR; /* Expecting the stream to be a starting or started state. */ } - ma_free(pDevice->opensl.pBufferPlayback); + result = ma_wait_for_simple_state_transition__aaudio(pDevice->pContext, pStream, currentState, MA_AAUDIO_STREAM_STATE_STARTED); + if (result != MA_SUCCESS) { + return result; + } } - if (pDevice->type == ma_device_type_duplex) { - ma_pcm_rb_uninit(&pDevice->opensl.duplexRB); - } + return MA_SUCCESS; } -#if defined(MA_ANDROID) && __ANDROID_API__ >= 21 -typedef SLAndroidDataFormat_PCM_EX ma_SLDataFormat_PCM; -#else -typedef SLDataFormat_PCM ma_SLDataFormat_PCM; -#endif - -ma_result ma_SLDataFormat_PCM_init__opensl(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, const ma_channel* channelMap, ma_SLDataFormat_PCM* pDataFormat) +static ma_result ma_device_stop_stream__aaudio(ma_device* pDevice, ma_AAudioStream* pStream) { -#if defined(MA_ANDROID) && __ANDROID_API__ >= 21 - if (format == ma_format_f32) { - pDataFormat->formatType = SL_ANDROID_DATAFORMAT_PCM_EX; - pDataFormat->representation = SL_ANDROID_PCM_REPRESENTATION_FLOAT; - } else { - pDataFormat->formatType = SL_DATAFORMAT_PCM; - } -#else - pDataFormat->formatType = SL_DATAFORMAT_PCM; -#endif + ma_aaudio_result_t resultAA; + ma_aaudio_stream_state_t currentState; - pDataFormat->numChannels = channels; - ((SLDataFormat_PCM*)pDataFormat)->samplesPerSec = ma_round_to_standard_sample_rate__opensl(sampleRate * 1000); /* In millihertz. Annoyingly, the sample rate variable is named differently between SLAndroidDataFormat_PCM_EX and SLDataFormat_PCM */ - pDataFormat->bitsPerSample = ma_get_bytes_per_sample(format)*8; - pDataFormat->channelMask = ma_channel_map_to_channel_mask__opensl(channelMap, channels); - pDataFormat->endianness = (ma_is_little_endian()) ? SL_BYTEORDER_LITTLEENDIAN : SL_BYTEORDER_BIGENDIAN; + MA_ASSERT(pDevice != NULL); /* - Android has a few restrictions on the format as documented here: https://developer.android.com/ndk/guides/audio/opensl-for-android.html - - Only mono and stereo is supported. - - Only u8 and s16 formats are supported. - - Maximum sample rate of 48000. + From the AAudio documentation: + + The stream will stop after all of the data currently buffered has been played. + + This maps with miniaudio's requirement that device's be drained which means we don't need to implement any draining logic. */ -#ifdef MA_ANDROID - if (pDataFormat->numChannels > 2) { - pDataFormat->numChannels = 2; + + resultAA = ((MA_PFN_AAudioStream_requestStop)pDevice->pContext->aaudio.AAudioStream_requestStop)(pStream); + if (resultAA != MA_AAUDIO_OK) { + return ma_result_from_aaudio(resultAA); } -#if __ANDROID_API__ >= 21 - if (pDataFormat->formatType == SL_ANDROID_DATAFORMAT_PCM_EX) { - /* It's floating point. */ - ma_assert(pDataFormat->representation == SL_ANDROID_PCM_REPRESENTATION_FLOAT); - if (pDataFormat->bitsPerSample > 32) { - pDataFormat->bitsPerSample = 32; + + /* The device should be in either a stopping or stopped state. If it's not set to started we need to wait for it to transition. It should go from stopping to stopped. */ + currentState = ((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream); + if (currentState != MA_AAUDIO_STREAM_STATE_STOPPED) { + ma_result result; + + if (currentState != MA_AAUDIO_STREAM_STATE_STOPPING) { + return MA_ERROR; /* Expecting the stream to be a stopping or stopped state. */ } - } else { - if (pDataFormat->bitsPerSample > 16) { - pDataFormat->bitsPerSample = 16; + + result = ma_wait_for_simple_state_transition__aaudio(pDevice->pContext, pStream, currentState, MA_AAUDIO_STREAM_STATE_STOPPED); + if (result != MA_SUCCESS) { + return result; } } -#else - if (pDataFormat->bitsPerSample > 16) { - pDataFormat->bitsPerSample = 16; - } -#endif - if (((SLDataFormat_PCM*)pDataFormat)->samplesPerSec > SL_SAMPLINGRATE_48) { - ((SLDataFormat_PCM*)pDataFormat)->samplesPerSec = SL_SAMPLINGRATE_48; - } -#endif - - pDataFormat->containerSize = pDataFormat->bitsPerSample; /* Always tightly packed for now. */ return MA_SUCCESS; } -ma_result ma_deconstruct_SLDataFormat_PCM__opensl(ma_SLDataFormat_PCM* pDataFormat, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap) +static ma_result ma_device_start__aaudio(ma_device* pDevice) { - ma_bool32 isFloatingPoint = MA_FALSE; -#if defined(MA_ANDROID) && __ANDROID_API__ >= 21 - if (pDataFormat->formatType == SL_ANDROID_DATAFORMAT_PCM_EX) { - ma_assert(pDataFormat->representation == SL_ANDROID_PCM_REPRESENTATION_FLOAT); - isFloatingPoint = MA_TRUE; - } -#endif - if (isFloatingPoint) { - if (pDataFormat->bitsPerSample == 32) { - *pFormat = ma_format_f32; - } - } else { - if (pDataFormat->bitsPerSample == 8) { - *pFormat = ma_format_u8; - } else if (pDataFormat->bitsPerSample == 16) { - *pFormat = ma_format_s16; - } else if (pDataFormat->bitsPerSample == 24) { - *pFormat = ma_format_s24; - } else if (pDataFormat->bitsPerSample == 32) { - *pFormat = ma_format_s32; + MA_ASSERT(pDevice != NULL); + + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + ma_result result = ma_device_start_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture); + if (result != MA_SUCCESS) { + return result; } } - *pChannels = pDataFormat->numChannels; - *pSampleRate = ((SLDataFormat_PCM*)pDataFormat)->samplesPerSec / 1000; - ma_channel_mask_to_channel_map__opensl(pDataFormat->channelMask, pDataFormat->numChannels, pChannelMap); + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + ma_result result = ma_device_start_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); + if (result != MA_SUCCESS) { + if (pDevice->type == ma_device_type_duplex) { + ma_device_stop_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture); + } + return result; + } + } return MA_SUCCESS; } -ma_result ma_device_init__opensl(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) +static ma_result ma_device_stop__aaudio(ma_device* pDevice) { -#ifdef MA_ANDROID - SLDataLocator_AndroidSimpleBufferQueue queue; - SLresult resultSL; - ma_uint32 bufferSizeInFrames; - size_t bufferSizeInBytes; - const SLInterfaceID itfIDs1[] = {SL_IID_ANDROIDSIMPLEBUFFERQUEUE}; - const SLboolean itfIDsRequired1[] = {SL_BOOLEAN_TRUE}; -#endif + ma_stop_proc onStop; - (void)pContext; + MA_ASSERT(pDevice != NULL); - ma_assert(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to initialize a new device. */ - if (g_maOpenSLInitCounter == 0) { - return MA_INVALID_OPERATION; + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + ma_result result = ma_device_stop_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture); + if (result != MA_SUCCESS) { + return result; + } } - if (pConfig->deviceType == ma_device_type_loopback) { - return MA_DEVICE_TYPE_NOT_SUPPORTED; + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + ma_result result = ma_device_stop_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback); + if (result != MA_SUCCESS) { + return result; + } } - /* - For now, only supporting Android implementations of OpenSL|ES since that's the only one I've - been able to test with and I currently depend on Android-specific extensions (simple buffer - queues). - */ -#ifdef MA_ANDROID - /* No exclusive mode with OpenSL|ES. */ - if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) || - ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) { - return MA_SHARE_MODE_NOT_SUPPORTED; + onStop = pDevice->onStop; + if (onStop) { + onStop(pDevice); } - /* Now we can start initializing the device properly. */ - ma_assert(pDevice != NULL); - ma_zero_object(&pDevice->opensl); - - queue.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE; - queue.numBuffers = pConfig->periods; - - - if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { - ma_SLDataFormat_PCM pcm; - SLDataLocator_IODevice locatorDevice; - SLDataSource source; - SLDataSink sink; - - ma_SLDataFormat_PCM_init__opensl(pConfig->capture.format, pConfig->capture.channels, pConfig->sampleRate, pConfig->capture.channelMap, &pcm); - - locatorDevice.locatorType = SL_DATALOCATOR_IODEVICE; - locatorDevice.deviceType = SL_IODEVICE_AUDIOINPUT; - locatorDevice.deviceID = (pConfig->capture.pDeviceID == NULL) ? SL_DEFAULTDEVICEID_AUDIOINPUT : pConfig->capture.pDeviceID->opensl; - locatorDevice.device = NULL; - - source.pLocator = &locatorDevice; - source.pFormat = NULL; - - sink.pLocator = &queue; - sink.pFormat = (SLDataFormat_PCM*)&pcm; - - resultSL = (*g_maEngineSL)->CreateAudioRecorder(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioRecorderObj, &source, &sink, 1, itfIDs1, itfIDsRequired1); - if (resultSL == SL_RESULT_CONTENT_UNSUPPORTED) { - /* Unsupported format. Fall back to something safer and try again. If this fails, just abort. */ - pcm.formatType = SL_DATAFORMAT_PCM; - pcm.numChannels = 1; - ((SLDataFormat_PCM*)&pcm)->samplesPerSec = SL_SAMPLINGRATE_16; /* The name of the sample rate variable is different between SLAndroidDataFormat_PCM_EX and SLDataFormat_PCM. */ - pcm.bitsPerSample = 16; - pcm.containerSize = pcm.bitsPerSample; /* Always tightly packed for now. */ - pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT; - resultSL = (*g_maEngineSL)->CreateAudioRecorder(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioRecorderObj, &source, &sink, 1, itfIDs1, itfIDsRequired1); - } - - if (resultSL != SL_RESULT_SUCCESS) { - ma_device_uninit__opensl(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to create audio recorder.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } - - if (MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->Realize((SLObjectItf)pDevice->opensl.pAudioRecorderObj, SL_BOOLEAN_FALSE) != SL_RESULT_SUCCESS) { - ma_device_uninit__opensl(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to realize audio recorder.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } - - if (MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioRecorderObj, SL_IID_RECORD, &pDevice->opensl.pAudioRecorder) != SL_RESULT_SUCCESS) { - ma_device_uninit__opensl(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_RECORD interface.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } + return MA_SUCCESS; +} - if (MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioRecorderObj, SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &pDevice->opensl.pBufferQueueCapture) != SL_RESULT_SUCCESS) { - ma_device_uninit__opensl(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_ANDROIDSIMPLEBUFFERQUEUE interface.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } - if (MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->RegisterCallback((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture, ma_buffer_queue_callback_capture__opensl_android, pDevice) != SL_RESULT_SUCCESS) { - ma_device_uninit__opensl(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to register buffer queue callback.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } +static ma_result ma_context_uninit__aaudio(ma_context* pContext) +{ + MA_ASSERT(pContext != NULL); + MA_ASSERT(pContext->backend == ma_backend_aaudio); + + ma_dlclose(pContext, pContext->aaudio.hAAudio); + pContext->aaudio.hAAudio = NULL; - /* The internal format is determined by the "pcm" object. */ - ma_deconstruct_SLDataFormat_PCM__opensl(&pcm, &pDevice->capture.internalFormat, &pDevice->capture.internalChannels, &pDevice->capture.internalSampleRate, pDevice->capture.internalChannelMap); + return MA_SUCCESS; +} - /* Buffer. */ - bufferSizeInFrames = pConfig->bufferSizeInFrames; - if (bufferSizeInFrames == 0) { - bufferSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->bufferSizeInMilliseconds, pDevice->capture.internalSampleRate); - } - pDevice->capture.internalPeriods = pConfig->periods; - pDevice->capture.internalBufferSizeInFrames = (bufferSizeInFrames / pDevice->capture.internalPeriods) * pDevice->capture.internalPeriods; - pDevice->opensl.currentBufferIndexCapture = 0; +static ma_result ma_context_init__aaudio(const ma_context_config* pConfig, ma_context* pContext) +{ + const char* libNames[] = { + "libaaudio.so" + }; + size_t i; - bufferSizeInBytes = pDevice->capture.internalBufferSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - pDevice->opensl.pBufferCapture = (ma_uint8*)ma_malloc(bufferSizeInBytes); - if (pDevice->opensl.pBufferCapture == NULL) { - ma_device_uninit__opensl(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to allocate memory for data buffer.", MA_OUT_OF_MEMORY); + for (i = 0; i < ma_countof(libNames); ++i) { + pContext->aaudio.hAAudio = ma_dlopen(pContext, libNames[i]); + if (pContext->aaudio.hAAudio != NULL) { + break; } - MA_ZERO_MEMORY(pDevice->opensl.pBufferCapture, bufferSizeInBytes); } - if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { - ma_SLDataFormat_PCM pcm; - SLDataSource source; - SLDataLocator_OutputMix outmixLocator; - SLDataSink sink; + if (pContext->aaudio.hAAudio == NULL) { + return MA_FAILED_TO_INIT_BACKEND; + } - ma_SLDataFormat_PCM_init__opensl(pConfig->playback.format, pConfig->playback.channels, pConfig->sampleRate, pConfig->playback.channelMap, &pcm); + pContext->aaudio.AAudio_createStreamBuilder = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudio_createStreamBuilder"); + pContext->aaudio.AAudioStreamBuilder_delete = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_delete"); + pContext->aaudio.AAudioStreamBuilder_setDeviceId = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setDeviceId"); + pContext->aaudio.AAudioStreamBuilder_setDirection = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setDirection"); + pContext->aaudio.AAudioStreamBuilder_setSharingMode = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setSharingMode"); + pContext->aaudio.AAudioStreamBuilder_setFormat = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setFormat"); + pContext->aaudio.AAudioStreamBuilder_setChannelCount = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setChannelCount"); + pContext->aaudio.AAudioStreamBuilder_setSampleRate = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setSampleRate"); + pContext->aaudio.AAudioStreamBuilder_setBufferCapacityInFrames = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setBufferCapacityInFrames"); + pContext->aaudio.AAudioStreamBuilder_setFramesPerDataCallback = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setFramesPerDataCallback"); + pContext->aaudio.AAudioStreamBuilder_setDataCallback = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setDataCallback"); + pContext->aaudio.AAudioStreamBuilder_setErrorCallback = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setErrorCallback"); + pContext->aaudio.AAudioStreamBuilder_setPerformanceMode = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setPerformanceMode"); + pContext->aaudio.AAudioStreamBuilder_openStream = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_openStream"); + pContext->aaudio.AAudioStream_close = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_close"); + pContext->aaudio.AAudioStream_getState = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getState"); + pContext->aaudio.AAudioStream_waitForStateChange = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_waitForStateChange"); + pContext->aaudio.AAudioStream_getFormat = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getFormat"); + pContext->aaudio.AAudioStream_getChannelCount = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getChannelCount"); + pContext->aaudio.AAudioStream_getSampleRate = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getSampleRate"); + pContext->aaudio.AAudioStream_getBufferCapacityInFrames = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getBufferCapacityInFrames"); + pContext->aaudio.AAudioStream_getFramesPerDataCallback = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getFramesPerDataCallback"); + pContext->aaudio.AAudioStream_getFramesPerBurst = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getFramesPerBurst"); + pContext->aaudio.AAudioStream_requestStart = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_requestStart"); + pContext->aaudio.AAudioStream_requestStop = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_requestStop"); - resultSL = (*g_maEngineSL)->CreateOutputMix(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pOutputMixObj, 0, NULL, NULL); - if (resultSL != SL_RESULT_SUCCESS) { - ma_device_uninit__opensl(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to create output mix.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } + pContext->isBackendAsynchronous = MA_TRUE; - if (MA_OPENSL_OBJ(pDevice->opensl.pOutputMixObj)->Realize((SLObjectItf)pDevice->opensl.pOutputMixObj, SL_BOOLEAN_FALSE)) { - ma_device_uninit__opensl(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to realize output mix object.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } + pContext->onUninit = ma_context_uninit__aaudio; + pContext->onDeviceIDEqual = ma_context_is_device_id_equal__aaudio; + pContext->onEnumDevices = ma_context_enumerate_devices__aaudio; + pContext->onGetDeviceInfo = ma_context_get_device_info__aaudio; + pContext->onDeviceInit = ma_device_init__aaudio; + pContext->onDeviceUninit = ma_device_uninit__aaudio; + pContext->onDeviceStart = ma_device_start__aaudio; + pContext->onDeviceStop = ma_device_stop__aaudio; - if (MA_OPENSL_OBJ(pDevice->opensl.pOutputMixObj)->GetInterface((SLObjectItf)pDevice->opensl.pOutputMixObj, SL_IID_OUTPUTMIX, &pDevice->opensl.pOutputMix) != SL_RESULT_SUCCESS) { - ma_device_uninit__opensl(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_OUTPUTMIX interface.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } + (void)pConfig; + return MA_SUCCESS; +} +#endif /* AAudio */ - /* Set the output device. */ - if (pConfig->playback.pDeviceID != NULL) { - SLuint32 deviceID_OpenSL = pConfig->playback.pDeviceID->opensl; - MA_OPENSL_OUTPUTMIX(pDevice->opensl.pOutputMix)->ReRoute((SLOutputMixItf)pDevice->opensl.pOutputMix, 1, &deviceID_OpenSL); - } - - source.pLocator = &queue; - source.pFormat = (SLDataFormat_PCM*)&pcm; - outmixLocator.locatorType = SL_DATALOCATOR_OUTPUTMIX; - outmixLocator.outputMix = (SLObjectItf)pDevice->opensl.pOutputMixObj; +/****************************************************************************** - sink.pLocator = &outmixLocator; - sink.pFormat = NULL; +OpenSL|ES Backend - resultSL = (*g_maEngineSL)->CreateAudioPlayer(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioPlayerObj, &source, &sink, 1, itfIDs1, itfIDsRequired1); - if (resultSL == SL_RESULT_CONTENT_UNSUPPORTED) { - /* Unsupported format. Fall back to something safer and try again. If this fails, just abort. */ - pcm.formatType = SL_DATAFORMAT_PCM; - pcm.numChannels = 2; - ((SLDataFormat_PCM*)&pcm)->samplesPerSec = SL_SAMPLINGRATE_16; - pcm.bitsPerSample = 16; - pcm.containerSize = pcm.bitsPerSample; /* Always tightly packed for now. */ - pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT; - resultSL = (*g_maEngineSL)->CreateAudioPlayer(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioPlayerObj, &source, &sink, 1, itfIDs1, itfIDsRequired1); - } +******************************************************************************/ +#ifdef MA_HAS_OPENSL +#include +#ifdef MA_ANDROID +#include +#endif - if (resultSL != SL_RESULT_SUCCESS) { - ma_device_uninit__opensl(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to create audio player.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } +/* OpenSL|ES has one-per-application objects :( */ +SLObjectItf g_maEngineObjectSL = NULL; +SLEngineItf g_maEngineSL = NULL; +ma_uint32 g_maOpenSLInitCounter = 0; - if (MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->Realize((SLObjectItf)pDevice->opensl.pAudioPlayerObj, SL_BOOLEAN_FALSE) != SL_RESULT_SUCCESS) { - ma_device_uninit__opensl(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to realize audio player.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } +#define MA_OPENSL_OBJ(p) (*((SLObjectItf)(p))) +#define MA_OPENSL_OUTPUTMIX(p) (*((SLOutputMixItf)(p))) +#define MA_OPENSL_PLAY(p) (*((SLPlayItf)(p))) +#define MA_OPENSL_RECORD(p) (*((SLRecordItf)(p))) - if (MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioPlayerObj, SL_IID_PLAY, &pDevice->opensl.pAudioPlayer) != SL_RESULT_SUCCESS) { - ma_device_uninit__opensl(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_PLAY interface.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } +#ifdef MA_ANDROID +#define MA_OPENSL_BUFFERQUEUE(p) (*((SLAndroidSimpleBufferQueueItf)(p))) +#else +#define MA_OPENSL_BUFFERQUEUE(p) (*((SLBufferQueueItf)(p))) +#endif - if (MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioPlayerObj, SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &pDevice->opensl.pBufferQueuePlayback) != SL_RESULT_SUCCESS) { - ma_device_uninit__opensl(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_ANDROIDSIMPLEBUFFERQUEUE interface.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } +/* Converts an individual OpenSL-style channel identifier (SL_SPEAKER_FRONT_LEFT, etc.) to miniaudio. */ +static ma_uint8 ma_channel_id_to_ma__opensl(SLuint32 id) +{ + switch (id) + { + case SL_SPEAKER_FRONT_LEFT: return MA_CHANNEL_FRONT_LEFT; + case SL_SPEAKER_FRONT_RIGHT: return MA_CHANNEL_FRONT_RIGHT; + case SL_SPEAKER_FRONT_CENTER: return MA_CHANNEL_FRONT_CENTER; + case SL_SPEAKER_LOW_FREQUENCY: return MA_CHANNEL_LFE; + case SL_SPEAKER_BACK_LEFT: return MA_CHANNEL_BACK_LEFT; + case SL_SPEAKER_BACK_RIGHT: return MA_CHANNEL_BACK_RIGHT; + case SL_SPEAKER_FRONT_LEFT_OF_CENTER: return MA_CHANNEL_FRONT_LEFT_CENTER; + case SL_SPEAKER_FRONT_RIGHT_OF_CENTER: return MA_CHANNEL_FRONT_RIGHT_CENTER; + case SL_SPEAKER_BACK_CENTER: return MA_CHANNEL_BACK_CENTER; + case SL_SPEAKER_SIDE_LEFT: return MA_CHANNEL_SIDE_LEFT; + case SL_SPEAKER_SIDE_RIGHT: return MA_CHANNEL_SIDE_RIGHT; + case SL_SPEAKER_TOP_CENTER: return MA_CHANNEL_TOP_CENTER; + case SL_SPEAKER_TOP_FRONT_LEFT: return MA_CHANNEL_TOP_FRONT_LEFT; + case SL_SPEAKER_TOP_FRONT_CENTER: return MA_CHANNEL_TOP_FRONT_CENTER; + case SL_SPEAKER_TOP_FRONT_RIGHT: return MA_CHANNEL_TOP_FRONT_RIGHT; + case SL_SPEAKER_TOP_BACK_LEFT: return MA_CHANNEL_TOP_BACK_LEFT; + case SL_SPEAKER_TOP_BACK_CENTER: return MA_CHANNEL_TOP_BACK_CENTER; + case SL_SPEAKER_TOP_BACK_RIGHT: return MA_CHANNEL_TOP_BACK_RIGHT; + default: return 0; + } +} - if (MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->RegisterCallback((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback, ma_buffer_queue_callback_playback__opensl_android, pDevice) != SL_RESULT_SUCCESS) { - ma_device_uninit__opensl(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to register buffer queue callback.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); - } +/* Converts an individual miniaudio channel identifier (MA_CHANNEL_FRONT_LEFT, etc.) to OpenSL-style. */ +static SLuint32 ma_channel_id_to_opensl(ma_uint8 id) +{ + switch (id) + { + case MA_CHANNEL_MONO: return SL_SPEAKER_FRONT_CENTER; + case MA_CHANNEL_FRONT_LEFT: return SL_SPEAKER_FRONT_LEFT; + case MA_CHANNEL_FRONT_RIGHT: return SL_SPEAKER_FRONT_RIGHT; + case MA_CHANNEL_FRONT_CENTER: return SL_SPEAKER_FRONT_CENTER; + case MA_CHANNEL_LFE: return SL_SPEAKER_LOW_FREQUENCY; + case MA_CHANNEL_BACK_LEFT: return SL_SPEAKER_BACK_LEFT; + case MA_CHANNEL_BACK_RIGHT: return SL_SPEAKER_BACK_RIGHT; + case MA_CHANNEL_FRONT_LEFT_CENTER: return SL_SPEAKER_FRONT_LEFT_OF_CENTER; + case MA_CHANNEL_FRONT_RIGHT_CENTER: return SL_SPEAKER_FRONT_RIGHT_OF_CENTER; + case MA_CHANNEL_BACK_CENTER: return SL_SPEAKER_BACK_CENTER; + case MA_CHANNEL_SIDE_LEFT: return SL_SPEAKER_SIDE_LEFT; + case MA_CHANNEL_SIDE_RIGHT: return SL_SPEAKER_SIDE_RIGHT; + case MA_CHANNEL_TOP_CENTER: return SL_SPEAKER_TOP_CENTER; + case MA_CHANNEL_TOP_FRONT_LEFT: return SL_SPEAKER_TOP_FRONT_LEFT; + case MA_CHANNEL_TOP_FRONT_CENTER: return SL_SPEAKER_TOP_FRONT_CENTER; + case MA_CHANNEL_TOP_FRONT_RIGHT: return SL_SPEAKER_TOP_FRONT_RIGHT; + case MA_CHANNEL_TOP_BACK_LEFT: return SL_SPEAKER_TOP_BACK_LEFT; + case MA_CHANNEL_TOP_BACK_CENTER: return SL_SPEAKER_TOP_BACK_CENTER; + case MA_CHANNEL_TOP_BACK_RIGHT: return SL_SPEAKER_TOP_BACK_RIGHT; + default: return 0; + } +} - /* The internal format is determined by the "pcm" object. */ - ma_deconstruct_SLDataFormat_PCM__opensl(&pcm, &pDevice->playback.internalFormat, &pDevice->playback.internalChannels, &pDevice->playback.internalSampleRate, pDevice->playback.internalChannelMap); +/* Converts a channel mapping to an OpenSL-style channel mask. */ +static SLuint32 ma_channel_map_to_channel_mask__opensl(const ma_channel channelMap[MA_MAX_CHANNELS], ma_uint32 channels) +{ + SLuint32 channelMask = 0; + ma_uint32 iChannel; + for (iChannel = 0; iChannel < channels; ++iChannel) { + channelMask |= ma_channel_id_to_opensl(channelMap[iChannel]); + } - /* Buffer. */ - bufferSizeInFrames = pConfig->bufferSizeInFrames; - if (bufferSizeInFrames == 0) { - bufferSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->bufferSizeInMilliseconds, pDevice->playback.internalSampleRate); - } - pDevice->playback.internalPeriods = pConfig->periods; - pDevice->playback.internalBufferSizeInFrames = (bufferSizeInFrames / pDevice->playback.internalPeriods) * pDevice->playback.internalPeriods; - pDevice->opensl.currentBufferIndexPlayback = 0; + return channelMask; +} - bufferSizeInBytes = pDevice->playback.internalBufferSizeInFrames * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - pDevice->opensl.pBufferPlayback = (ma_uint8*)ma_malloc(bufferSizeInBytes); - if (pDevice->opensl.pBufferPlayback == NULL) { - ma_device_uninit__opensl(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to allocate memory for data buffer.", MA_OUT_OF_MEMORY); +/* Converts an OpenSL-style channel mask to a miniaudio channel map. */ +static void ma_channel_mask_to_channel_map__opensl(SLuint32 channelMask, ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) +{ + if (channels == 1 && channelMask == 0) { + channelMap[0] = MA_CHANNEL_MONO; + } else if (channels == 2 && channelMask == 0) { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + } else { + if (channels == 1 && (channelMask & SL_SPEAKER_FRONT_CENTER) != 0) { + channelMap[0] = MA_CHANNEL_MONO; + } else { + /* Just iterate over each bit. */ + ma_uint32 iChannel = 0; + ma_uint32 iBit; + for (iBit = 0; iBit < 32; ++iBit) { + SLuint32 bitValue = (channelMask & (1UL << iBit)); + if (bitValue != 0) { + /* The bit is set. */ + channelMap[iChannel] = ma_channel_id_to_ma__opensl(bitValue); + iChannel += 1; + } + } } - MA_ZERO_MEMORY(pDevice->opensl.pBufferPlayback, bufferSizeInBytes); } +} + +static SLuint32 ma_round_to_standard_sample_rate__opensl(SLuint32 samplesPerSec) +{ + if (samplesPerSec <= SL_SAMPLINGRATE_8) { + return SL_SAMPLINGRATE_8; + } + if (samplesPerSec <= SL_SAMPLINGRATE_11_025) { + return SL_SAMPLINGRATE_11_025; + } + if (samplesPerSec <= SL_SAMPLINGRATE_12) { + return SL_SAMPLINGRATE_12; + } + if (samplesPerSec <= SL_SAMPLINGRATE_16) { + return SL_SAMPLINGRATE_16; + } + if (samplesPerSec <= SL_SAMPLINGRATE_22_05) { + return SL_SAMPLINGRATE_22_05; + } + if (samplesPerSec <= SL_SAMPLINGRATE_24) { + return SL_SAMPLINGRATE_24; + } + if (samplesPerSec <= SL_SAMPLINGRATE_32) { + return SL_SAMPLINGRATE_32; + } + if (samplesPerSec <= SL_SAMPLINGRATE_44_1) { + return SL_SAMPLINGRATE_44_1; + } + if (samplesPerSec <= SL_SAMPLINGRATE_48) { + return SL_SAMPLINGRATE_48; + } + + /* Android doesn't support more than 48000. */ +#ifndef MA_ANDROID + if (samplesPerSec <= SL_SAMPLINGRATE_64) { + return SL_SAMPLINGRATE_64; + } + if (samplesPerSec <= SL_SAMPLINGRATE_88_2) { + return SL_SAMPLINGRATE_88_2; + } + if (samplesPerSec <= SL_SAMPLINGRATE_96) { + return SL_SAMPLINGRATE_96; + } + if (samplesPerSec <= SL_SAMPLINGRATE_192) { + return SL_SAMPLINGRATE_192; + } +#endif - if (pConfig->deviceType == ma_device_type_duplex) { - ma_uint32 rbSizeInFrames = (ma_uint32)ma_calculate_frame_count_after_src(pDevice->sampleRate, pDevice->capture.internalSampleRate, pDevice->capture.internalBufferSizeInFrames); - ma_result result = ma_pcm_rb_init(pDevice->capture.format, pDevice->capture.channels, rbSizeInFrames, NULL, &pDevice->opensl.duplexRB); - if (result != MA_SUCCESS) { - ma_device_uninit__opensl(pDevice); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to initialize ring buffer.", result); - } + return SL_SAMPLINGRATE_16; +} - /* We need a period to act as a buffer for cases where the playback and capture device's end up desyncing. */ - { - ma_uint32 marginSizeInFrames = rbSizeInFrames / pDevice->capture.internalPeriods; - void* pMarginData; - ma_pcm_rb_acquire_write(&pDevice->opensl.duplexRB, &marginSizeInFrames, &pMarginData); - { - ma_zero_memory(pMarginData, marginSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels)); - } - ma_pcm_rb_commit_write(&pDevice->opensl.duplexRB, marginSizeInFrames, pMarginData); - } - } - return MA_SUCCESS; -#else - return MA_NO_BACKEND; /* Non-Android implementations are not supported. */ -#endif +static ma_bool32 ma_context_is_device_id_equal__opensl(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) +{ + MA_ASSERT(pContext != NULL); + MA_ASSERT(pID0 != NULL); + MA_ASSERT(pID1 != NULL); + (void)pContext; + + return pID0->opensl == pID1->opensl; } -ma_result ma_device_start__opensl(ma_device* pDevice) +static ma_result ma_context_enumerate_devices__opensl(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { - SLresult resultSL; - size_t periodSizeInBytes; - ma_uint32 iPeriod; + ma_bool32 cbResult; - ma_assert(pDevice != NULL); + MA_ASSERT(pContext != NULL); + MA_ASSERT(callback != NULL); - ma_assert(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to start the device. */ + MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to enumerate devices. */ if (g_maOpenSLInitCounter == 0) { return MA_INVALID_OPERATION; } - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - resultSL = MA_OPENSL_RECORD(pDevice->opensl.pAudioRecorder)->SetRecordState((SLRecordItf)pDevice->opensl.pAudioRecorder, SL_RECORDSTATE_RECORDING); + /* + TODO: Test Me. + + This is currently untested, so for now we are just returning default devices. + */ +#if 0 && !defined(MA_ANDROID) + ma_bool32 isTerminated = MA_FALSE; + + SLuint32 pDeviceIDs[128]; + SLint32 deviceCount = sizeof(pDeviceIDs) / sizeof(pDeviceIDs[0]); + + SLAudioIODeviceCapabilitiesItf deviceCaps; + SLresult resultSL = (*g_maEngineObjectSL)->GetInterface(g_maEngineObjectSL, SL_IID_AUDIOIODEVICECAPABILITIES, &deviceCaps); + if (resultSL != SL_RESULT_SUCCESS) { + /* The interface may not be supported so just report a default device. */ + goto return_default_device; + } + + /* Playback */ + if (!isTerminated) { + resultSL = (*deviceCaps)->GetAvailableAudioOutputs(deviceCaps, &deviceCount, pDeviceIDs); if (resultSL != SL_RESULT_SUCCESS) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to start internal capture device.", MA_FAILED_TO_START_BACKEND_DEVICE); + return MA_NO_DEVICE; } - periodSizeInBytes = (pDevice->capture.internalBufferSizeInFrames / pDevice->capture.internalPeriods) * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); - for (iPeriod = 0; iPeriod < pDevice->capture.internalPeriods; ++iPeriod) { - resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture, pDevice->opensl.pBufferCapture + (periodSizeInBytes * iPeriod), periodSizeInBytes); - if (resultSL != SL_RESULT_SUCCESS) { - MA_OPENSL_RECORD(pDevice->opensl.pAudioRecorder)->SetRecordState((SLRecordItf)pDevice->opensl.pAudioRecorder, SL_RECORDSTATE_STOPPED); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to enqueue buffer for capture device.", MA_FAILED_TO_START_BACKEND_DEVICE); + for (SLint32 iDevice = 0; iDevice < deviceCount; ++iDevice) { + ma_device_info deviceInfo; + MA_ZERO_OBJECT(&deviceInfo); + deviceInfo.id.opensl = pDeviceIDs[iDevice]; + + SLAudioOutputDescriptor desc; + resultSL = (*deviceCaps)->QueryAudioOutputCapabilities(deviceCaps, deviceInfo.id.opensl, &desc); + if (resultSL == SL_RESULT_SUCCESS) { + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), (const char*)desc.pDeviceName, (size_t)-1); + + ma_bool32 cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); + if (cbResult == MA_FALSE) { + isTerminated = MA_TRUE; + break; + } } } } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - resultSL = MA_OPENSL_PLAY(pDevice->opensl.pAudioPlayer)->SetPlayState((SLPlayItf)pDevice->opensl.pAudioPlayer, SL_PLAYSTATE_PLAYING); + /* Capture */ + if (!isTerminated) { + resultSL = (*deviceCaps)->GetAvailableAudioInputs(deviceCaps, &deviceCount, pDeviceIDs); if (resultSL != SL_RESULT_SUCCESS) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to start internal playback device.", MA_FAILED_TO_START_BACKEND_DEVICE); + return MA_NO_DEVICE; } - /* In playback mode (no duplex) we need to load some initial buffers. In duplex mode we need to enqueu silent buffers. */ - if (pDevice->type == ma_device_type_duplex) { - MA_ZERO_MEMORY(pDevice->opensl.pBufferPlayback, pDevice->playback.internalBufferSizeInFrames * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); - } else { - ma_device__read_frames_from_client(pDevice, pDevice->playback.internalBufferSizeInFrames, pDevice->opensl.pBufferPlayback); - } + for (SLint32 iDevice = 0; iDevice < deviceCount; ++iDevice) { + ma_device_info deviceInfo; + MA_ZERO_OBJECT(&deviceInfo); + deviceInfo.id.opensl = pDeviceIDs[iDevice]; - periodSizeInBytes = (pDevice->playback.internalBufferSizeInFrames / pDevice->playback.internalPeriods) * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); - for (iPeriod = 0; iPeriod < pDevice->playback.internalPeriods; ++iPeriod) { - resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback, pDevice->opensl.pBufferPlayback + (periodSizeInBytes * iPeriod), periodSizeInBytes); - if (resultSL != SL_RESULT_SUCCESS) { - MA_OPENSL_PLAY(pDevice->opensl.pAudioPlayer)->SetPlayState((SLPlayItf)pDevice->opensl.pAudioPlayer, SL_PLAYSTATE_STOPPED); - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to enqueue buffer for playback device.", MA_FAILED_TO_START_BACKEND_DEVICE); + SLAudioInputDescriptor desc; + resultSL = (*deviceCaps)->QueryAudioInputCapabilities(deviceCaps, deviceInfo.id.opensl, &desc); + if (resultSL == SL_RESULT_SUCCESS) { + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), (const char*)desc.deviceName, (size_t)-1); + + ma_bool32 cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); + if (cbResult == MA_FALSE) { + isTerminated = MA_TRUE; + break; + } } } } + return MA_SUCCESS; +#else + goto return_default_device; +#endif + +return_default_device:; + cbResult = MA_TRUE; + + /* Playback. */ + if (cbResult) { + ma_device_info deviceInfo; + MA_ZERO_OBJECT(&deviceInfo); + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); + cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); + } + + /* Capture. */ + if (cbResult) { + ma_device_info deviceInfo; + MA_ZERO_OBJECT(&deviceInfo); + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); + cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); + } + return MA_SUCCESS; } -ma_result ma_device_stop__opensl(ma_device* pDevice) +static ma_result ma_context_get_device_info__opensl(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) { - SLresult resultSL; - ma_stop_proc onStop; - - ma_assert(pDevice != NULL); + MA_ASSERT(pContext != NULL); - ma_assert(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it before stopping/uninitializing the device. */ + MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to get device info. */ if (g_maOpenSLInitCounter == 0) { return MA_INVALID_OPERATION; } - /* TODO: Wait until all buffers have been processed. Hint: Maybe SLAndroidSimpleBufferQueue::GetState() could be used in a loop? */ - - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - resultSL = MA_OPENSL_RECORD(pDevice->opensl.pAudioRecorder)->SetRecordState((SLRecordItf)pDevice->opensl.pAudioRecorder, SL_RECORDSTATE_STOPPED); - if (resultSL != SL_RESULT_SUCCESS) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to stop internal capture device.", MA_FAILED_TO_STOP_BACKEND_DEVICE); - } + /* No exclusive mode with OpenSL|ES. */ + if (shareMode == ma_share_mode_exclusive) { + return MA_SHARE_MODE_NOT_SUPPORTED; + } - MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->Clear((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture); + /* + TODO: Test Me. + + This is currently untested, so for now we are just returning default devices. + */ +#if 0 && !defined(MA_ANDROID) + SLAudioIODeviceCapabilitiesItf deviceCaps; + SLresult resultSL = (*g_maEngineObjectSL)->GetInterface(g_maEngineObjectSL, SL_IID_AUDIOIODEVICECAPABILITIES, &deviceCaps); + if (resultSL != SL_RESULT_SUCCESS) { + /* The interface may not be supported so just report a default device. */ + goto return_default_device; } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - resultSL = MA_OPENSL_PLAY(pDevice->opensl.pAudioPlayer)->SetPlayState((SLPlayItf)pDevice->opensl.pAudioPlayer, SL_PLAYSTATE_STOPPED); + if (deviceType == ma_device_type_playback) { + SLAudioOutputDescriptor desc; + resultSL = (*deviceCaps)->QueryAudioOutputCapabilities(deviceCaps, pDeviceID->opensl, &desc); if (resultSL != SL_RESULT_SUCCESS) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to stop internal playback device.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + return MA_NO_DEVICE; } - MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->Clear((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback); - } + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), (const char*)desc.pDeviceName, (size_t)-1); + } else { + SLAudioInputDescriptor desc; + resultSL = (*deviceCaps)->QueryAudioInputCapabilities(deviceCaps, pDeviceID->opensl, &desc); + if (resultSL != SL_RESULT_SUCCESS) { + return MA_NO_DEVICE; + } - /* Make sure the client is aware that the device has stopped. There may be an OpenSL|ES callback for this, but I haven't found it. */ - onStop = pDevice->onStop; - if (onStop) { - onStop(pDevice); + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), (const char*)desc.deviceName, (size_t)-1); } - return MA_SUCCESS; -} - - -ma_result ma_context_uninit__opensl(ma_context* pContext) -{ - ma_assert(pContext != NULL); - ma_assert(pContext->backend == ma_backend_opensl); - (void)pContext; + goto return_detailed_info; +#else + goto return_default_device; +#endif - /* Uninit global data. */ - if (g_maOpenSLInitCounter > 0) { - if (ma_atomic_decrement_32(&g_maOpenSLInitCounter) == 0) { - (*g_maEngineObjectSL)->Destroy(g_maEngineObjectSL); +return_default_device: + if (pDeviceID != NULL) { + if ((deviceType == ma_device_type_playback && pDeviceID->opensl != SL_DEFAULTDEVICEID_AUDIOOUTPUT) || + (deviceType == ma_device_type_capture && pDeviceID->opensl != SL_DEFAULTDEVICEID_AUDIOINPUT)) { + return MA_NO_DEVICE; /* Don't know the device. */ } } - return MA_SUCCESS; -} - -ma_result ma_context_init__opensl(const ma_context_config* pConfig, ma_context* pContext) -{ - ma_assert(pContext != NULL); - - (void)pConfig; - - /* Initialize global data first if applicable. */ - if (ma_atomic_increment_32(&g_maOpenSLInitCounter) == 1) { - SLresult resultSL = slCreateEngine(&g_maEngineObjectSL, 0, NULL, 0, NULL, NULL); - if (resultSL != SL_RESULT_SUCCESS) { - ma_atomic_decrement_32(&g_maOpenSLInitCounter); - return MA_NO_BACKEND; - } + /* Name / Description */ + if (deviceType == ma_device_type_playback) { + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); + } else { + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); + } - (*g_maEngineObjectSL)->Realize(g_maEngineObjectSL, SL_BOOLEAN_FALSE); + goto return_detailed_info; - resultSL = (*g_maEngineObjectSL)->GetInterface(g_maEngineObjectSL, SL_IID_ENGINE, &g_maEngineSL); - if (resultSL != SL_RESULT_SUCCESS) { - (*g_maEngineObjectSL)->Destroy(g_maEngineObjectSL); - ma_atomic_decrement_32(&g_maOpenSLInitCounter); - return MA_NO_BACKEND; - } - } - pContext->isBackendAsynchronous = MA_TRUE; +return_detailed_info: - pContext->onUninit = ma_context_uninit__opensl; - pContext->onDeviceIDEqual = ma_context_is_device_id_equal__opensl; - pContext->onEnumDevices = ma_context_enumerate_devices__opensl; - pContext->onGetDeviceInfo = ma_context_get_device_info__opensl; - pContext->onDeviceInit = ma_device_init__opensl; - pContext->onDeviceUninit = ma_device_uninit__opensl; - pContext->onDeviceStart = ma_device_start__opensl; - pContext->onDeviceStop = ma_device_stop__opensl; + /* + For now we're just outputting a set of values that are supported by the API but not necessarily supported + by the device natively. Later on we should work on this so that it more closely reflects the device's + actual native format. + */ + pDeviceInfo->minChannels = 1; + pDeviceInfo->maxChannels = 2; + pDeviceInfo->minSampleRate = 8000; + pDeviceInfo->maxSampleRate = 48000; + pDeviceInfo->formatCount = 2; + pDeviceInfo->formats[0] = ma_format_u8; + pDeviceInfo->formats[1] = ma_format_s16; +#if defined(MA_ANDROID) && __ANDROID_API__ >= 21 + pDeviceInfo->formats[pDeviceInfo->formatCount] = ma_format_f32; + pDeviceInfo->formatCount += 1; +#endif return MA_SUCCESS; } -#endif /* OpenSL|ES */ -/****************************************************************************** +#ifdef MA_ANDROID +/*void ma_buffer_queue_callback_capture__opensl_android(SLAndroidSimpleBufferQueueItf pBufferQueue, SLuint32 eventFlags, const void* pBuffer, SLuint32 bufferSize, SLuint32 dataUsed, void* pContext)*/ +static void ma_buffer_queue_callback_capture__opensl_android(SLAndroidSimpleBufferQueueItf pBufferQueue, void* pUserData) +{ + ma_device* pDevice = (ma_device*)pUserData; + size_t periodSizeInBytes; + ma_uint8* pBuffer; + SLresult resultSL; -Web Audio Backend + MA_ASSERT(pDevice != NULL); -******************************************************************************/ -#ifdef MA_HAS_WEBAUDIO -#include + (void)pBufferQueue; -ma_bool32 ma_is_capture_supported__webaudio() -{ - return EM_ASM_INT({ - return (navigator.mediaDevices !== undefined && navigator.mediaDevices.getUserMedia !== undefined); - }, 0) != 0; /* Must pass in a dummy argument for C99 compatibility. */ -} + /* + For now, don't do anything unless the buffer was fully processed. From what I can tell, it looks like + OpenSL|ES 1.1 improves on buffer queues to the point that we could much more intelligently handle this, + but unfortunately it looks like Android is only supporting OpenSL|ES 1.0.1 for now :( + */ -#ifdef __cplusplus -extern "C" { -#endif -EMSCRIPTEN_KEEPALIVE void ma_device_process_pcm_frames_capture__webaudio(ma_device* pDevice, int frameCount, float* pFrames) -{ - if (pDevice->type == ma_device_type_duplex) { - ma_device__handle_duplex_callback_capture(pDevice, (ma_uint32)frameCount, pFrames, &pDevice->webaudio.duplexRB); - } else { - ma_device__send_frames_to_client(pDevice, (ma_uint32)frameCount, pFrames); /* Send directly to the client. */ + /* Don't do anything if the device is not started. */ + if (pDevice->state != MA_STATE_STARTED) { + return; } -} -EMSCRIPTEN_KEEPALIVE void ma_device_process_pcm_frames_playback__webaudio(ma_device* pDevice, int frameCount, float* pFrames) -{ + /* Don't do anything if the device is being drained. */ + if (pDevice->opensl.isDrainingCapture) { + return; + } + + periodSizeInBytes = pDevice->capture.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + pBuffer = pDevice->opensl.pBufferCapture + (pDevice->opensl.currentBufferIndexCapture * periodSizeInBytes); + if (pDevice->type == ma_device_type_duplex) { - ma_device__handle_duplex_callback_playback(pDevice, (ma_uint32)frameCount, pFrames, &pDevice->webaudio.duplexRB); + ma_device__handle_duplex_callback_capture(pDevice, pDevice->capture.internalPeriodSizeInFrames, pBuffer, &pDevice->opensl.duplexRB); } else { - ma_device__read_frames_from_client(pDevice, (ma_uint32)frameCount, pFrames); /* Read directly from the device. */ + ma_device__send_frames_to_client(pDevice, pDevice->capture.internalPeriodSizeInFrames, pBuffer); } -} -#ifdef __cplusplus -} -#endif -ma_bool32 ma_context_is_device_id_equal__webaudio(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) -{ - ma_assert(pContext != NULL); - ma_assert(pID0 != NULL); - ma_assert(pID1 != NULL); - (void)pContext; + resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture, pBuffer, periodSizeInBytes); + if (resultSL != SL_RESULT_SUCCESS) { + return; + } - return ma_strcmp(pID0->webaudio, pID1->webaudio) == 0; + pDevice->opensl.currentBufferIndexCapture = (pDevice->opensl.currentBufferIndexCapture + 1) % pDevice->capture.internalPeriods; } -ma_result ma_context_enumerate_devices__webaudio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +static void ma_buffer_queue_callback_playback__opensl_android(SLAndroidSimpleBufferQueueItf pBufferQueue, void* pUserData) { - ma_bool32 cbResult = MA_TRUE; + ma_device* pDevice = (ma_device*)pUserData; + size_t periodSizeInBytes; + ma_uint8* pBuffer; + SLresult resultSL; - ma_assert(pContext != NULL); - ma_assert(callback != NULL); + MA_ASSERT(pDevice != NULL); - /* Only supporting default devices for now. */ + (void)pBufferQueue; - /* Playback. */ - if (cbResult) { - ma_device_info deviceInfo; - ma_zero_object(&deviceInfo); - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); - cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); + /* Don't do anything if the device is not started. */ + if (pDevice->state != MA_STATE_STARTED) { + return; } - /* Capture. */ - if (cbResult) { - if (ma_is_capture_supported__webaudio()) { - ma_device_info deviceInfo; - ma_zero_object(&deviceInfo); - ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); - cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); - } + /* Don't do anything if the device is being drained. */ + if (pDevice->opensl.isDrainingPlayback) { + return; } - return MA_SUCCESS; -} - -ma_result ma_context_get_device_info__webaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) -{ - ma_assert(pContext != NULL); + periodSizeInBytes = pDevice->playback.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + pBuffer = pDevice->opensl.pBufferPlayback + (pDevice->opensl.currentBufferIndexPlayback * periodSizeInBytes); - /* No exclusive mode with Web Audio. */ - if (shareMode == ma_share_mode_exclusive) { - return MA_SHARE_MODE_NOT_SUPPORTED; + if (pDevice->type == ma_device_type_duplex) { + ma_device__handle_duplex_callback_playback(pDevice, pDevice->playback.internalPeriodSizeInFrames, pBuffer, &pDevice->opensl.duplexRB); + } else { + ma_device__read_frames_from_client(pDevice, pDevice->playback.internalPeriodSizeInFrames, pBuffer); } - if (deviceType == ma_device_type_capture && !ma_is_capture_supported__webaudio()) { - return MA_NO_DEVICE; + resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback, pBuffer, periodSizeInBytes); + if (resultSL != SL_RESULT_SUCCESS) { + return; } + pDevice->opensl.currentBufferIndexPlayback = (pDevice->opensl.currentBufferIndexPlayback + 1) % pDevice->playback.internalPeriods; +} +#endif - ma_zero_memory(pDeviceInfo->id.webaudio, sizeof(pDeviceInfo->id.webaudio)); +static void ma_device_uninit__opensl(ma_device* pDevice) +{ + MA_ASSERT(pDevice != NULL); - /* Only supporting default devices for now. */ - if (deviceType == ma_device_type_playback) { - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); - } else { - ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); + MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it before uninitializing the device. */ + if (g_maOpenSLInitCounter == 0) { + return; } - /* Web Audio can support any number of channels and sample rates. It only supports f32 formats, however. */ - pDeviceInfo->minChannels = 1; - pDeviceInfo->maxChannels = MA_MAX_CHANNELS; - if (pDeviceInfo->maxChannels > 32) { - pDeviceInfo->maxChannels = 32; /* Maximum output channel count is 32 for createScriptProcessor() (JavaScript). */ + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + if (pDevice->opensl.pAudioRecorderObj) { + MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->Destroy((SLObjectItf)pDevice->opensl.pAudioRecorderObj); + } + + ma__free_from_callbacks(pDevice->opensl.pBufferCapture, &pDevice->pContext->allocationCallbacks); } - /* We can query the sample rate by just using a temporary audio context. */ - pDeviceInfo->minSampleRate = EM_ASM_INT({ - try { - var temp = new (window.AudioContext || window.webkitAudioContext)(); - var sampleRate = temp.sampleRate; - temp.close(); - return sampleRate; - } catch(e) { - return 0; + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + if (pDevice->opensl.pAudioPlayerObj) { + MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->Destroy((SLObjectItf)pDevice->opensl.pAudioPlayerObj); + } + if (pDevice->opensl.pOutputMixObj) { + MA_OPENSL_OBJ(pDevice->opensl.pOutputMixObj)->Destroy((SLObjectItf)pDevice->opensl.pOutputMixObj); } - }, 0); /* Must pass in a dummy argument for C99 compatibility. */ - pDeviceInfo->maxSampleRate = pDeviceInfo->minSampleRate; - if (pDeviceInfo->minSampleRate == 0) { - return MA_NO_DEVICE; - } - /* Web Audio only supports f32. */ - pDeviceInfo->formatCount = 1; - pDeviceInfo->formats[0] = ma_format_f32; + ma__free_from_callbacks(pDevice->opensl.pBufferPlayback, &pDevice->pContext->allocationCallbacks); + } - return MA_SUCCESS; + if (pDevice->type == ma_device_type_duplex) { + ma_pcm_rb_uninit(&pDevice->opensl.duplexRB); + } } +#if defined(MA_ANDROID) && __ANDROID_API__ >= 21 +typedef SLAndroidDataFormat_PCM_EX ma_SLDataFormat_PCM; +#else +typedef SLDataFormat_PCM ma_SLDataFormat_PCM; +#endif -void ma_device_uninit_by_index__webaudio(ma_device* pDevice, ma_device_type deviceType, int deviceIndex) +static ma_result ma_SLDataFormat_PCM_init__opensl(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, const ma_channel* channelMap, ma_SLDataFormat_PCM* pDataFormat) { - ma_assert(pDevice != NULL); +#if defined(MA_ANDROID) && __ANDROID_API__ >= 21 + if (format == ma_format_f32) { + pDataFormat->formatType = SL_ANDROID_DATAFORMAT_PCM_EX; + pDataFormat->representation = SL_ANDROID_PCM_REPRESENTATION_FLOAT; + } else { + pDataFormat->formatType = SL_DATAFORMAT_PCM; + } +#else + pDataFormat->formatType = SL_DATAFORMAT_PCM; +#endif - EM_ASM({ - var device = miniaudio.get_device_by_index($0); + pDataFormat->numChannels = channels; + ((SLDataFormat_PCM*)pDataFormat)->samplesPerSec = ma_round_to_standard_sample_rate__opensl(sampleRate * 1000); /* In millihertz. Annoyingly, the sample rate variable is named differently between SLAndroidDataFormat_PCM_EX and SLDataFormat_PCM */ + pDataFormat->bitsPerSample = ma_get_bytes_per_sample(format)*8; + pDataFormat->channelMask = ma_channel_map_to_channel_mask__opensl(channelMap, channels); + pDataFormat->endianness = (ma_is_little_endian()) ? SL_BYTEORDER_LITTLEENDIAN : SL_BYTEORDER_BIGENDIAN; - /* Make sure all nodes are disconnected and marked for collection. */ - if (device.scriptNode !== undefined) { - device.scriptNode.onaudioprocess = function(e) {}; /* We want to reset the callback to ensure it doesn't get called after AudioContext.close() has returned. Shouldn't happen since we're disconnecting, but just to be safe... */ - device.scriptNode.disconnect(); - device.scriptNode = undefined; + /* + Android has a few restrictions on the format as documented here: https://developer.android.com/ndk/guides/audio/opensl-for-android.html + - Only mono and stereo is supported. + - Only u8 and s16 formats are supported. + - Maximum sample rate of 48000. + */ +#ifdef MA_ANDROID + if (pDataFormat->numChannels > 2) { + pDataFormat->numChannels = 2; + } +#if __ANDROID_API__ >= 21 + if (pDataFormat->formatType == SL_ANDROID_DATAFORMAT_PCM_EX) { + /* It's floating point. */ + MA_ASSERT(pDataFormat->representation == SL_ANDROID_PCM_REPRESENTATION_FLOAT); + if (pDataFormat->bitsPerSample > 32) { + pDataFormat->bitsPerSample = 32; } - if (device.streamNode !== undefined) { - device.streamNode.disconnect(); - device.streamNode = undefined; + } else { + if (pDataFormat->bitsPerSample > 16) { + pDataFormat->bitsPerSample = 16; } + } +#else + if (pDataFormat->bitsPerSample > 16) { + pDataFormat->bitsPerSample = 16; + } +#endif + if (((SLDataFormat_PCM*)pDataFormat)->samplesPerSec > SL_SAMPLINGRATE_48) { + ((SLDataFormat_PCM*)pDataFormat)->samplesPerSec = SL_SAMPLINGRATE_48; + } +#endif - /* - Stop the device. I think there is a chance the callback could get fired after calling this, hence why we want - to clear the callback before closing. - */ - device.webaudio.close(); - device.webaudio = undefined; + pDataFormat->containerSize = pDataFormat->bitsPerSample; /* Always tightly packed for now. */ - /* Can't forget to free the intermediary buffer. This is the buffer that's shared between JavaScript and C. */ - if (device.intermediaryBuffer !== undefined) { - Module._free(device.intermediaryBuffer); - device.intermediaryBuffer = undefined; - device.intermediaryBufferView = undefined; - device.intermediaryBufferSizeInBytes = undefined; + return MA_SUCCESS; +} + +static ma_result ma_deconstruct_SLDataFormat_PCM__opensl(ma_SLDataFormat_PCM* pDataFormat, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap) +{ + ma_bool32 isFloatingPoint = MA_FALSE; +#if defined(MA_ANDROID) && __ANDROID_API__ >= 21 + if (pDataFormat->formatType == SL_ANDROID_DATAFORMAT_PCM_EX) { + MA_ASSERT(pDataFormat->representation == SL_ANDROID_PCM_REPRESENTATION_FLOAT); + isFloatingPoint = MA_TRUE; + } +#endif + if (isFloatingPoint) { + if (pDataFormat->bitsPerSample == 32) { + *pFormat = ma_format_f32; + } + } else { + if (pDataFormat->bitsPerSample == 8) { + *pFormat = ma_format_u8; + } else if (pDataFormat->bitsPerSample == 16) { + *pFormat = ma_format_s16; + } else if (pDataFormat->bitsPerSample == 24) { + *pFormat = ma_format_s24; + } else if (pDataFormat->bitsPerSample == 32) { + *pFormat = ma_format_s32; } + } - /* Make sure the device is untracked so the slot can be reused later. */ - miniaudio.untrack_device_by_index($0); - }, deviceIndex, deviceType); + *pChannels = pDataFormat->numChannels; + *pSampleRate = ((SLDataFormat_PCM*)pDataFormat)->samplesPerSec / 1000; + ma_channel_mask_to_channel_map__opensl(pDataFormat->channelMask, pDataFormat->numChannels, pChannelMap); + + return MA_SUCCESS; } -void ma_device_uninit__webaudio(ma_device* pDevice) -{ - ma_assert(pDevice != NULL); +static ma_result ma_device_init__opensl(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) +{ +#ifdef MA_ANDROID + SLDataLocator_AndroidSimpleBufferQueue queue; + SLresult resultSL; + ma_uint32 periodSizeInFrames; + size_t bufferSizeInBytes; + const SLInterfaceID itfIDs1[] = {SL_IID_ANDROIDSIMPLEBUFFERQUEUE}; + const SLboolean itfIDsRequired1[] = {SL_BOOLEAN_TRUE}; +#endif + + (void)pContext; - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_capture, pDevice->webaudio.indexCapture); + MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to initialize a new device. */ + if (g_maOpenSLInitCounter == 0) { + return MA_INVALID_OPERATION; } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_playback, pDevice->webaudio.indexPlayback); + if (pConfig->deviceType == ma_device_type_loopback) { + return MA_DEVICE_TYPE_NOT_SUPPORTED; } - if (pDevice->type == ma_device_type_duplex) { - ma_pcm_rb_uninit(&pDevice->webaudio.duplexRB); + /* + For now, only supporting Android implementations of OpenSL|ES since that's the only one I've + been able to test with and I currently depend on Android-specific extensions (simple buffer + queues). + */ +#ifdef MA_ANDROID + /* No exclusive mode with OpenSL|ES. */ + if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) || + ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) { + return MA_SHARE_MODE_NOT_SUPPORTED; } -} -ma_result ma_device_init_by_type__webaudio(ma_context* pContext, const ma_device_config* pConfig, ma_device_type deviceType, ma_device* pDevice) -{ - int deviceIndex; - ma_uint32 internalBufferSizeInFrames; + /* Now we can start initializing the device properly. */ + MA_ASSERT(pDevice != NULL); + MA_ZERO_OBJECT(&pDevice->opensl); + + queue.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE; + queue.numBuffers = pConfig->periods; - ma_assert(pContext != NULL); - ma_assert(pConfig != NULL); - ma_assert(deviceType != ma_device_type_duplex); - ma_assert(pDevice != NULL); - if (deviceType == ma_device_type_capture && !ma_is_capture_supported__webaudio()) { - return MA_NO_DEVICE; - } + if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { + ma_SLDataFormat_PCM pcm; + SLDataLocator_IODevice locatorDevice; + SLDataSource source; + SLDataSink sink; - /* Try calculating an appropriate buffer size. */ - internalBufferSizeInFrames = pConfig->bufferSizeInFrames; - if (internalBufferSizeInFrames == 0) { - internalBufferSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->bufferSizeInMilliseconds, pConfig->sampleRate); - } + ma_SLDataFormat_PCM_init__opensl(pConfig->capture.format, pConfig->capture.channels, pConfig->sampleRate, pConfig->capture.channelMap, &pcm); - /* The size of the buffer must be a power of 2 and between 256 and 16384. */ - if (internalBufferSizeInFrames < 256) { - internalBufferSizeInFrames = 256; - } else if (internalBufferSizeInFrames > 16384) { - internalBufferSizeInFrames = 16384; - } else { - internalBufferSizeInFrames = ma_next_power_of_2(internalBufferSizeInFrames); - } + locatorDevice.locatorType = SL_DATALOCATOR_IODEVICE; + locatorDevice.deviceType = SL_IODEVICE_AUDIOINPUT; + locatorDevice.deviceID = (pConfig->capture.pDeviceID == NULL) ? SL_DEFAULTDEVICEID_AUDIOINPUT : pConfig->capture.pDeviceID->opensl; + locatorDevice.device = NULL; - /* We create the device on the JavaScript side and reference it using an index. We use this to make it possible to reference the device between JavaScript and C. */ - deviceIndex = EM_ASM_INT({ - var channels = $0; - var sampleRate = $1; - var bufferSize = $2; /* In PCM frames. */ - var isCapture = $3; - var pDevice = $4; + source.pLocator = &locatorDevice; + source.pFormat = NULL; - if (typeof(miniaudio) === 'undefined') { - return -1; /* Context not initialized. */ + sink.pLocator = &queue; + sink.pFormat = (SLDataFormat_PCM*)&pcm; + + resultSL = (*g_maEngineSL)->CreateAudioRecorder(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioRecorderObj, &source, &sink, 1, itfIDs1, itfIDsRequired1); + if (resultSL == SL_RESULT_CONTENT_UNSUPPORTED) { + /* Unsupported format. Fall back to something safer and try again. If this fails, just abort. */ + pcm.formatType = SL_DATAFORMAT_PCM; + pcm.numChannels = 1; + ((SLDataFormat_PCM*)&pcm)->samplesPerSec = SL_SAMPLINGRATE_16; /* The name of the sample rate variable is different between SLAndroidDataFormat_PCM_EX and SLDataFormat_PCM. */ + pcm.bitsPerSample = 16; + pcm.containerSize = pcm.bitsPerSample; /* Always tightly packed for now. */ + pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT; + resultSL = (*g_maEngineSL)->CreateAudioRecorder(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioRecorderObj, &source, &sink, 1, itfIDs1, itfIDsRequired1); } - var device = {}; + if (resultSL != SL_RESULT_SUCCESS) { + ma_device_uninit__opensl(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to create audio recorder.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - /* The AudioContext must be created in a suspended state. */ - device.webaudio = new (window.AudioContext || window.webkitAudioContext)({sampleRate:sampleRate}); - device.webaudio.suspend(); + if (MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->Realize((SLObjectItf)pDevice->opensl.pAudioRecorderObj, SL_BOOLEAN_FALSE) != SL_RESULT_SUCCESS) { + ma_device_uninit__opensl(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to realize audio recorder.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - /* - We need an intermediary buffer which we use for JavaScript and C interop. This buffer stores interleaved f32 PCM data. Because it's passed between - JavaScript and C it needs to be allocated and freed using Module._malloc() and Module._free(). - */ - device.intermediaryBufferSizeInBytes = channels * bufferSize * 4; - device.intermediaryBuffer = Module._malloc(device.intermediaryBufferSizeInBytes); - device.intermediaryBufferView = new Float32Array(Module.HEAPF32.buffer, device.intermediaryBuffer, device.intermediaryBufferSizeInBytes); + if (MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioRecorderObj, SL_IID_RECORD, &pDevice->opensl.pAudioRecorder) != SL_RESULT_SUCCESS) { + ma_device_uninit__opensl(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_RECORD interface.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - /* - Both playback and capture devices use a ScriptProcessorNode for performing per-sample operations. + if (MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioRecorderObj, SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &pDevice->opensl.pBufferQueueCapture) != SL_RESULT_SUCCESS) { + ma_device_uninit__opensl(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_ANDROIDSIMPLEBUFFERQUEUE interface.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - ScriptProcessorNode is actually deprecated so this is likely to be temporary. The way this works for playback is very simple. You just set a callback - that's periodically fired, just like a normal audio callback function. But apparently this design is "flawed" and is now deprecated in favour of - something called AudioWorklets which _forces_ you to load a _separate_ .js file at run time... nice... Hopefully ScriptProcessorNode will continue to - work for years to come, but this may need to change to use AudioSourceBufferNode instead, which I think is what Emscripten uses for it's built-in SDL - implementation. I'll be avoiding that insane AudioWorklet API like the plague... + if (MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->RegisterCallback((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture, ma_buffer_queue_callback_capture__opensl_android, pDevice) != SL_RESULT_SUCCESS) { + ma_device_uninit__opensl(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to register buffer queue callback.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - For capture it is a bit unintuitive. We use the ScriptProccessorNode _only_ to get the raw PCM data. It is connected to an AudioContext just like the - playback case, however we just output silence to the AudioContext instead of passing any real data. It would make more sense to me to use the - MediaRecorder API, but unfortunately you need to specify a MIME time (Opus, Vorbis, etc.) for the binary blob that's returned to the client, but I've - been unable to figure out how to get this as raw PCM. The closes I can think is to use the MIME type for WAV files and just parse it, but I don't know - how well this would work. Although ScriptProccessorNode is deprecated, in practice it seems to have pretty good browser support so I'm leaving it like - this for now. If anything knows how I could get raw PCM data using the MediaRecorder API please let me know! - */ - device.scriptNode = device.webaudio.createScriptProcessor(bufferSize, channels, channels); + /* The internal format is determined by the "pcm" object. */ + ma_deconstruct_SLDataFormat_PCM__opensl(&pcm, &pDevice->capture.internalFormat, &pDevice->capture.internalChannels, &pDevice->capture.internalSampleRate, pDevice->capture.internalChannelMap); - if (isCapture) { - device.scriptNode.onaudioprocess = function(e) { - if (device.intermediaryBuffer === undefined) { - return; /* This means the device has been uninitialized. */ - } + /* Buffer. */ + periodSizeInFrames = pConfig->periodSizeInFrames; + if (periodSizeInFrames == 0) { + periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->periodSizeInMilliseconds, pDevice->capture.internalSampleRate); + } + pDevice->capture.internalPeriods = pConfig->periods; + pDevice->capture.internalPeriodSizeInFrames = periodSizeInFrames; + pDevice->opensl.currentBufferIndexCapture = 0; - /* Make sure silence it output to the AudioContext destination. Not doing this will cause sound to come out of the speakers! */ - for (var iChannel = 0; iChannel < e.outputBuffer.numberOfChannels; ++iChannel) { - e.outputBuffer.getChannelData(iChannel).fill(0.0); - } + bufferSizeInBytes = pDevice->capture.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels) * pDevice->capture.internalPeriods; + pDevice->opensl.pBufferCapture = (ma_uint8*)ma__calloc_from_callbacks(bufferSizeInBytes, &pContext->allocationCallbacks); + if (pDevice->opensl.pBufferCapture == NULL) { + ma_device_uninit__opensl(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to allocate memory for data buffer.", MA_OUT_OF_MEMORY); + } + MA_ZERO_MEMORY(pDevice->opensl.pBufferCapture, bufferSizeInBytes); + } - /* There are some situations where we may want to send silence to the client. */ - var sendSilence = false; - if (device.streamNode === undefined) { - sendSilence = true; - } + if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { + ma_SLDataFormat_PCM pcm; + SLDataSource source; + SLDataLocator_OutputMix outmixLocator; + SLDataSink sink; - /* Sanity check. This will never happen, right? */ - if (e.inputBuffer.numberOfChannels != channels) { - console.log("Capture: Channel count mismatch. " + e.inputBufer.numberOfChannels + " != " + channels + ". Sending silence."); - sendSilence = true; - } + ma_SLDataFormat_PCM_init__opensl(pConfig->playback.format, pConfig->playback.channels, pConfig->sampleRate, pConfig->playback.channelMap, &pcm); - /* This looped design guards against the situation where e.inputBuffer is a different size to the original buffer size. Should never happen in practice. */ - var totalFramesProcessed = 0; - while (totalFramesProcessed < e.inputBuffer.length) { - var framesRemaining = e.inputBuffer.length - totalFramesProcessed; - var framesToProcess = framesRemaining; - if (framesToProcess > (device.intermediaryBufferSizeInBytes/channels/4)) { - framesToProcess = (device.intermediaryBufferSizeInBytes/channels/4); - } + resultSL = (*g_maEngineSL)->CreateOutputMix(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pOutputMixObj, 0, NULL, NULL); + if (resultSL != SL_RESULT_SUCCESS) { + ma_device_uninit__opensl(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to create output mix.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - /* We need to do the reverse of the playback case. We need to interleave the input data and copy it into the intermediary buffer. Then we send it to the client. */ - if (sendSilence) { - device.intermediaryBufferView.fill(0.0); - } else { - for (var iFrame = 0; iFrame < framesToProcess; ++iFrame) { - for (var iChannel = 0; iChannel < e.inputBuffer.numberOfChannels; ++iChannel) { - device.intermediaryBufferView[iFrame*channels + iChannel] = e.inputBuffer.getChannelData(iChannel)[totalFramesProcessed + iFrame]; - } - } - } + if (MA_OPENSL_OBJ(pDevice->opensl.pOutputMixObj)->Realize((SLObjectItf)pDevice->opensl.pOutputMixObj, SL_BOOLEAN_FALSE)) { + ma_device_uninit__opensl(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to realize output mix object.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - /* Send data to the client from our intermediary buffer. */ - ccall("ma_device_process_pcm_frames_capture__webaudio", "undefined", ["number", "number", "number"], [pDevice, framesToProcess, device.intermediaryBuffer]); + if (MA_OPENSL_OBJ(pDevice->opensl.pOutputMixObj)->GetInterface((SLObjectItf)pDevice->opensl.pOutputMixObj, SL_IID_OUTPUTMIX, &pDevice->opensl.pOutputMix) != SL_RESULT_SUCCESS) { + ma_device_uninit__opensl(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_OUTPUTMIX interface.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - totalFramesProcessed += framesToProcess; - } - }; + /* Set the output device. */ + if (pConfig->playback.pDeviceID != NULL) { + SLuint32 deviceID_OpenSL = pConfig->playback.pDeviceID->opensl; + MA_OPENSL_OUTPUTMIX(pDevice->opensl.pOutputMix)->ReRoute((SLOutputMixItf)pDevice->opensl.pOutputMix, 1, &deviceID_OpenSL); + } + + source.pLocator = &queue; + source.pFormat = (SLDataFormat_PCM*)&pcm; - navigator.mediaDevices.getUserMedia({audio:true, video:false}) - .then(function(stream) { - device.streamNode = device.webaudio.createMediaStreamSource(stream); - device.streamNode.connect(device.scriptNode); - device.scriptNode.connect(device.webaudio.destination); - }) - .catch(function(error) { - /* I think this should output silence... */ - device.scriptNode.connect(device.webaudio.destination); - }); - } else { - device.scriptNode.onaudioprocess = function(e) { - if (device.intermediaryBuffer === undefined) { - return; /* This means the device has been uninitialized. */ - } + outmixLocator.locatorType = SL_DATALOCATOR_OUTPUTMIX; + outmixLocator.outputMix = (SLObjectItf)pDevice->opensl.pOutputMixObj; - var outputSilence = false; + sink.pLocator = &outmixLocator; + sink.pFormat = NULL; - /* Sanity check. This will never happen, right? */ - if (e.outputBuffer.numberOfChannels != channels) { - console.log("Playback: Channel count mismatch. " + e.outputBufer.numberOfChannels + " != " + channels + ". Outputting silence."); - outputSilence = true; - return; - } + resultSL = (*g_maEngineSL)->CreateAudioPlayer(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioPlayerObj, &source, &sink, 1, itfIDs1, itfIDsRequired1); + if (resultSL == SL_RESULT_CONTENT_UNSUPPORTED) { + /* Unsupported format. Fall back to something safer and try again. If this fails, just abort. */ + pcm.formatType = SL_DATAFORMAT_PCM; + pcm.numChannels = 2; + ((SLDataFormat_PCM*)&pcm)->samplesPerSec = SL_SAMPLINGRATE_16; + pcm.bitsPerSample = 16; + pcm.containerSize = pcm.bitsPerSample; /* Always tightly packed for now. */ + pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT; + resultSL = (*g_maEngineSL)->CreateAudioPlayer(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioPlayerObj, &source, &sink, 1, itfIDs1, itfIDsRequired1); + } - /* This looped design guards against the situation where e.outputBuffer is a different size to the original buffer size. Should never happen in practice. */ - var totalFramesProcessed = 0; - while (totalFramesProcessed < e.outputBuffer.length) { - var framesRemaining = e.outputBuffer.length - totalFramesProcessed; - var framesToProcess = framesRemaining; - if (framesToProcess > (device.intermediaryBufferSizeInBytes/channels/4)) { - framesToProcess = (device.intermediaryBufferSizeInBytes/channels/4); - } + if (resultSL != SL_RESULT_SUCCESS) { + ma_device_uninit__opensl(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to create audio player.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - /* Read data from the client into our intermediary buffer. */ - ccall("ma_device_process_pcm_frames_playback__webaudio", "undefined", ["number", "number", "number"], [pDevice, framesToProcess, device.intermediaryBuffer]); + if (MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->Realize((SLObjectItf)pDevice->opensl.pAudioPlayerObj, SL_BOOLEAN_FALSE) != SL_RESULT_SUCCESS) { + ma_device_uninit__opensl(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to realize audio player.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - /* At this point we'll have data in our intermediary buffer which we now need to deinterleave and copy over to the output buffers. */ - if (outputSilence) { - for (var iChannel = 0; iChannel < e.outputBuffer.numberOfChannels; ++iChannel) { - e.outputBuffer.getChannelData(iChannel).fill(0.0); - } - } else { - for (var iChannel = 0; iChannel < e.outputBuffer.numberOfChannels; ++iChannel) { - for (var iFrame = 0; iFrame < framesToProcess; ++iFrame) { - e.outputBuffer.getChannelData(iChannel)[totalFramesProcessed + iFrame] = device.intermediaryBufferView[iFrame*channels + iChannel]; - } - } - } + if (MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioPlayerObj, SL_IID_PLAY, &pDevice->opensl.pAudioPlayer) != SL_RESULT_SUCCESS) { + ma_device_uninit__opensl(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_PLAY interface.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - totalFramesProcessed += framesToProcess; - } - }; + if (MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioPlayerObj, SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &pDevice->opensl.pBufferQueuePlayback) != SL_RESULT_SUCCESS) { + ma_device_uninit__opensl(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_ANDROIDSIMPLEBUFFERQUEUE interface.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); + } - device.scriptNode.connect(device.webaudio.destination); + if (MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->RegisterCallback((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback, ma_buffer_queue_callback_playback__opensl_android, pDevice) != SL_RESULT_SUCCESS) { + ma_device_uninit__opensl(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to register buffer queue callback.", MA_FAILED_TO_OPEN_BACKEND_DEVICE); } - return miniaudio.track_device(device); - }, (deviceType == ma_device_type_capture) ? pConfig->capture.channels : pConfig->playback.channels, pConfig->sampleRate, internalBufferSizeInFrames, deviceType == ma_device_type_capture, pDevice); + /* The internal format is determined by the "pcm" object. */ + ma_deconstruct_SLDataFormat_PCM__opensl(&pcm, &pDevice->playback.internalFormat, &pDevice->playback.internalChannels, &pDevice->playback.internalSampleRate, pDevice->playback.internalChannelMap); - if (deviceIndex < 0) { - return MA_FAILED_TO_OPEN_BACKEND_DEVICE; + /* Buffer. */ + periodSizeInFrames = pConfig->periodSizeInFrames; + if (periodSizeInFrames == 0) { + periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->periodSizeInMilliseconds, pDevice->playback.internalSampleRate); + } + pDevice->playback.internalPeriods = pConfig->periods; + pDevice->playback.internalPeriodSizeInFrames = periodSizeInFrames; + pDevice->opensl.currentBufferIndexPlayback = 0; + + bufferSizeInBytes = pDevice->playback.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels) * pDevice->playback.internalPeriods; + pDevice->opensl.pBufferPlayback = (ma_uint8*)ma__calloc_from_callbacks(bufferSizeInBytes, &pContext->allocationCallbacks); + if (pDevice->opensl.pBufferPlayback == NULL) { + ma_device_uninit__opensl(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to allocate memory for data buffer.", MA_OUT_OF_MEMORY); + } + MA_ZERO_MEMORY(pDevice->opensl.pBufferPlayback, bufferSizeInBytes); } - if (deviceType == ma_device_type_capture) { - pDevice->webaudio.indexCapture = deviceIndex; - pDevice->capture.internalFormat = ma_format_f32; - pDevice->capture.internalChannels = pConfig->capture.channels; - ma_get_standard_channel_map(ma_standard_channel_map_webaudio, pDevice->capture.internalChannels, pDevice->capture.internalChannelMap); - pDevice->capture.internalSampleRate = EM_ASM_INT({ return miniaudio.get_device_by_index($0).webaudio.sampleRate; }, deviceIndex); - pDevice->capture.internalBufferSizeInFrames = internalBufferSizeInFrames; - pDevice->capture.internalPeriods = 1; - } else { - pDevice->webaudio.indexPlayback = deviceIndex; - pDevice->playback.internalFormat = ma_format_f32; - pDevice->playback.internalChannels = pConfig->playback.channels; - ma_get_standard_channel_map(ma_standard_channel_map_webaudio, pDevice->playback.internalChannels, pDevice->playback.internalChannelMap); - pDevice->playback.internalSampleRate = EM_ASM_INT({ return miniaudio.get_device_by_index($0).webaudio.sampleRate; }, deviceIndex); - pDevice->playback.internalBufferSizeInFrames = internalBufferSizeInFrames; - pDevice->playback.internalPeriods = 1; + if (pConfig->deviceType == ma_device_type_duplex) { + ma_uint32 rbSizeInFrames = (ma_uint32)ma_calculate_frame_count_after_resampling(pDevice->sampleRate, pDevice->capture.internalSampleRate, pDevice->capture.internalPeriodSizeInFrames) * pDevice->capture.internalPeriods; + ma_result result = ma_pcm_rb_init(pDevice->capture.format, pDevice->capture.channels, rbSizeInFrames, NULL, &pDevice->pContext->allocationCallbacks, &pDevice->opensl.duplexRB); + if (result != MA_SUCCESS) { + ma_device_uninit__opensl(pDevice); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to initialize ring buffer.", result); + } + + /* We need a period to act as a buffer for cases where the playback and capture device's end up desyncing. */ + { + ma_uint32 marginSizeInFrames = rbSizeInFrames / pDevice->capture.internalPeriods; + void* pMarginData; + ma_pcm_rb_acquire_write(&pDevice->opensl.duplexRB, &marginSizeInFrames, &pMarginData); + { + MA_ZERO_MEMORY(pMarginData, marginSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels)); + } + ma_pcm_rb_commit_write(&pDevice->opensl.duplexRB, marginSizeInFrames, pMarginData); + } } return MA_SUCCESS; +#else + return MA_NO_BACKEND; /* Non-Android implementations are not supported. */ +#endif } -ma_result ma_device_init__webaudio(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) +static ma_result ma_device_start__opensl(ma_device* pDevice) { - ma_result result; + SLresult resultSL; + size_t periodSizeInBytes; + ma_uint32 iPeriod; - if (pConfig->deviceType == ma_device_type_loopback) { - return MA_DEVICE_TYPE_NOT_SUPPORTED; - } + MA_ASSERT(pDevice != NULL); - /* No exclusive mode with Web Audio. */ - if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) || - ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) { - return MA_SHARE_MODE_NOT_SUPPORTED; + MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to start the device. */ + if (g_maOpenSLInitCounter == 0) { + return MA_INVALID_OPERATION; } - if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { - result = ma_device_init_by_type__webaudio(pContext, pConfig, ma_device_type_capture, pDevice); - if (result != MA_SUCCESS) { - return result; + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + resultSL = MA_OPENSL_RECORD(pDevice->opensl.pAudioRecorder)->SetRecordState((SLRecordItf)pDevice->opensl.pAudioRecorder, SL_RECORDSTATE_RECORDING); + if (resultSL != SL_RESULT_SUCCESS) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to start internal capture device.", MA_FAILED_TO_START_BACKEND_DEVICE); } - } - if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { - result = ma_device_init_by_type__webaudio(pContext, pConfig, ma_device_type_playback, pDevice); - if (result != MA_SUCCESS) { - if (pConfig->deviceType == ma_device_type_duplex) { - ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_capture, pDevice->webaudio.indexCapture); + periodSizeInBytes = pDevice->capture.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels); + for (iPeriod = 0; iPeriod < pDevice->capture.internalPeriods; ++iPeriod) { + resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture, pDevice->opensl.pBufferCapture + (periodSizeInBytes * iPeriod), periodSizeInBytes); + if (resultSL != SL_RESULT_SUCCESS) { + MA_OPENSL_RECORD(pDevice->opensl.pAudioRecorder)->SetRecordState((SLRecordItf)pDevice->opensl.pAudioRecorder, SL_RECORDSTATE_STOPPED); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to enqueue buffer for capture device.", MA_FAILED_TO_START_BACKEND_DEVICE); } - return result; } } - /* - We need a ring buffer for moving data from the capture device to the playback device. The capture callback is the producer - and the playback callback is the consumer. The buffer needs to be large enough to hold internalBufferSizeInFrames based on - the external sample rate. - */ - if (pConfig->deviceType == ma_device_type_duplex) { - ma_uint32 rbSizeInFrames = (ma_uint32)ma_calculate_frame_count_after_src(pDevice->sampleRate, pDevice->capture.internalSampleRate, pDevice->capture.internalBufferSizeInFrames) * 2; - result = ma_pcm_rb_init(pDevice->capture.format, pDevice->capture.channels, rbSizeInFrames, NULL, &pDevice->webaudio.duplexRB); - if (result != MA_SUCCESS) { - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_capture, pDevice->webaudio.indexCapture); - } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_playback, pDevice->webaudio.indexPlayback); - } - return result; + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + resultSL = MA_OPENSL_PLAY(pDevice->opensl.pAudioPlayer)->SetPlayState((SLPlayItf)pDevice->opensl.pAudioPlayer, SL_PLAYSTATE_PLAYING); + if (resultSL != SL_RESULT_SUCCESS) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to start internal playback device.", MA_FAILED_TO_START_BACKEND_DEVICE); } - /* We need a period to act as a buffer for cases where the playback and capture device's end up desyncing. */ - { - ma_uint32 marginSizeInFrames = rbSizeInFrames / 3; /* <-- Dividing by 3 because internalPeriods is always set to 1 for WebAudio. */ - void* pMarginData; - ma_pcm_rb_acquire_write(&pDevice->webaudio.duplexRB, &marginSizeInFrames, &pMarginData); - { - ma_zero_memory(pMarginData, marginSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels)); + /* In playback mode (no duplex) we need to load some initial buffers. In duplex mode we need to enqueu silent buffers. */ + if (pDevice->type == ma_device_type_duplex) { + MA_ZERO_MEMORY(pDevice->opensl.pBufferPlayback, pDevice->playback.internalPeriodSizeInFrames * pDevice->playback.internalPeriods * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels)); + } else { + ma_device__read_frames_from_client(pDevice, pDevice->playback.internalPeriodSizeInFrames * pDevice->playback.internalPeriods, pDevice->opensl.pBufferPlayback); + } + + periodSizeInBytes = pDevice->playback.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels); + for (iPeriod = 0; iPeriod < pDevice->playback.internalPeriods; ++iPeriod) { + resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback, pDevice->opensl.pBufferPlayback + (periodSizeInBytes * iPeriod), periodSizeInBytes); + if (resultSL != SL_RESULT_SUCCESS) { + MA_OPENSL_PLAY(pDevice->opensl.pAudioPlayer)->SetPlayState((SLPlayItf)pDevice->opensl.pAudioPlayer, SL_PLAYSTATE_STOPPED); + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to enqueue buffer for playback device.", MA_FAILED_TO_START_BACKEND_DEVICE); } - ma_pcm_rb_commit_write(&pDevice->webaudio.duplexRB, marginSizeInFrames, pMarginData); } } return MA_SUCCESS; } -ma_result ma_device_start__webaudio(ma_device* pDevice) +static ma_result ma_device_drain__opensl(ma_device* pDevice, ma_device_type deviceType) { - ma_assert(pDevice != NULL); + SLAndroidSimpleBufferQueueItf pBufferQueue; - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - EM_ASM({ - miniaudio.get_device_by_index($0).webaudio.resume(); - }, pDevice->webaudio.indexCapture); + MA_ASSERT(deviceType == ma_device_type_capture || deviceType == ma_device_type_playback); + + if (pDevice->type == ma_device_type_capture) { + pBufferQueue = (SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture; + pDevice->opensl.isDrainingCapture = MA_TRUE; + } else { + pBufferQueue = (SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback; + pDevice->opensl.isDrainingPlayback = MA_TRUE; } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - EM_ASM({ - miniaudio.get_device_by_index($0).webaudio.resume(); - }, pDevice->webaudio.indexPlayback); + for (;;) { + SLAndroidSimpleBufferQueueState state; + + MA_OPENSL_BUFFERQUEUE(pBufferQueue)->GetState(pBufferQueue, &state); + if (state.count == 0) { + break; + } + + ma_sleep(10); + } + + if (pDevice->type == ma_device_type_capture) { + pDevice->opensl.isDrainingCapture = MA_FALSE; + } else { + pDevice->opensl.isDrainingPlayback = MA_FALSE; } return MA_SUCCESS; } -ma_result ma_device_stop__webaudio(ma_device* pDevice) +static ma_result ma_device_stop__opensl(ma_device* pDevice) { - ma_assert(pDevice != NULL); + SLresult resultSL; + ma_stop_proc onStop; + + MA_ASSERT(pDevice != NULL); + + MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it before stopping/uninitializing the device. */ + if (g_maOpenSLInitCounter == 0) { + return MA_INVALID_OPERATION; + } if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - EM_ASM({ - miniaudio.get_device_by_index($0).webaudio.suspend(); - }, pDevice->webaudio.indexCapture); + ma_device_drain__opensl(pDevice, ma_device_type_capture); + + resultSL = MA_OPENSL_RECORD(pDevice->opensl.pAudioRecorder)->SetRecordState((SLRecordItf)pDevice->opensl.pAudioRecorder, SL_RECORDSTATE_STOPPED); + if (resultSL != SL_RESULT_SUCCESS) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to stop internal capture device.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + } + + MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->Clear((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture); } if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - EM_ASM({ - miniaudio.get_device_by_index($0).webaudio.suspend(); - }, pDevice->webaudio.indexPlayback); + ma_device_drain__opensl(pDevice, ma_device_type_playback); + + resultSL = MA_OPENSL_PLAY(pDevice->opensl.pAudioPlayer)->SetPlayState((SLPlayItf)pDevice->opensl.pAudioPlayer, SL_PLAYSTATE_STOPPED); + if (resultSL != SL_RESULT_SUCCESS) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to stop internal playback device.", MA_FAILED_TO_STOP_BACKEND_DEVICE); + } + + MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->Clear((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback); } - ma_stop_proc onStop = pDevice->onStop; + /* Make sure the client is aware that the device has stopped. There may be an OpenSL|ES callback for this, but I haven't found it. */ + onStop = pDevice->onStop; if (onStop) { onStop(pDevice); } @@ -25370,1075 +27035,1088 @@ ma_result ma_device_stop__webaudio(ma_device* pDevice) return MA_SUCCESS; } -ma_result ma_context_uninit__webaudio(ma_context* pContext) -{ - ma_assert(pContext != NULL); - ma_assert(pContext->backend == ma_backend_webaudio); - /* Nothing needs to be done here. */ +static ma_result ma_context_uninit__opensl(ma_context* pContext) +{ + MA_ASSERT(pContext != NULL); + MA_ASSERT(pContext->backend == ma_backend_opensl); (void)pContext; + /* Uninit global data. */ + if (g_maOpenSLInitCounter > 0) { + if (ma_atomic_decrement_32(&g_maOpenSLInitCounter) == 0) { + (*g_maEngineObjectSL)->Destroy(g_maEngineObjectSL); + } + } + return MA_SUCCESS; } -ma_result ma_context_init__webaudio(const ma_context_config* pConfig, ma_context* pContext) +static ma_result ma_context_init__opensl(const ma_context_config* pConfig, ma_context* pContext) { - int resultFromJS; - - ma_assert(pContext != NULL); + MA_ASSERT(pContext != NULL); - /* Here is where our global JavaScript object is initialized. */ - resultFromJS = EM_ASM_INT({ - if ((window.AudioContext || window.webkitAudioContext) === undefined) { - return 0; /* Web Audio not supported. */ - } + (void)pConfig; - if (typeof(miniaudio) === 'undefined') { - miniaudio = {}; - miniaudio.devices = []; /* Device cache for mapping devices to indexes for JavaScript/C interop. */ - - miniaudio.track_device = function(device) { - /* Try inserting into a free slot first. */ - for (var iDevice = 0; iDevice < miniaudio.devices.length; ++iDevice) { - if (miniaudio.devices[iDevice] == null) { - miniaudio.devices[iDevice] = device; - return iDevice; - } - } - - /* Getting here means there is no empty slots in the array so we just push to the end. */ - miniaudio.devices.push(device); - return miniaudio.devices.length - 1; - }; - - miniaudio.untrack_device_by_index = function(deviceIndex) { - /* We just set the device's slot to null. The slot will get reused in the next call to ma_track_device. */ - miniaudio.devices[deviceIndex] = null; - - /* Trim the array if possible. */ - while (miniaudio.devices.length > 0) { - if (miniaudio.devices[miniaudio.devices.length-1] == null) { - miniaudio.devices.pop(); - } else { - break; - } - } - }; - - miniaudio.untrack_device = function(device) { - for (var iDevice = 0; iDevice < miniaudio.devices.length; ++iDevice) { - if (miniaudio.devices[iDevice] == device) { - return miniaudio.untrack_device_by_index(iDevice); - } - } - }; - - miniaudio.get_device_by_index = function(deviceIndex) { - return miniaudio.devices[deviceIndex]; - }; + /* Initialize global data first if applicable. */ + if (ma_atomic_increment_32(&g_maOpenSLInitCounter) == 1) { + SLresult resultSL = slCreateEngine(&g_maEngineObjectSL, 0, NULL, 0, NULL, NULL); + if (resultSL != SL_RESULT_SUCCESS) { + ma_atomic_decrement_32(&g_maOpenSLInitCounter); + return MA_NO_BACKEND; } - - return 1; - }, 0); /* Must pass in a dummy argument for C99 compatibility. */ - if (resultFromJS != 1) { - return MA_FAILED_TO_INIT_BACKEND; - } + (*g_maEngineObjectSL)->Realize(g_maEngineObjectSL, SL_BOOLEAN_FALSE); + resultSL = (*g_maEngineObjectSL)->GetInterface(g_maEngineObjectSL, SL_IID_ENGINE, &g_maEngineSL); + if (resultSL != SL_RESULT_SUCCESS) { + (*g_maEngineObjectSL)->Destroy(g_maEngineObjectSL); + ma_atomic_decrement_32(&g_maOpenSLInitCounter); + return MA_NO_BACKEND; + } + } pContext->isBackendAsynchronous = MA_TRUE; - pContext->onUninit = ma_context_uninit__webaudio; - pContext->onDeviceIDEqual = ma_context_is_device_id_equal__webaudio; - pContext->onEnumDevices = ma_context_enumerate_devices__webaudio; - pContext->onGetDeviceInfo = ma_context_get_device_info__webaudio; - pContext->onDeviceInit = ma_device_init__webaudio; - pContext->onDeviceUninit = ma_device_uninit__webaudio; - pContext->onDeviceStart = ma_device_start__webaudio; - pContext->onDeviceStop = ma_device_stop__webaudio; + pContext->onUninit = ma_context_uninit__opensl; + pContext->onDeviceIDEqual = ma_context_is_device_id_equal__opensl; + pContext->onEnumDevices = ma_context_enumerate_devices__opensl; + pContext->onGetDeviceInfo = ma_context_get_device_info__opensl; + pContext->onDeviceInit = ma_device_init__opensl; + pContext->onDeviceUninit = ma_device_uninit__opensl; + pContext->onDeviceStart = ma_device_start__opensl; + pContext->onDeviceStop = ma_device_stop__opensl; - (void)pConfig; /* Unused. */ return MA_SUCCESS; } -#endif /* Web Audio */ +#endif /* OpenSL|ES */ +/****************************************************************************** -ma_bool32 ma__is_channel_map_valid(const ma_channel* channelMap, ma_uint32 channels) -{ - /* A blank channel map should be allowed, in which case it should use an appropriate default which will depend on context. */ - if (channelMap[0] != MA_CHANNEL_NONE) { - ma_uint32 iChannel; +Web Audio Backend - if (channels == 0) { - return MA_FALSE; /* No channels. */ - } +******************************************************************************/ +#ifdef MA_HAS_WEBAUDIO +#include - /* A channel cannot be present in the channel map more than once. */ - for (iChannel = 0; iChannel < channels; ++iChannel) { - ma_uint32 jChannel; - for (jChannel = iChannel + 1; jChannel < channels; ++jChannel) { - if (channelMap[iChannel] == channelMap[jChannel]) { - return MA_FALSE; - } - } - } - } +static ma_bool32 ma_is_capture_supported__webaudio() +{ + return EM_ASM_INT({ + return (navigator.mediaDevices !== undefined && navigator.mediaDevices.getUserMedia !== undefined); + }, 0) != 0; /* Must pass in a dummy argument for C99 compatibility. */ +} - return MA_TRUE; +#ifdef __cplusplus +extern "C" { +#endif +EMSCRIPTEN_KEEPALIVE void ma_device_process_pcm_frames_capture__webaudio(ma_device* pDevice, int frameCount, float* pFrames) +{ + if (pDevice->type == ma_device_type_duplex) { + ma_device__handle_duplex_callback_capture(pDevice, (ma_uint32)frameCount, pFrames, &pDevice->webaudio.duplexRB); + } else { + ma_device__send_frames_to_client(pDevice, (ma_uint32)frameCount, pFrames); /* Send directly to the client. */ + } } +EMSCRIPTEN_KEEPALIVE void ma_device_process_pcm_frames_playback__webaudio(ma_device* pDevice, int frameCount, float* pFrames) +{ + if (pDevice->type == ma_device_type_duplex) { + ma_device__handle_duplex_callback_playback(pDevice, (ma_uint32)frameCount, pFrames, &pDevice->webaudio.duplexRB); + } else { + ma_device__read_frames_from_client(pDevice, (ma_uint32)frameCount, pFrames); /* Read directly from the device. */ + } +} +#ifdef __cplusplus +} +#endif -void ma_device__post_init_setup(ma_device* pDevice, ma_device_type deviceType) +static ma_bool32 ma_context_is_device_id_equal__webaudio(ma_context* pContext, const ma_device_id* pID0, const ma_device_id* pID1) { - ma_assert(pDevice != NULL); + MA_ASSERT(pContext != NULL); + MA_ASSERT(pID0 != NULL); + MA_ASSERT(pID1 != NULL); + (void)pContext; - if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex) { - if (pDevice->capture.usingDefaultFormat) { - pDevice->capture.format = pDevice->capture.internalFormat; - } - if (pDevice->capture.usingDefaultChannels) { - pDevice->capture.channels = pDevice->capture.internalChannels; - } - if (pDevice->capture.usingDefaultChannelMap) { - if (pDevice->capture.internalChannels == pDevice->capture.channels) { - ma_channel_map_copy(pDevice->capture.channelMap, pDevice->capture.internalChannelMap, pDevice->capture.channels); - } else { - ma_get_standard_channel_map(ma_standard_channel_map_default, pDevice->capture.channels, pDevice->capture.channelMap); - } - } - } + return ma_strcmp(pID0->webaudio, pID1->webaudio) == 0; +} - if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) { - if (pDevice->playback.usingDefaultFormat) { - pDevice->playback.format = pDevice->playback.internalFormat; - } - if (pDevice->playback.usingDefaultChannels) { - pDevice->playback.channels = pDevice->playback.internalChannels; - } - if (pDevice->playback.usingDefaultChannelMap) { - if (pDevice->playback.internalChannels == pDevice->playback.channels) { - ma_channel_map_copy(pDevice->playback.channelMap, pDevice->playback.internalChannelMap, pDevice->playback.channels); - } else { - ma_get_standard_channel_map(ma_standard_channel_map_default, pDevice->playback.channels, pDevice->playback.channelMap); - } - } - } +static ma_result ma_context_enumerate_devices__webaudio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) +{ + ma_bool32 cbResult = MA_TRUE; - if (pDevice->usingDefaultSampleRate) { - if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex) { - pDevice->sampleRate = pDevice->capture.internalSampleRate; - } else { - pDevice->sampleRate = pDevice->playback.internalSampleRate; - } - } + MA_ASSERT(pContext != NULL); + MA_ASSERT(callback != NULL); - /* PCM converters. */ - if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) { - /* Converting from internal device format to public format. */ - ma_pcm_converter_config converterConfig = ma_pcm_converter_config_init_new(); - converterConfig.neverConsumeEndOfInput = MA_TRUE; - converterConfig.pUserData = pDevice; - converterConfig.formatIn = pDevice->capture.internalFormat; - converterConfig.channelsIn = pDevice->capture.internalChannels; - converterConfig.sampleRateIn = pDevice->capture.internalSampleRate; - ma_channel_map_copy(converterConfig.channelMapIn, pDevice->capture.internalChannelMap, pDevice->capture.internalChannels); - converterConfig.formatOut = pDevice->capture.format; - converterConfig.channelsOut = pDevice->capture.channels; - converterConfig.sampleRateOut = pDevice->sampleRate; - ma_channel_map_copy(converterConfig.channelMapOut, pDevice->capture.channelMap, pDevice->capture.channels); - converterConfig.onRead = ma_device__pcm_converter__on_read_from_buffer_capture; - ma_pcm_converter_init(&converterConfig, &pDevice->capture.converter); + /* Only supporting default devices for now. */ + + /* Playback. */ + if (cbResult) { + ma_device_info deviceInfo; + MA_ZERO_OBJECT(&deviceInfo); + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); + cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData); } - if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) { - /* Converting from public format to device format. */ - ma_pcm_converter_config converterConfig = ma_pcm_converter_config_init_new(); - converterConfig.neverConsumeEndOfInput = MA_TRUE; - converterConfig.pUserData = pDevice; - converterConfig.formatIn = pDevice->playback.format; - converterConfig.channelsIn = pDevice->playback.channels; - converterConfig.sampleRateIn = pDevice->sampleRate; - ma_channel_map_copy(converterConfig.channelMapIn, pDevice->playback.channelMap, pDevice->playback.channels); - converterConfig.formatOut = pDevice->playback.internalFormat; - converterConfig.channelsOut = pDevice->playback.internalChannels; - converterConfig.sampleRateOut = pDevice->playback.internalSampleRate; - ma_channel_map_copy(converterConfig.channelMapOut, pDevice->playback.internalChannelMap, pDevice->playback.internalChannels); - if (deviceType == ma_device_type_playback) { - if (pDevice->type == ma_device_type_playback) { - converterConfig.onRead = ma_device__on_read_from_client; - } else { - converterConfig.onRead = ma_device__pcm_converter__on_read_from_buffer_playback; - } - } else { - converterConfig.onRead = ma_device__pcm_converter__on_read_from_buffer_playback; + /* Capture. */ + if (cbResult) { + if (ma_is_capture_supported__webaudio()) { + ma_device_info deviceInfo; + MA_ZERO_OBJECT(&deviceInfo); + ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); + cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData); } - ma_pcm_converter_init(&converterConfig, &pDevice->playback.converter); } -} + return MA_SUCCESS; +} -ma_thread_result MA_THREADCALL ma_worker_thread(void* pData) +static ma_result ma_context_get_device_info__webaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) { - ma_device* pDevice = (ma_device*)pData; - ma_assert(pDevice != NULL); + MA_ASSERT(pContext != NULL); -#ifdef MA_WIN32 - ma_CoInitializeEx(pDevice->pContext, NULL, MA_COINIT_VALUE); -#endif + /* No exclusive mode with Web Audio. */ + if (shareMode == ma_share_mode_exclusive) { + return MA_SHARE_MODE_NOT_SUPPORTED; + } - /* - When the device is being initialized it's initial state is set to MA_STATE_UNINITIALIZED. Before returning from - ma_device_init(), the state needs to be set to something valid. In miniaudio the device's default state immediately - after initialization is stopped, so therefore we need to mark the device as such. miniaudio will wait on the worker - thread to signal an event to know when the worker thread is ready for action. - */ - ma_device__set_state(pDevice, MA_STATE_STOPPED); - ma_event_signal(&pDevice->stopEvent); + if (deviceType == ma_device_type_capture && !ma_is_capture_supported__webaudio()) { + return MA_NO_DEVICE; + } - for (;;) { /* <-- This loop just keeps the thread alive. The main audio loop is inside. */ - ma_stop_proc onStop; - /* We wait on an event to know when something has requested that the device be started and the main loop entered. */ - ma_event_wait(&pDevice->wakeupEvent); + MA_ZERO_MEMORY(pDeviceInfo->id.webaudio, sizeof(pDeviceInfo->id.webaudio)); - /* Default result code. */ - pDevice->workResult = MA_SUCCESS; + /* Only supporting default devices for now. */ + if (deviceType == ma_device_type_playback) { + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); + } else { + ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); + } - /* If the reason for the wake up is that we are terminating, just break from the loop. */ - if (ma_device__get_state(pDevice) == MA_STATE_UNINITIALIZED) { - break; + /* Web Audio can support any number of channels and sample rates. It only supports f32 formats, however. */ + pDeviceInfo->minChannels = 1; + pDeviceInfo->maxChannels = MA_MAX_CHANNELS; + if (pDeviceInfo->maxChannels > 32) { + pDeviceInfo->maxChannels = 32; /* Maximum output channel count is 32 for createScriptProcessor() (JavaScript). */ + } + + /* We can query the sample rate by just using a temporary audio context. */ + pDeviceInfo->minSampleRate = EM_ASM_INT({ + try { + var temp = new (window.AudioContext || window.webkitAudioContext)(); + var sampleRate = temp.sampleRate; + temp.close(); + return sampleRate; + } catch(e) { + return 0; } + }, 0); /* Must pass in a dummy argument for C99 compatibility. */ + pDeviceInfo->maxSampleRate = pDeviceInfo->minSampleRate; + if (pDeviceInfo->minSampleRate == 0) { + return MA_NO_DEVICE; + } - /* - Getting to this point means the device is wanting to get started. The function that has requested that the device - be started will be waiting on an event (pDevice->startEvent) which means we need to make sure we signal the event - in both the success and error case. It's important that the state of the device is set _before_ signaling the event. - */ - ma_assert(ma_device__get_state(pDevice) == MA_STATE_STARTING); + /* Web Audio only supports f32. */ + pDeviceInfo->formatCount = 1; + pDeviceInfo->formats[0] = ma_format_f32; - /* Make sure the state is set appropriately. */ - ma_device__set_state(pDevice, MA_STATE_STARTED); - ma_event_signal(&pDevice->startEvent); + return MA_SUCCESS; +} - if (pDevice->pContext->onDeviceMainLoop != NULL) { - pDevice->pContext->onDeviceMainLoop(pDevice); - } else { - ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "No main loop implementation.", MA_API_NOT_FOUND); - } - /* - Getting here means we have broken from the main loop which happens the application has requested that device be stopped. Note that this - may have actually already happened above if the device was lost and miniaudio has attempted to re-initialize the device. In this case we - don't want to be doing this a second time. - */ - if (ma_device__get_state(pDevice) != MA_STATE_UNINITIALIZED) { - if (pDevice->pContext->onDeviceStop) { - pDevice->pContext->onDeviceStop(pDevice); - } - } +static void ma_device_uninit_by_index__webaudio(ma_device* pDevice, ma_device_type deviceType, int deviceIndex) +{ + MA_ASSERT(pDevice != NULL); - /* After the device has stopped, make sure an event is posted. */ - onStop = pDevice->onStop; - if (onStop) { - onStop(pDevice); - } + EM_ASM({ + var device = miniaudio.get_device_by_index($0); - /* - A function somewhere is waiting for the device to have stopped for real so we need to signal an event to allow it to continue. Note that - it's possible that the device has been uninitialized which means we need to _not_ change the status to stopped. We cannot go from an - uninitialized state to stopped state. - */ - if (ma_device__get_state(pDevice) != MA_STATE_UNINITIALIZED) { - ma_device__set_state(pDevice, MA_STATE_STOPPED); - ma_event_signal(&pDevice->stopEvent); + /* Make sure all nodes are disconnected and marked for collection. */ + if (device.scriptNode !== undefined) { + device.scriptNode.onaudioprocess = function(e) {}; /* We want to reset the callback to ensure it doesn't get called after AudioContext.close() has returned. Shouldn't happen since we're disconnecting, but just to be safe... */ + device.scriptNode.disconnect(); + device.scriptNode = undefined; + } + if (device.streamNode !== undefined) { + device.streamNode.disconnect(); + device.streamNode = undefined; } - } - /* Make sure we aren't continuously waiting on a stop event. */ - ma_event_signal(&pDevice->stopEvent); /* <-- Is this still needed? */ + /* + Stop the device. I think there is a chance the callback could get fired after calling this, hence why we want + to clear the callback before closing. + */ + device.webaudio.close(); + device.webaudio = undefined; -#ifdef MA_WIN32 - ma_CoUninitialize(pDevice->pContext); -#endif + /* Can't forget to free the intermediary buffer. This is the buffer that's shared between JavaScript and C. */ + if (device.intermediaryBuffer !== undefined) { + Module._free(device.intermediaryBuffer); + device.intermediaryBuffer = undefined; + device.intermediaryBufferView = undefined; + device.intermediaryBufferSizeInBytes = undefined; + } - return (ma_thread_result)0; + /* Make sure the device is untracked so the slot can be reused later. */ + miniaudio.untrack_device_by_index($0); + }, deviceIndex, deviceType); } - -/* Helper for determining whether or not the given device is initialized. */ -ma_bool32 ma_device__is_initialized(ma_device* pDevice) +static void ma_device_uninit__webaudio(ma_device* pDevice) { - if (pDevice == NULL) { - return MA_FALSE; + MA_ASSERT(pDevice != NULL); + + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_capture, pDevice->webaudio.indexCapture); } - return ma_device__get_state(pDevice) != MA_STATE_UNINITIALIZED; -} + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_playback, pDevice->webaudio.indexPlayback); + } + if (pDevice->type == ma_device_type_duplex) { + ma_pcm_rb_uninit(&pDevice->webaudio.duplexRB); + } +} -#ifdef MA_WIN32 -ma_result ma_context_uninit_backend_apis__win32(ma_context* pContext) +static ma_result ma_device_init_by_type__webaudio(ma_context* pContext, const ma_device_config* pConfig, ma_device_type deviceType, ma_device* pDevice) { - ma_CoUninitialize(pContext); - ma_dlclose(pContext, pContext->win32.hUser32DLL); - ma_dlclose(pContext, pContext->win32.hOle32DLL); - ma_dlclose(pContext, pContext->win32.hAdvapi32DLL); + int deviceIndex; + ma_uint32 internalPeriodSizeInFrames; - return MA_SUCCESS; -} + MA_ASSERT(pContext != NULL); + MA_ASSERT(pConfig != NULL); + MA_ASSERT(deviceType != ma_device_type_duplex); + MA_ASSERT(pDevice != NULL); -ma_result ma_context_init_backend_apis__win32(ma_context* pContext) -{ -#ifdef MA_WIN32_DESKTOP - /* Ole32.dll */ - pContext->win32.hOle32DLL = ma_dlopen(pContext, "ole32.dll"); - if (pContext->win32.hOle32DLL == NULL) { - return MA_FAILED_TO_INIT_BACKEND; + if (deviceType == ma_device_type_capture && !ma_is_capture_supported__webaudio()) { + return MA_NO_DEVICE; } - pContext->win32.CoInitializeEx = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "CoInitializeEx"); - pContext->win32.CoUninitialize = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "CoUninitialize"); - pContext->win32.CoCreateInstance = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "CoCreateInstance"); - pContext->win32.CoTaskMemFree = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "CoTaskMemFree"); - pContext->win32.PropVariantClear = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "PropVariantClear"); - pContext->win32.StringFromGUID2 = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "StringFromGUID2"); - + /* Try calculating an appropriate buffer size. */ + internalPeriodSizeInFrames = pConfig->periodSizeInFrames; + if (internalPeriodSizeInFrames == 0) { + internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->periodSizeInMilliseconds, pConfig->sampleRate); + } - /* User32.dll */ - pContext->win32.hUser32DLL = ma_dlopen(pContext, "user32.dll"); - if (pContext->win32.hUser32DLL == NULL) { - return MA_FAILED_TO_INIT_BACKEND; + /* The size of the buffer must be a power of 2 and between 256 and 16384. */ + if (internalPeriodSizeInFrames < 256) { + internalPeriodSizeInFrames = 256; + } else if (internalPeriodSizeInFrames > 16384) { + internalPeriodSizeInFrames = 16384; + } else { + internalPeriodSizeInFrames = ma_next_power_of_2(internalPeriodSizeInFrames); } - pContext->win32.GetForegroundWindow = (ma_proc)ma_dlsym(pContext, pContext->win32.hUser32DLL, "GetForegroundWindow"); - pContext->win32.GetDesktopWindow = (ma_proc)ma_dlsym(pContext, pContext->win32.hUser32DLL, "GetDesktopWindow"); + /* We create the device on the JavaScript side and reference it using an index. We use this to make it possible to reference the device between JavaScript and C. */ + deviceIndex = EM_ASM_INT({ + var channels = $0; + var sampleRate = $1; + var bufferSize = $2; /* In PCM frames. */ + var isCapture = $3; + var pDevice = $4; + if (typeof(miniaudio) === 'undefined') { + return -1; /* Context not initialized. */ + } - /* Advapi32.dll */ - pContext->win32.hAdvapi32DLL = ma_dlopen(pContext, "advapi32.dll"); - if (pContext->win32.hAdvapi32DLL == NULL) { - return MA_FAILED_TO_INIT_BACKEND; - } + var device = {}; - pContext->win32.RegOpenKeyExA = (ma_proc)ma_dlsym(pContext, pContext->win32.hAdvapi32DLL, "RegOpenKeyExA"); - pContext->win32.RegCloseKey = (ma_proc)ma_dlsym(pContext, pContext->win32.hAdvapi32DLL, "RegCloseKey"); - pContext->win32.RegQueryValueExA = (ma_proc)ma_dlsym(pContext, pContext->win32.hAdvapi32DLL, "RegQueryValueExA"); -#endif + /* The AudioContext must be created in a suspended state. */ + device.webaudio = new (window.AudioContext || window.webkitAudioContext)({sampleRate:sampleRate}); + device.webaudio.suspend(); - ma_CoInitializeEx(pContext, NULL, MA_COINIT_VALUE); - return MA_SUCCESS; -} -#else -ma_result ma_context_uninit_backend_apis__nix(ma_context* pContext) -{ -#if defined(MA_USE_RUNTIME_LINKING_FOR_PTHREAD) && !defined(MA_NO_RUNTIME_LINKING) - ma_dlclose(pContext, pContext->posix.pthreadSO); -#else - (void)pContext; -#endif + /* + We need an intermediary buffer which we use for JavaScript and C interop. This buffer stores interleaved f32 PCM data. Because it's passed between + JavaScript and C it needs to be allocated and freed using Module._malloc() and Module._free(). + */ + device.intermediaryBufferSizeInBytes = channels * bufferSize * 4; + device.intermediaryBuffer = Module._malloc(device.intermediaryBufferSizeInBytes); + device.intermediaryBufferView = new Float32Array(Module.HEAPF32.buffer, device.intermediaryBuffer, device.intermediaryBufferSizeInBytes); - return MA_SUCCESS; -} + /* + Both playback and capture devices use a ScriptProcessorNode for performing per-sample operations. -ma_result ma_context_init_backend_apis__nix(ma_context* pContext) -{ - /* pthread */ -#if defined(MA_USE_RUNTIME_LINKING_FOR_PTHREAD) && !defined(MA_NO_RUNTIME_LINKING) - const char* libpthreadFileNames[] = { - "libpthread.so", - "libpthread.so.0", - "libpthread.dylib" - }; - size_t i; + ScriptProcessorNode is actually deprecated so this is likely to be temporary. The way this works for playback is very simple. You just set a callback + that's periodically fired, just like a normal audio callback function. But apparently this design is "flawed" and is now deprecated in favour of + something called AudioWorklets which _forces_ you to load a _separate_ .js file at run time... nice... Hopefully ScriptProcessorNode will continue to + work for years to come, but this may need to change to use AudioSourceBufferNode instead, which I think is what Emscripten uses for it's built-in SDL + implementation. I'll be avoiding that insane AudioWorklet API like the plague... - for (i = 0; i < sizeof(libpthreadFileNames) / sizeof(libpthreadFileNames[0]); ++i) { - pContext->posix.pthreadSO = ma_dlopen(pContext, libpthreadFileNames[i]); - if (pContext->posix.pthreadSO != NULL) { - break; - } - } + For capture it is a bit unintuitive. We use the ScriptProccessorNode _only_ to get the raw PCM data. It is connected to an AudioContext just like the + playback case, however we just output silence to the AudioContext instead of passing any real data. It would make more sense to me to use the + MediaRecorder API, but unfortunately you need to specify a MIME time (Opus, Vorbis, etc.) for the binary blob that's returned to the client, but I've + been unable to figure out how to get this as raw PCM. The closes I can think is to use the MIME type for WAV files and just parse it, but I don't know + how well this would work. Although ScriptProccessorNode is deprecated, in practice it seems to have pretty good browser support so I'm leaving it like + this for now. If anything knows how I could get raw PCM data using the MediaRecorder API please let me know! + */ + device.scriptNode = device.webaudio.createScriptProcessor(bufferSize, channels, channels); - if (pContext->posix.pthreadSO == NULL) { - return MA_FAILED_TO_INIT_BACKEND; - } + if (isCapture) { + device.scriptNode.onaudioprocess = function(e) { + if (device.intermediaryBuffer === undefined) { + return; /* This means the device has been uninitialized. */ + } - pContext->posix.pthread_create = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_create"); - pContext->posix.pthread_join = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_join"); - pContext->posix.pthread_mutex_init = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_mutex_init"); - pContext->posix.pthread_mutex_destroy = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_mutex_destroy"); - pContext->posix.pthread_mutex_lock = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_mutex_lock"); - pContext->posix.pthread_mutex_unlock = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_mutex_unlock"); - pContext->posix.pthread_cond_init = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_cond_init"); - pContext->posix.pthread_cond_destroy = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_cond_destroy"); - pContext->posix.pthread_cond_wait = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_cond_wait"); - pContext->posix.pthread_cond_signal = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_cond_signal"); - pContext->posix.pthread_attr_init = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_init"); - pContext->posix.pthread_attr_destroy = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_destroy"); - pContext->posix.pthread_attr_setschedpolicy = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_setschedpolicy"); - pContext->posix.pthread_attr_getschedparam = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_getschedparam"); - pContext->posix.pthread_attr_setschedparam = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_setschedparam"); -#else - pContext->posix.pthread_create = (ma_proc)pthread_create; - pContext->posix.pthread_join = (ma_proc)pthread_join; - pContext->posix.pthread_mutex_init = (ma_proc)pthread_mutex_init; - pContext->posix.pthread_mutex_destroy = (ma_proc)pthread_mutex_destroy; - pContext->posix.pthread_mutex_lock = (ma_proc)pthread_mutex_lock; - pContext->posix.pthread_mutex_unlock = (ma_proc)pthread_mutex_unlock; - pContext->posix.pthread_cond_init = (ma_proc)pthread_cond_init; - pContext->posix.pthread_cond_destroy = (ma_proc)pthread_cond_destroy; - pContext->posix.pthread_cond_wait = (ma_proc)pthread_cond_wait; - pContext->posix.pthread_cond_signal = (ma_proc)pthread_cond_signal; - pContext->posix.pthread_attr_init = (ma_proc)pthread_attr_init; - pContext->posix.pthread_attr_destroy = (ma_proc)pthread_attr_destroy; -#if !defined(__EMSCRIPTEN__) - pContext->posix.pthread_attr_setschedpolicy = (ma_proc)pthread_attr_setschedpolicy; - pContext->posix.pthread_attr_getschedparam = (ma_proc)pthread_attr_getschedparam; - pContext->posix.pthread_attr_setschedparam = (ma_proc)pthread_attr_setschedparam; -#endif -#endif + /* Make sure silence it output to the AudioContext destination. Not doing this will cause sound to come out of the speakers! */ + for (var iChannel = 0; iChannel < e.outputBuffer.numberOfChannels; ++iChannel) { + e.outputBuffer.getChannelData(iChannel).fill(0.0); + } - return MA_SUCCESS; -} -#endif + /* There are some situations where we may want to send silence to the client. */ + var sendSilence = false; + if (device.streamNode === undefined) { + sendSilence = true; + } -ma_result ma_context_init_backend_apis(ma_context* pContext) -{ - ma_result result; -#ifdef MA_WIN32 - result = ma_context_init_backend_apis__win32(pContext); -#else - result = ma_context_init_backend_apis__nix(pContext); -#endif + /* Sanity check. This will never happen, right? */ + if (e.inputBuffer.numberOfChannels != channels) { + console.log("Capture: Channel count mismatch. " + e.inputBufer.numberOfChannels + " != " + channels + ". Sending silence."); + sendSilence = true; + } + + /* This looped design guards against the situation where e.inputBuffer is a different size to the original buffer size. Should never happen in practice. */ + var totalFramesProcessed = 0; + while (totalFramesProcessed < e.inputBuffer.length) { + var framesRemaining = e.inputBuffer.length - totalFramesProcessed; + var framesToProcess = framesRemaining; + if (framesToProcess > (device.intermediaryBufferSizeInBytes/channels/4)) { + framesToProcess = (device.intermediaryBufferSizeInBytes/channels/4); + } + + /* We need to do the reverse of the playback case. We need to interleave the input data and copy it into the intermediary buffer. Then we send it to the client. */ + if (sendSilence) { + device.intermediaryBufferView.fill(0.0); + } else { + for (var iFrame = 0; iFrame < framesToProcess; ++iFrame) { + for (var iChannel = 0; iChannel < e.inputBuffer.numberOfChannels; ++iChannel) { + device.intermediaryBufferView[iFrame*channels + iChannel] = e.inputBuffer.getChannelData(iChannel)[totalFramesProcessed + iFrame]; + } + } + } + + /* Send data to the client from our intermediary buffer. */ + ccall("ma_device_process_pcm_frames_capture__webaudio", "undefined", ["number", "number", "number"], [pDevice, framesToProcess, device.intermediaryBuffer]); + + totalFramesProcessed += framesToProcess; + } + }; + + navigator.mediaDevices.getUserMedia({audio:true, video:false}) + .then(function(stream) { + device.streamNode = device.webaudio.createMediaStreamSource(stream); + device.streamNode.connect(device.scriptNode); + device.scriptNode.connect(device.webaudio.destination); + }) + .catch(function(error) { + /* I think this should output silence... */ + device.scriptNode.connect(device.webaudio.destination); + }); + } else { + device.scriptNode.onaudioprocess = function(e) { + if (device.intermediaryBuffer === undefined) { + return; /* This means the device has been uninitialized. */ + } - return result; -} + var outputSilence = false; -ma_result ma_context_uninit_backend_apis(ma_context* pContext) -{ - ma_result result; -#ifdef MA_WIN32 - result = ma_context_uninit_backend_apis__win32(pContext); -#else - result = ma_context_uninit_backend_apis__nix(pContext); -#endif + /* Sanity check. This will never happen, right? */ + if (e.outputBuffer.numberOfChannels != channels) { + console.log("Playback: Channel count mismatch. " + e.outputBufer.numberOfChannels + " != " + channels + ". Outputting silence."); + outputSilence = true; + return; + } - return result; -} + /* This looped design guards against the situation where e.outputBuffer is a different size to the original buffer size. Should never happen in practice. */ + var totalFramesProcessed = 0; + while (totalFramesProcessed < e.outputBuffer.length) { + var framesRemaining = e.outputBuffer.length - totalFramesProcessed; + var framesToProcess = framesRemaining; + if (framesToProcess > (device.intermediaryBufferSizeInBytes/channels/4)) { + framesToProcess = (device.intermediaryBufferSizeInBytes/channels/4); + } + /* Read data from the client into our intermediary buffer. */ + ccall("ma_device_process_pcm_frames_playback__webaudio", "undefined", ["number", "number", "number"], [pDevice, framesToProcess, device.intermediaryBuffer]); -ma_bool32 ma_context_is_backend_asynchronous(ma_context* pContext) -{ - return pContext->isBackendAsynchronous; -} + /* At this point we'll have data in our intermediary buffer which we now need to deinterleave and copy over to the output buffers. */ + if (outputSilence) { + for (var iChannel = 0; iChannel < e.outputBuffer.numberOfChannels; ++iChannel) { + e.outputBuffer.getChannelData(iChannel).fill(0.0); + } + } else { + for (var iChannel = 0; iChannel < e.outputBuffer.numberOfChannels; ++iChannel) { + for (var iFrame = 0; iFrame < framesToProcess; ++iFrame) { + e.outputBuffer.getChannelData(iChannel)[totalFramesProcessed + iFrame] = device.intermediaryBufferView[iFrame*channels + iChannel]; + } + } + } -ma_result ma_context_init(const ma_backend backends[], ma_uint32 backendCount, const ma_context_config* pConfig, ma_context* pContext) -{ - ma_result result; - ma_context_config config; - ma_backend defaultBackends[ma_backend_null+1]; - ma_uint32 iBackend; - ma_backend* pBackendsToIterate; - ma_uint32 backendsToIterateCount; + totalFramesProcessed += framesToProcess; + } + }; - if (pContext == NULL) { - return MA_INVALID_ARGS; - } + device.scriptNode.connect(device.webaudio.destination); + } - ma_zero_object(pContext); + return miniaudio.track_device(device); + }, (deviceType == ma_device_type_capture) ? pConfig->capture.channels : pConfig->playback.channels, pConfig->sampleRate, internalPeriodSizeInFrames, deviceType == ma_device_type_capture, pDevice); - /* Always make sure the config is set first to ensure properties are available as soon as possible. */ - if (pConfig != NULL) { - config = *pConfig; + if (deviceIndex < 0) { + return MA_FAILED_TO_OPEN_BACKEND_DEVICE; + } + + if (deviceType == ma_device_type_capture) { + pDevice->webaudio.indexCapture = deviceIndex; + pDevice->capture.internalFormat = ma_format_f32; + pDevice->capture.internalChannels = pConfig->capture.channels; + ma_get_standard_channel_map(ma_standard_channel_map_webaudio, pDevice->capture.internalChannels, pDevice->capture.internalChannelMap); + pDevice->capture.internalSampleRate = EM_ASM_INT({ return miniaudio.get_device_by_index($0).webaudio.sampleRate; }, deviceIndex); + pDevice->capture.internalPeriodSizeInFrames = internalPeriodSizeInFrames; + pDevice->capture.internalPeriods = 1; } else { - config = ma_context_config_init(); + pDevice->webaudio.indexPlayback = deviceIndex; + pDevice->playback.internalFormat = ma_format_f32; + pDevice->playback.internalChannels = pConfig->playback.channels; + ma_get_standard_channel_map(ma_standard_channel_map_webaudio, pDevice->playback.internalChannels, pDevice->playback.internalChannelMap); + pDevice->playback.internalSampleRate = EM_ASM_INT({ return miniaudio.get_device_by_index($0).webaudio.sampleRate; }, deviceIndex); + pDevice->playback.internalPeriodSizeInFrames = internalPeriodSizeInFrames; + pDevice->playback.internalPeriods = 1; } - pContext->logCallback = config.logCallback; - pContext->threadPriority = config.threadPriority; - pContext->pUserData = config.pUserData; + return MA_SUCCESS; +} - /* Backend APIs need to be initialized first. This is where external libraries will be loaded and linked. */ - result = ma_context_init_backend_apis(pContext); - if (result != MA_SUCCESS) { - return result; +static ma_result ma_device_init__webaudio(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) +{ + ma_result result; + + if (pConfig->deviceType == ma_device_type_loopback) { + return MA_DEVICE_TYPE_NOT_SUPPORTED; } - for (iBackend = 0; iBackend <= ma_backend_null; ++iBackend) { - defaultBackends[iBackend] = (ma_backend)iBackend; + /* No exclusive mode with Web Audio. */ + if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) || + ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) { + return MA_SHARE_MODE_NOT_SUPPORTED; } - pBackendsToIterate = (ma_backend*)backends; - backendsToIterateCount = backendCount; - if (pBackendsToIterate == NULL) { - pBackendsToIterate = (ma_backend*)defaultBackends; - backendsToIterateCount = ma_countof(defaultBackends); + if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) { + result = ma_device_init_by_type__webaudio(pContext, pConfig, ma_device_type_capture, pDevice); + if (result != MA_SUCCESS) { + return result; + } } - ma_assert(pBackendsToIterate != NULL); + if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) { + result = ma_device_init_by_type__webaudio(pContext, pConfig, ma_device_type_playback, pDevice); + if (result != MA_SUCCESS) { + if (pConfig->deviceType == ma_device_type_duplex) { + ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_capture, pDevice->webaudio.indexCapture); + } + return result; + } + } - for (iBackend = 0; iBackend < backendsToIterateCount; ++iBackend) { - ma_backend backend = pBackendsToIterate[iBackend]; + /* + We need a ring buffer for moving data from the capture device to the playback device. The capture callback is the producer + and the playback callback is the consumer. The buffer needs to be large enough to hold internalPeriodSizeInFrames based on + the external sample rate. + */ + if (pConfig->deviceType == ma_device_type_duplex) { + ma_uint32 rbSizeInFrames = (ma_uint32)ma_calculate_frame_count_after_resampling(pDevice->sampleRate, pDevice->capture.internalSampleRate, pDevice->capture.internalPeriodSizeInFrames) * 2; + result = ma_pcm_rb_init(pDevice->capture.format, pDevice->capture.channels, rbSizeInFrames, NULL, &pDevice->pContext->allocationCallbacks, &pDevice->webaudio.duplexRB); + if (result != MA_SUCCESS) { + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_capture, pDevice->webaudio.indexCapture); + } + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_playback, pDevice->webaudio.indexPlayback); + } + return result; + } - result = MA_NO_BACKEND; - switch (backend) { - #ifdef MA_HAS_WASAPI - case ma_backend_wasapi: - { - result = ma_context_init__wasapi(&config, pContext); - } break; - #endif - #ifdef MA_HAS_DSOUND - case ma_backend_dsound: - { - result = ma_context_init__dsound(&config, pContext); - } break; - #endif - #ifdef MA_HAS_WINMM - case ma_backend_winmm: - { - result = ma_context_init__winmm(&config, pContext); - } break; - #endif - #ifdef MA_HAS_ALSA - case ma_backend_alsa: - { - result = ma_context_init__alsa(&config, pContext); - } break; - #endif - #ifdef MA_HAS_PULSEAUDIO - case ma_backend_pulseaudio: - { - result = ma_context_init__pulse(&config, pContext); - } break; - #endif - #ifdef MA_HAS_JACK - case ma_backend_jack: - { - result = ma_context_init__jack(&config, pContext); - } break; - #endif - #ifdef MA_HAS_COREAUDIO - case ma_backend_coreaudio: - { - result = ma_context_init__coreaudio(&config, pContext); - } break; - #endif - #ifdef MA_HAS_SNDIO - case ma_backend_sndio: - { - result = ma_context_init__sndio(&config, pContext); - } break; - #endif - #ifdef MA_HAS_AUDIO4 - case ma_backend_audio4: - { - result = ma_context_init__audio4(&config, pContext); - } break; - #endif - #ifdef MA_HAS_OSS - case ma_backend_oss: - { - result = ma_context_init__oss(&config, pContext); - } break; - #endif - #ifdef MA_HAS_AAUDIO - case ma_backend_aaudio: - { - result = ma_context_init__aaudio(&config, pContext); - } break; - #endif - #ifdef MA_HAS_OPENSL - case ma_backend_opensl: - { - result = ma_context_init__opensl(&config, pContext); - } break; - #endif - #ifdef MA_HAS_WEBAUDIO - case ma_backend_webaudio: - { - result = ma_context_init__webaudio(&config, pContext); - } break; - #endif - #ifdef MA_HAS_NULL - case ma_backend_null: + /* We need a period to act as a buffer for cases where the playback and capture device's end up desyncing. */ + { + ma_uint32 marginSizeInFrames = rbSizeInFrames / 3; /* <-- Dividing by 3 because internalPeriods is always set to 1 for WebAudio. */ + void* pMarginData; + ma_pcm_rb_acquire_write(&pDevice->webaudio.duplexRB, &marginSizeInFrames, &pMarginData); { - result = ma_context_init__null(&config, pContext); - } break; - #endif - - default: break; + MA_ZERO_MEMORY(pMarginData, marginSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels)); + } + ma_pcm_rb_commit_write(&pDevice->webaudio.duplexRB, marginSizeInFrames, pMarginData); } + } - /* If this iteration was successful, return. */ - if (result == MA_SUCCESS) { - result = ma_mutex_init(pContext, &pContext->deviceEnumLock); - if (result != MA_SUCCESS) { - ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_WARNING, "Failed to initialize mutex for device enumeration. ma_context_get_devices() is not thread safe.", MA_FAILED_TO_CREATE_MUTEX); - } - result = ma_mutex_init(pContext, &pContext->deviceInfoLock); - if (result != MA_SUCCESS) { - ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_WARNING, "Failed to initialize mutex for device info retrieval. ma_context_get_device_info() is not thread safe.", MA_FAILED_TO_CREATE_MUTEX); - } + return MA_SUCCESS; +} -#ifdef MA_DEBUG_OUTPUT - printf("[miniaudio] Endian: %s\n", ma_is_little_endian() ? "LE" : "BE"); - printf("[miniaudio] SSE2: %s\n", ma_has_sse2() ? "YES" : "NO"); - printf("[miniaudio] AVX2: %s\n", ma_has_avx2() ? "YES" : "NO"); - printf("[miniaudio] AVX512F: %s\n", ma_has_avx512f() ? "YES" : "NO"); - printf("[miniaudio] NEON: %s\n", ma_has_neon() ? "YES" : "NO"); -#endif +static ma_result ma_device_start__webaudio(ma_device* pDevice) +{ + MA_ASSERT(pDevice != NULL); + + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + EM_ASM({ + miniaudio.get_device_by_index($0).webaudio.resume(); + }, pDevice->webaudio.indexCapture); + } - pContext->backend = backend; - return result; - } + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + EM_ASM({ + miniaudio.get_device_by_index($0).webaudio.resume(); + }, pDevice->webaudio.indexPlayback); } - /* If we get here it means an error occurred. */ - ma_zero_object(pContext); /* Safety. */ - return MA_NO_BACKEND; + return MA_SUCCESS; } -ma_result ma_context_uninit(ma_context* pContext) +static ma_result ma_device_stop__webaudio(ma_device* pDevice) { - if (pContext == NULL) { - return MA_INVALID_ARGS; - } - - pContext->onUninit(pContext); + MA_ASSERT(pDevice != NULL); - ma_mutex_uninit(&pContext->deviceEnumLock); - ma_mutex_uninit(&pContext->deviceInfoLock); - ma_free(pContext->pDeviceInfos); - ma_context_uninit_backend_apis(pContext); + /* + From the WebAudio API documentation for AudioContext.suspend(): - return MA_SUCCESS; -} + Suspends the progression of AudioContext's currentTime, allows any current context processing blocks that are already processed to be played to the + destination, and then allows the system to release its claim on audio hardware. + I read this to mean that "any current context processing blocks" are processed by suspend() - i.e. They they are drained. We therefore shouldn't need to + do any kind of explicit draining. + */ -ma_result ma_context_enumerate_devices(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) -{ - ma_result result; + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + EM_ASM({ + miniaudio.get_device_by_index($0).webaudio.suspend(); + }, pDevice->webaudio.indexCapture); + } - if (pContext == NULL || pContext->onEnumDevices == NULL || callback == NULL) { - return MA_INVALID_ARGS; + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + EM_ASM({ + miniaudio.get_device_by_index($0).webaudio.suspend(); + }, pDevice->webaudio.indexPlayback); } - ma_mutex_lock(&pContext->deviceEnumLock); - { - result = pContext->onEnumDevices(pContext, callback, pUserData); + ma_stop_proc onStop = pDevice->onStop; + if (onStop) { + onStop(pDevice); } - ma_mutex_unlock(&pContext->deviceEnumLock); - return result; + return MA_SUCCESS; } +static ma_result ma_context_uninit__webaudio(ma_context* pContext) +{ + MA_ASSERT(pContext != NULL); + MA_ASSERT(pContext->backend == ma_backend_webaudio); + + /* Nothing needs to be done here. */ + (void)pContext; + + return MA_SUCCESS; +} -ma_bool32 ma_context_get_devices__enum_callback(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pInfo, void* pUserData) +static ma_result ma_context_init__webaudio(const ma_context_config* pConfig, ma_context* pContext) { - /* - We need to insert the device info into our main internal buffer. Where it goes depends on the device type. If it's a capture device - it's just appended to the end. If it's a playback device it's inserted just before the first capture device. - */ + int resultFromJS; - /* - First make sure we have room. Since the number of devices we add to the list is usually relatively small I've decided to use a - simple fixed size increment for buffer expansion. - */ - const ma_uint32 bufferExpansionCount = 2; - const ma_uint32 totalDeviceInfoCount = pContext->playbackDeviceInfoCount + pContext->captureDeviceInfoCount; + MA_ASSERT(pContext != NULL); - if (pContext->deviceInfoCapacity >= totalDeviceInfoCount) { - ma_uint32 newCapacity = totalDeviceInfoCount + bufferExpansionCount; - ma_device_info* pNewInfos = (ma_device_info*)ma_realloc(pContext->pDeviceInfos, sizeof(*pContext->pDeviceInfos)*newCapacity); - if (pNewInfos == NULL) { - return MA_FALSE; /* Out of memory. */ + /* Here is where our global JavaScript object is initialized. */ + resultFromJS = EM_ASM_INT({ + if ((window.AudioContext || window.webkitAudioContext) === undefined) { + return 0; /* Web Audio not supported. */ } - pContext->pDeviceInfos = pNewInfos; - pContext->deviceInfoCapacity = newCapacity; + if (typeof(miniaudio) === 'undefined') { + miniaudio = {}; + miniaudio.devices = []; /* Device cache for mapping devices to indexes for JavaScript/C interop. */ + + miniaudio.track_device = function(device) { + /* Try inserting into a free slot first. */ + for (var iDevice = 0; iDevice < miniaudio.devices.length; ++iDevice) { + if (miniaudio.devices[iDevice] == null) { + miniaudio.devices[iDevice] = device; + return iDevice; + } + } + + /* Getting here means there is no empty slots in the array so we just push to the end. */ + miniaudio.devices.push(device); + return miniaudio.devices.length - 1; + }; + + miniaudio.untrack_device_by_index = function(deviceIndex) { + /* We just set the device's slot to null. The slot will get reused in the next call to ma_track_device. */ + miniaudio.devices[deviceIndex] = null; + + /* Trim the array if possible. */ + while (miniaudio.devices.length > 0) { + if (miniaudio.devices[miniaudio.devices.length-1] == null) { + miniaudio.devices.pop(); + } else { + break; + } + } + }; + + miniaudio.untrack_device = function(device) { + for (var iDevice = 0; iDevice < miniaudio.devices.length; ++iDevice) { + if (miniaudio.devices[iDevice] == device) { + return miniaudio.untrack_device_by_index(iDevice); + } + } + }; + + miniaudio.get_device_by_index = function(deviceIndex) { + return miniaudio.devices[deviceIndex]; + }; + } + + return 1; + }, 0); /* Must pass in a dummy argument for C99 compatibility. */ + + if (resultFromJS != 1) { + return MA_FAILED_TO_INIT_BACKEND; } - if (deviceType == ma_device_type_playback) { - /* Playback. Insert just before the first capture device. */ - /* The first thing to do is move all of the capture devices down a slot. */ - ma_uint32 iFirstCaptureDevice = pContext->playbackDeviceInfoCount; - size_t iCaptureDevice; - for (iCaptureDevice = totalDeviceInfoCount; iCaptureDevice > iFirstCaptureDevice; --iCaptureDevice) { - pContext->pDeviceInfos[iCaptureDevice] = pContext->pDeviceInfos[iCaptureDevice-1]; - } + pContext->isBackendAsynchronous = MA_TRUE; - /* Now just insert where the first capture device was before moving it down a slot. */ - pContext->pDeviceInfos[iFirstCaptureDevice] = *pInfo; - pContext->playbackDeviceInfoCount += 1; - } else { - /* Capture. Insert at the end. */ - pContext->pDeviceInfos[totalDeviceInfoCount] = *pInfo; - pContext->captureDeviceInfoCount += 1; - } + pContext->onUninit = ma_context_uninit__webaudio; + pContext->onDeviceIDEqual = ma_context_is_device_id_equal__webaudio; + pContext->onEnumDevices = ma_context_enumerate_devices__webaudio; + pContext->onGetDeviceInfo = ma_context_get_device_info__webaudio; + pContext->onDeviceInit = ma_device_init__webaudio; + pContext->onDeviceUninit = ma_device_uninit__webaudio; + pContext->onDeviceStart = ma_device_start__webaudio; + pContext->onDeviceStop = ma_device_stop__webaudio; - (void)pUserData; - return MA_TRUE; + (void)pConfig; /* Unused. */ + return MA_SUCCESS; } +#endif /* Web Audio */ -ma_result ma_context_get_devices(ma_context* pContext, ma_device_info** ppPlaybackDeviceInfos, ma_uint32* pPlaybackDeviceCount, ma_device_info** ppCaptureDeviceInfos, ma_uint32* pCaptureDeviceCount) -{ - ma_result result; - - /* Safety. */ - if (ppPlaybackDeviceInfos != NULL) *ppPlaybackDeviceInfos = NULL; - if (pPlaybackDeviceCount != NULL) *pPlaybackDeviceCount = 0; - if (ppCaptureDeviceInfos != NULL) *ppCaptureDeviceInfos = NULL; - if (pCaptureDeviceCount != NULL) *pCaptureDeviceCount = 0; - if (pContext == NULL || pContext->onEnumDevices == NULL) { - return MA_INVALID_ARGS; - } - /* Note that we don't use ma_context_enumerate_devices() here because we want to do locking at a higher level. */ - ma_mutex_lock(&pContext->deviceEnumLock); - { - /* Reset everything first. */ - pContext->playbackDeviceInfoCount = 0; - pContext->captureDeviceInfoCount = 0; +static ma_bool32 ma__is_channel_map_valid(const ma_channel* channelMap, ma_uint32 channels) +{ + /* A blank channel map should be allowed, in which case it should use an appropriate default which will depend on context. */ + if (channelMap[0] != MA_CHANNEL_NONE) { + ma_uint32 iChannel; - /* Now enumerate over available devices. */ - result = pContext->onEnumDevices(pContext, ma_context_get_devices__enum_callback, NULL); - if (result == MA_SUCCESS) { - /* Playback devices. */ - if (ppPlaybackDeviceInfos != NULL) { - *ppPlaybackDeviceInfos = pContext->pDeviceInfos; - } - if (pPlaybackDeviceCount != NULL) { - *pPlaybackDeviceCount = pContext->playbackDeviceInfoCount; - } + if (channels == 0) { + return MA_FALSE; /* No channels. */ + } - /* Capture devices. */ - if (ppCaptureDeviceInfos != NULL) { - *ppCaptureDeviceInfos = pContext->pDeviceInfos + pContext->playbackDeviceInfoCount; /* Capture devices come after playback devices. */ - } - if (pCaptureDeviceCount != NULL) { - *pCaptureDeviceCount = pContext->captureDeviceInfoCount; + /* A channel cannot be present in the channel map more than once. */ + for (iChannel = 0; iChannel < channels; ++iChannel) { + ma_uint32 jChannel; + for (jChannel = iChannel + 1; jChannel < channels; ++jChannel) { + if (channelMap[iChannel] == channelMap[jChannel]) { + return MA_FALSE; + } } } } - ma_mutex_unlock(&pContext->deviceEnumLock); - return result; + return MA_TRUE; } -ma_result ma_context_get_device_info(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) + +static ma_result ma_device__post_init_setup(ma_device* pDevice, ma_device_type deviceType) { - ma_device_info deviceInfo; + ma_result result; - /* NOTE: Do not clear pDeviceInfo on entry. The reason is the pDeviceID may actually point to pDeviceInfo->id which will break things. */ - if (pContext == NULL || pDeviceInfo == NULL) { - return MA_INVALID_ARGS; - } + MA_ASSERT(pDevice != NULL); - ma_zero_object(&deviceInfo); + if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex) { + if (pDevice->capture.usingDefaultFormat) { + pDevice->capture.format = pDevice->capture.internalFormat; + } + if (pDevice->capture.usingDefaultChannels) { + pDevice->capture.channels = pDevice->capture.internalChannels; + } + if (pDevice->capture.usingDefaultChannelMap) { + if (pDevice->capture.internalChannels == pDevice->capture.channels) { + ma_channel_map_copy(pDevice->capture.channelMap, pDevice->capture.internalChannelMap, pDevice->capture.channels); + } else { + ma_get_standard_channel_map(ma_standard_channel_map_default, pDevice->capture.channels, pDevice->capture.channelMap); + } + } + } - /* Help the backend out by copying over the device ID if we have one. */ - if (pDeviceID != NULL) { - ma_copy_memory(&deviceInfo.id, pDeviceID, sizeof(*pDeviceID)); + if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) { + if (pDevice->playback.usingDefaultFormat) { + pDevice->playback.format = pDevice->playback.internalFormat; + } + if (pDevice->playback.usingDefaultChannels) { + pDevice->playback.channels = pDevice->playback.internalChannels; + } + if (pDevice->playback.usingDefaultChannelMap) { + if (pDevice->playback.internalChannels == pDevice->playback.channels) { + ma_channel_map_copy(pDevice->playback.channelMap, pDevice->playback.internalChannelMap, pDevice->playback.channels); + } else { + ma_get_standard_channel_map(ma_standard_channel_map_default, pDevice->playback.channels, pDevice->playback.channelMap); + } + } } - /* The backend may have an optimized device info retrieval function. If so, try that first. */ - if (pContext->onGetDeviceInfo != NULL) { - ma_result result; - ma_mutex_lock(&pContext->deviceInfoLock); - { - result = pContext->onGetDeviceInfo(pContext, deviceType, pDeviceID, shareMode, &deviceInfo); + if (pDevice->usingDefaultSampleRate) { + if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex) { + pDevice->sampleRate = pDevice->capture.internalSampleRate; + } else { + pDevice->sampleRate = pDevice->playback.internalSampleRate; } - ma_mutex_unlock(&pContext->deviceInfoLock); + } - /* Clamp ranges. */ - deviceInfo.minChannels = ma_max(deviceInfo.minChannels, MA_MIN_CHANNELS); - deviceInfo.maxChannels = ma_min(deviceInfo.maxChannels, MA_MAX_CHANNELS); - deviceInfo.minSampleRate = ma_max(deviceInfo.minSampleRate, MA_MIN_SAMPLE_RATE); - deviceInfo.maxSampleRate = ma_min(deviceInfo.maxSampleRate, MA_MAX_SAMPLE_RATE); + /* PCM converters. */ + if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) { + /* Converting from internal device format to client format. */ + ma_data_converter_config converterConfig = ma_data_converter_config_init_default(); + converterConfig.formatIn = pDevice->capture.internalFormat; + converterConfig.channelsIn = pDevice->capture.internalChannels; + converterConfig.sampleRateIn = pDevice->capture.internalSampleRate; + ma_channel_map_copy(converterConfig.channelMapIn, pDevice->capture.internalChannelMap, pDevice->capture.internalChannels); + converterConfig.formatOut = pDevice->capture.format; + converterConfig.channelsOut = pDevice->capture.channels; + converterConfig.sampleRateOut = pDevice->sampleRate; + ma_channel_map_copy(converterConfig.channelMapOut, pDevice->capture.channelMap, pDevice->capture.channels); + converterConfig.resampling.allowDynamicSampleRate = MA_FALSE; + converterConfig.resampling.algorithm = pDevice->resampling.algorithm; + converterConfig.resampling.linear.lpfCount = pDevice->resampling.linear.lpfCount; + converterConfig.resampling.speex.quality = pDevice->resampling.speex.quality; - *pDeviceInfo = deviceInfo; - return result; + result = ma_data_converter_init(&converterConfig, &pDevice->capture.converter); + if (result != MA_SUCCESS) { + return result; + } } - /* Getting here means onGetDeviceInfo has not been set. */ - return MA_ERROR; -} + if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) { + /* Converting from client format to device format. */ + ma_data_converter_config converterConfig = ma_data_converter_config_init_default(); + converterConfig.formatIn = pDevice->playback.format; + converterConfig.channelsIn = pDevice->playback.channels; + converterConfig.sampleRateIn = pDevice->sampleRate; + ma_channel_map_copy(converterConfig.channelMapIn, pDevice->playback.channelMap, pDevice->playback.channels); + converterConfig.formatOut = pDevice->playback.internalFormat; + converterConfig.channelsOut = pDevice->playback.internalChannels; + converterConfig.sampleRateOut = pDevice->playback.internalSampleRate; + ma_channel_map_copy(converterConfig.channelMapOut, pDevice->playback.internalChannelMap, pDevice->playback.internalChannels); + converterConfig.resampling.allowDynamicSampleRate = MA_FALSE; + converterConfig.resampling.algorithm = pDevice->resampling.algorithm; + converterConfig.resampling.linear.lpfCount = pDevice->resampling.linear.lpfCount; + converterConfig.resampling.speex.quality = pDevice->resampling.speex.quality; -ma_bool32 ma_context_is_loopback_supported(ma_context* pContext) -{ - if (pContext == NULL) { - return MA_FALSE; + result = ma_data_converter_init(&converterConfig, &pDevice->playback.converter); + if (result != MA_SUCCESS) { + return result; + } } - return ma_is_loopback_supported(pContext->backend); + return MA_SUCCESS; } -ma_result ma_device_init(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) +static ma_thread_result MA_THREADCALL ma_worker_thread(void* pData) { - ma_result result; - ma_device_config config; + ma_device* pDevice = (ma_device*)pData; + MA_ASSERT(pDevice != NULL); - if (pContext == NULL) { - return ma_device_init_ex(NULL, 0, NULL, pConfig, pDevice); - } - if (pDevice == NULL) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with invalid arguments (pDevice == NULL).", MA_INVALID_ARGS); - } - if (pConfig == NULL) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with invalid arguments (pConfig == NULL).", MA_INVALID_ARGS); - } +#ifdef MA_WIN32 + ma_CoInitializeEx(pDevice->pContext, NULL, MA_COINIT_VALUE); +#endif - /* We need to make a copy of the config so we can set default values if they were left unset in the input config. */ - config = *pConfig; + /* + When the device is being initialized it's initial state is set to MA_STATE_UNINITIALIZED. Before returning from + ma_device_init(), the state needs to be set to something valid. In miniaudio the device's default state immediately + after initialization is stopped, so therefore we need to mark the device as such. miniaudio will wait on the worker + thread to signal an event to know when the worker thread is ready for action. + */ + ma_device__set_state(pDevice, MA_STATE_STOPPED); + ma_event_signal(&pDevice->stopEvent); - /* Basic config validation. */ - if (config.deviceType != ma_device_type_playback && config.deviceType != ma_device_type_capture && config.deviceType != ma_device_type_duplex && config.deviceType != ma_device_type_loopback) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with an invalid config. Device type is invalid. Make sure the device type has been set in the config.", MA_INVALID_DEVICE_CONFIG); - } + for (;;) { /* <-- This loop just keeps the thread alive. The main audio loop is inside. */ + ma_stop_proc onStop; - if (config.deviceType == ma_device_type_capture || config.deviceType == ma_device_type_duplex) { - if (config.capture.channels > MA_MAX_CHANNELS) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with an invalid config. Capture channel count cannot exceed 32.", MA_INVALID_DEVICE_CONFIG); - } - if (!ma__is_channel_map_valid(config.capture.channelMap, config.capture.channels)) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with invalid config. Capture channel map is invalid.", MA_INVALID_DEVICE_CONFIG); - } - } + /* We wait on an event to know when something has requested that the device be started and the main loop entered. */ + ma_event_wait(&pDevice->wakeupEvent); - if (config.deviceType == ma_device_type_playback || config.deviceType == ma_device_type_duplex || config.deviceType == ma_device_type_loopback) { - if (config.playback.channels > MA_MAX_CHANNELS) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with an invalid config. Playback channel count cannot exceed 32.", MA_INVALID_DEVICE_CONFIG); - } - if (!ma__is_channel_map_valid(config.playback.channelMap, config.playback.channels)) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with invalid config. Playback channel map is invalid.", MA_INVALID_DEVICE_CONFIG); - } - } + /* Default result code. */ + pDevice->workResult = MA_SUCCESS; + /* If the reason for the wake up is that we are terminating, just break from the loop. */ + if (ma_device__get_state(pDevice) == MA_STATE_UNINITIALIZED) { + break; + } - ma_zero_object(pDevice); - pDevice->pContext = pContext; + /* + Getting to this point means the device is wanting to get started. The function that has requested that the device + be started will be waiting on an event (pDevice->startEvent) which means we need to make sure we signal the event + in both the success and error case. It's important that the state of the device is set _before_ signaling the event. + */ + MA_ASSERT(ma_device__get_state(pDevice) == MA_STATE_STARTING); - /* Set the user data and log callback ASAP to ensure it is available for the entire initialization process. */ - pDevice->pUserData = config.pUserData; - pDevice->onData = config.dataCallback; - pDevice->onStop = config.stopCallback; + /* Make sure the state is set appropriately. */ + ma_device__set_state(pDevice, MA_STATE_STARTED); + ma_event_signal(&pDevice->startEvent); - if (((ma_uintptr)pDevice % sizeof(pDevice)) != 0) { - if (pContext->logCallback) { - pContext->logCallback(pContext, pDevice, MA_LOG_LEVEL_WARNING, "WARNING: ma_device_init() called for a device that is not properly aligned. Thread safety is not supported."); + if (pDevice->pContext->onDeviceMainLoop != NULL) { + pDevice->pContext->onDeviceMainLoop(pDevice); + } else { + ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "No main loop implementation.", MA_API_NOT_FOUND); } - } - pDevice->noPreZeroedOutputBuffer = config.noPreZeroedOutputBuffer; - pDevice->noClip = config.noClip; - pDevice->masterVolumeFactor = 1; + /* + Getting here means we have broken from the main loop which happens the application has requested that device be stopped. Note that this + may have actually already happened above if the device was lost and miniaudio has attempted to re-initialize the device. In this case we + don't want to be doing this a second time. + */ + if (ma_device__get_state(pDevice) != MA_STATE_UNINITIALIZED) { + if (pDevice->pContext->onDeviceStop) { + pDevice->pContext->onDeviceStop(pDevice); + } + } - /* - When passing in 0 for the format/channels/rate/chmap it means the device will be using whatever is chosen by the backend. If everything is set - to defaults it means the format conversion pipeline will run on a fast path where data transfer is just passed straight through to the backend. - */ - if (config.sampleRate == 0) { - config.sampleRate = MA_DEFAULT_SAMPLE_RATE; - pDevice->usingDefaultSampleRate = MA_TRUE; - } + /* After the device has stopped, make sure an event is posted. */ + onStop = pDevice->onStop; + if (onStop) { + onStop(pDevice); + } - if (config.capture.format == ma_format_unknown) { - config.capture.format = MA_DEFAULT_FORMAT; - pDevice->capture.usingDefaultFormat = MA_TRUE; - } - if (config.capture.channels == 0) { - config.capture.channels = MA_DEFAULT_CHANNELS; - pDevice->capture.usingDefaultChannels = MA_TRUE; - } - if (config.capture.channelMap[0] == MA_CHANNEL_NONE) { - pDevice->capture.usingDefaultChannelMap = MA_TRUE; + /* + A function somewhere is waiting for the device to have stopped for real so we need to signal an event to allow it to continue. Note that + it's possible that the device has been uninitialized which means we need to _not_ change the status to stopped. We cannot go from an + uninitialized state to stopped state. + */ + if (ma_device__get_state(pDevice) != MA_STATE_UNINITIALIZED) { + ma_device__set_state(pDevice, MA_STATE_STOPPED); + ma_event_signal(&pDevice->stopEvent); + } } - if (config.playback.format == ma_format_unknown) { - config.playback.format = MA_DEFAULT_FORMAT; - pDevice->playback.usingDefaultFormat = MA_TRUE; - } - if (config.playback.channels == 0) { - config.playback.channels = MA_DEFAULT_CHANNELS; - pDevice->playback.usingDefaultChannels = MA_TRUE; - } - if (config.playback.channelMap[0] == MA_CHANNEL_NONE) { - pDevice->playback.usingDefaultChannelMap = MA_TRUE; - } + /* Make sure we aren't continuously waiting on a stop event. */ + ma_event_signal(&pDevice->stopEvent); /* <-- Is this still needed? */ +#ifdef MA_WIN32 + ma_CoUninitialize(pDevice->pContext); +#endif - /* Default buffer size. */ - if (config.bufferSizeInMilliseconds == 0 && config.bufferSizeInFrames == 0) { - config.bufferSizeInMilliseconds = (config.performanceProfile == ma_performance_profile_low_latency) ? MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_LOW_LATENCY : MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_CONSERVATIVE; - pDevice->usingDefaultBufferSize = MA_TRUE; - } + return (ma_thread_result)0; +} - /* Default periods. */ - if (config.periods == 0) { - config.periods = MA_DEFAULT_PERIODS; - pDevice->usingDefaultPeriods = MA_TRUE; - } - /* - Must have at least 3 periods for full-duplex mode. The idea is that the playback and capture positions hang out in the middle period, with the surrounding - periods acting as a buffer in case the capture and playback devices get's slightly out of sync. - */ - if (config.deviceType == ma_device_type_duplex && config.periods < 3) { - config.periods = 3; +/* Helper for determining whether or not the given device is initialized. */ +static ma_bool32 ma_device__is_initialized(ma_device* pDevice) +{ + if (pDevice == NULL) { + return MA_FALSE; } - - pDevice->type = config.deviceType; - pDevice->sampleRate = config.sampleRate; - - pDevice->capture.shareMode = config.capture.shareMode; - pDevice->capture.format = config.capture.format; - pDevice->capture.channels = config.capture.channels; - ma_channel_map_copy(pDevice->capture.channelMap, config.capture.channelMap, config.capture.channels); - - pDevice->playback.shareMode = config.playback.shareMode; - pDevice->playback.format = config.playback.format; - pDevice->playback.channels = config.playback.channels; - ma_channel_map_copy(pDevice->playback.channelMap, config.playback.channelMap, config.playback.channels); + return ma_device__get_state(pDevice) != MA_STATE_UNINITIALIZED; +} - /* The internal format, channel count and sample rate can be modified by the backend. */ - pDevice->capture.internalFormat = pDevice->capture.format; - pDevice->capture.internalChannels = pDevice->capture.channels; - pDevice->capture.internalSampleRate = pDevice->sampleRate; - ma_channel_map_copy(pDevice->capture.internalChannelMap, pDevice->capture.channelMap, pDevice->capture.channels); +#ifdef MA_WIN32 +static ma_result ma_context_uninit_backend_apis__win32(ma_context* pContext) +{ + ma_CoUninitialize(pContext); + ma_dlclose(pContext, pContext->win32.hUser32DLL); + ma_dlclose(pContext, pContext->win32.hOle32DLL); + ma_dlclose(pContext, pContext->win32.hAdvapi32DLL); - pDevice->playback.internalFormat = pDevice->playback.format; - pDevice->playback.internalChannels = pDevice->playback.channels; - pDevice->playback.internalSampleRate = pDevice->sampleRate; - ma_channel_map_copy(pDevice->playback.internalChannelMap, pDevice->playback.channelMap, pDevice->playback.channels); - + return MA_SUCCESS; +} - if (ma_mutex_init(pContext, &pDevice->lock) != MA_SUCCESS) { - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "Failed to create mutex.", MA_FAILED_TO_CREATE_MUTEX); +static ma_result ma_context_init_backend_apis__win32(ma_context* pContext) +{ +#ifdef MA_WIN32_DESKTOP + /* Ole32.dll */ + pContext->win32.hOle32DLL = ma_dlopen(pContext, "ole32.dll"); + if (pContext->win32.hOle32DLL == NULL) { + return MA_FAILED_TO_INIT_BACKEND; } - /* - When the device is started, the worker thread is the one that does the actual startup of the backend device. We - use a semaphore to wait for the background thread to finish the work. The same applies for stopping the device. - - Each of these semaphores is released internally by the worker thread when the work is completed. The start - semaphore is also used to wake up the worker thread. - */ - if (ma_event_init(pContext, &pDevice->wakeupEvent) != MA_SUCCESS) { - ma_mutex_uninit(&pDevice->lock); - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "Failed to create worker thread wakeup event.", MA_FAILED_TO_CREATE_EVENT); - } - if (ma_event_init(pContext, &pDevice->startEvent) != MA_SUCCESS) { - ma_event_uninit(&pDevice->wakeupEvent); - ma_mutex_uninit(&pDevice->lock); - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "Failed to create worker thread start event.", MA_FAILED_TO_CREATE_EVENT); - } - if (ma_event_init(pContext, &pDevice->stopEvent) != MA_SUCCESS) { - ma_event_uninit(&pDevice->startEvent); - ma_event_uninit(&pDevice->wakeupEvent); - ma_mutex_uninit(&pDevice->lock); - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "Failed to create worker thread stop event.", MA_FAILED_TO_CREATE_EVENT); + pContext->win32.CoInitializeEx = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "CoInitializeEx"); + pContext->win32.CoUninitialize = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "CoUninitialize"); + pContext->win32.CoCreateInstance = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "CoCreateInstance"); + pContext->win32.CoTaskMemFree = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "CoTaskMemFree"); + pContext->win32.PropVariantClear = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "PropVariantClear"); + pContext->win32.StringFromGUID2 = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "StringFromGUID2"); + + + /* User32.dll */ + pContext->win32.hUser32DLL = ma_dlopen(pContext, "user32.dll"); + if (pContext->win32.hUser32DLL == NULL) { + return MA_FAILED_TO_INIT_BACKEND; } + pContext->win32.GetForegroundWindow = (ma_proc)ma_dlsym(pContext, pContext->win32.hUser32DLL, "GetForegroundWindow"); + pContext->win32.GetDesktopWindow = (ma_proc)ma_dlsym(pContext, pContext->win32.hUser32DLL, "GetDesktopWindow"); - result = pContext->onDeviceInit(pContext, &config, pDevice); - if (result != MA_SUCCESS) { - return MA_NO_BACKEND; /* The error message will have been posted with ma_post_error() by the source of the error so don't bother calling it here. */ + + /* Advapi32.dll */ + pContext->win32.hAdvapi32DLL = ma_dlopen(pContext, "advapi32.dll"); + if (pContext->win32.hAdvapi32DLL == NULL) { + return MA_FAILED_TO_INIT_BACKEND; } - ma_device__post_init_setup(pDevice, pConfig->deviceType); + pContext->win32.RegOpenKeyExA = (ma_proc)ma_dlsym(pContext, pContext->win32.hAdvapi32DLL, "RegOpenKeyExA"); + pContext->win32.RegCloseKey = (ma_proc)ma_dlsym(pContext, pContext->win32.hAdvapi32DLL, "RegCloseKey"); + pContext->win32.RegQueryValueExA = (ma_proc)ma_dlsym(pContext, pContext->win32.hAdvapi32DLL, "RegQueryValueExA"); +#endif + ma_CoInitializeEx(pContext, NULL, MA_COINIT_VALUE); + return MA_SUCCESS; +} +#else +static ma_result ma_context_uninit_backend_apis__nix(ma_context* pContext) +{ +#if defined(MA_USE_RUNTIME_LINKING_FOR_PTHREAD) && !defined(MA_NO_RUNTIME_LINKING) + ma_dlclose(pContext, pContext->posix.pthreadSO); +#else + (void)pContext; +#endif - /* If the backend did not fill out a name for the device, try a generic method. */ - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - if (pDevice->capture.name[0] == '\0') { - if (ma_context__try_get_device_name_by_id(pContext, ma_device_type_capture, config.capture.pDeviceID, pDevice->capture.name, sizeof(pDevice->capture.name)) != MA_SUCCESS) { - ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), (config.capture.pDeviceID == NULL) ? MA_DEFAULT_CAPTURE_DEVICE_NAME : "Capture Device", (size_t)-1); - } + return MA_SUCCESS; +} + +static ma_result ma_context_init_backend_apis__nix(ma_context* pContext) +{ + /* pthread */ +#if defined(MA_USE_RUNTIME_LINKING_FOR_PTHREAD) && !defined(MA_NO_RUNTIME_LINKING) + const char* libpthreadFileNames[] = { + "libpthread.so", + "libpthread.so.0", + "libpthread.dylib" + }; + size_t i; + + for (i = 0; i < sizeof(libpthreadFileNames) / sizeof(libpthreadFileNames[0]); ++i) { + pContext->posix.pthreadSO = ma_dlopen(pContext, libpthreadFileNames[i]); + if (pContext->posix.pthreadSO != NULL) { + break; } } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) { - if (pDevice->playback.name[0] == '\0') { - if (ma_context__try_get_device_name_by_id(pContext, ma_device_type_playback, config.playback.pDeviceID, pDevice->playback.name, sizeof(pDevice->playback.name)) != MA_SUCCESS) { - ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), (config.playback.pDeviceID == NULL) ? MA_DEFAULT_PLAYBACK_DEVICE_NAME : "Playback Device", (size_t)-1); - } - } + + if (pContext->posix.pthreadSO == NULL) { + return MA_FAILED_TO_INIT_BACKEND; } + pContext->posix.pthread_create = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_create"); + pContext->posix.pthread_join = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_join"); + pContext->posix.pthread_mutex_init = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_mutex_init"); + pContext->posix.pthread_mutex_destroy = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_mutex_destroy"); + pContext->posix.pthread_mutex_lock = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_mutex_lock"); + pContext->posix.pthread_mutex_unlock = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_mutex_unlock"); + pContext->posix.pthread_cond_init = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_cond_init"); + pContext->posix.pthread_cond_destroy = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_cond_destroy"); + pContext->posix.pthread_cond_wait = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_cond_wait"); + pContext->posix.pthread_cond_signal = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_cond_signal"); + pContext->posix.pthread_attr_init = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_init"); + pContext->posix.pthread_attr_destroy = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_destroy"); + pContext->posix.pthread_attr_setschedpolicy = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_setschedpolicy"); + pContext->posix.pthread_attr_getschedparam = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_getschedparam"); + pContext->posix.pthread_attr_setschedparam = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_setschedparam"); +#else + pContext->posix.pthread_create = (ma_proc)pthread_create; + pContext->posix.pthread_join = (ma_proc)pthread_join; + pContext->posix.pthread_mutex_init = (ma_proc)pthread_mutex_init; + pContext->posix.pthread_mutex_destroy = (ma_proc)pthread_mutex_destroy; + pContext->posix.pthread_mutex_lock = (ma_proc)pthread_mutex_lock; + pContext->posix.pthread_mutex_unlock = (ma_proc)pthread_mutex_unlock; + pContext->posix.pthread_cond_init = (ma_proc)pthread_cond_init; + pContext->posix.pthread_cond_destroy = (ma_proc)pthread_cond_destroy; + pContext->posix.pthread_cond_wait = (ma_proc)pthread_cond_wait; + pContext->posix.pthread_cond_signal = (ma_proc)pthread_cond_signal; + pContext->posix.pthread_attr_init = (ma_proc)pthread_attr_init; + pContext->posix.pthread_attr_destroy = (ma_proc)pthread_attr_destroy; +#if !defined(__EMSCRIPTEN__) + pContext->posix.pthread_attr_setschedpolicy = (ma_proc)pthread_attr_setschedpolicy; + pContext->posix.pthread_attr_getschedparam = (ma_proc)pthread_attr_getschedparam; + pContext->posix.pthread_attr_setschedparam = (ma_proc)pthread_attr_setschedparam; +#endif +#endif - /* Some backends don't require the worker thread. */ - if (!ma_context_is_backend_asynchronous(pContext)) { - /* The worker thread. */ - if (ma_thread_create(pContext, &pDevice->thread, ma_worker_thread, pDevice) != MA_SUCCESS) { - ma_device_uninit(pDevice); - return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "Failed to create worker thread.", MA_FAILED_TO_CREATE_THREAD); - } + return MA_SUCCESS; +} +#endif - /* Wait for the worker thread to put the device into it's stopped state for real. */ - ma_event_wait(&pDevice->stopEvent); - } else { - ma_device__set_state(pDevice, MA_STATE_STOPPED); - } +static ma_result ma_context_init_backend_apis(ma_context* pContext) +{ + ma_result result; +#ifdef MA_WIN32 + result = ma_context_init_backend_apis__win32(pContext); +#else + result = ma_context_init_backend_apis__nix(pContext); +#endif + return result; +} -#ifdef MA_DEBUG_OUTPUT - printf("[%s]\n", ma_get_backend_name(pDevice->pContext->backend)); - if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { - printf(" %s (%s)\n", pDevice->capture.name, "Capture"); - printf(" Format: %s -> %s\n", ma_get_format_name(pDevice->capture.format), ma_get_format_name(pDevice->capture.internalFormat)); - printf(" Channels: %d -> %d\n", pDevice->capture.channels, pDevice->capture.internalChannels); - printf(" Sample Rate: %d -> %d\n", pDevice->sampleRate, pDevice->capture.internalSampleRate); - printf(" Buffer Size: %d/%d (%d)\n", pDevice->capture.internalBufferSizeInFrames, pDevice->capture.internalPeriods, (pDevice->capture.internalBufferSizeInFrames / pDevice->capture.internalPeriods)); - printf(" Conversion:\n"); - printf(" Pre Format Conversion: %s\n", pDevice->capture.converter.isPreFormatConversionRequired ? "YES" : "NO"); - printf(" Post Format Conversion: %s\n", pDevice->capture.converter.isPostFormatConversionRequired ? "YES" : "NO"); - printf(" Channel Routing: %s\n", pDevice->capture.converter.isChannelRoutingRequired ? "YES" : "NO"); - printf(" SRC: %s\n", pDevice->capture.converter.isSRCRequired ? "YES" : "NO"); - printf(" Channel Routing at Start: %s\n", pDevice->capture.converter.isChannelRoutingAtStart ? "YES" : "NO"); - printf(" Passthrough: %s\n", pDevice->capture.converter.isPassthrough ? "YES" : "NO"); - } - if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { - printf(" %s (%s)\n", pDevice->playback.name, "Playback"); - printf(" Format: %s -> %s\n", ma_get_format_name(pDevice->playback.format), ma_get_format_name(pDevice->playback.internalFormat)); - printf(" Channels: %d -> %d\n", pDevice->playback.channels, pDevice->playback.internalChannels); - printf(" Sample Rate: %d -> %d\n", pDevice->sampleRate, pDevice->playback.internalSampleRate); - printf(" Buffer Size: %d/%d (%d)\n", pDevice->playback.internalBufferSizeInFrames, pDevice->playback.internalPeriods, (pDevice->playback.internalBufferSizeInFrames / pDevice->playback.internalPeriods)); - printf(" Conversion:\n"); - printf(" Pre Format Conversion: %s\n", pDevice->playback.converter.isPreFormatConversionRequired ? "YES" : "NO"); - printf(" Post Format Conversion: %s\n", pDevice->playback.converter.isPostFormatConversionRequired ? "YES" : "NO"); - printf(" Channel Routing: %s\n", pDevice->playback.converter.isChannelRoutingRequired ? "YES" : "NO"); - printf(" SRC: %s\n", pDevice->playback.converter.isSRCRequired ? "YES" : "NO"); - printf(" Channel Routing at Start: %s\n", pDevice->playback.converter.isChannelRoutingAtStart ? "YES" : "NO"); - printf(" Passthrough: %s\n", pDevice->playback.converter.isPassthrough ? "YES" : "NO"); - } +static ma_result ma_context_uninit_backend_apis(ma_context* pContext) +{ + ma_result result; +#ifdef MA_WIN32 + result = ma_context_uninit_backend_apis__win32(pContext); +#else + result = ma_context_uninit_backend_apis__nix(pContext); #endif + return result; +} - ma_assert(ma_device__get_state(pDevice) == MA_STATE_STOPPED); - return MA_SUCCESS; + +static ma_bool32 ma_context_is_backend_asynchronous(ma_context* pContext) +{ + return pContext->isBackendAsynchronous; } -ma_result ma_device_init_ex(const ma_backend backends[], ma_uint32 backendCount, const ma_context_config* pContextConfig, const ma_device_config* pConfig, ma_device* pDevice) + +ma_context_config ma_context_config_init() +{ + ma_context_config config; + MA_ZERO_OBJECT(&config); + + return config; +} + +ma_result ma_context_init(const ma_backend backends[], ma_uint32 backendCount, const ma_context_config* pConfig, ma_context* pContext) { ma_result result; - ma_context* pContext; + ma_context_config config; ma_backend defaultBackends[ma_backend_null+1]; ma_uint32 iBackend; ma_backend* pBackendsToIterate; ma_uint32 backendsToIterateCount; - if (pConfig == NULL) { + if (pContext == NULL) { return MA_INVALID_ARGS; } - pContext = (ma_context*)ma_malloc(sizeof(*pContext)); - if (pContext == NULL) { - return MA_OUT_OF_MEMORY; + MA_ZERO_OBJECT(pContext); + + /* Always make sure the config is set first to ensure properties are available as soon as possible. */ + if (pConfig != NULL) { + config = *pConfig; + } else { + config = ma_context_config_init(); + } + + pContext->logCallback = config.logCallback; + pContext->threadPriority = config.threadPriority; + pContext->pUserData = config.pUserData; + + result = ma_allocation_callbacks_init_copy(&pContext->allocationCallbacks, &config.allocationCallbacks); + if (result != MA_SUCCESS) { + return result; + } + + /* Backend APIs need to be initialized first. This is where external libraries will be loaded and linked. */ + result = ma_context_init_backend_apis(pContext); + if (result != MA_SUCCESS) { + return result; } for (iBackend = 0; iBackend <= ma_backend_null; ++iBackend) { @@ -26452,5473 +28130,6318 @@ ma_result ma_device_init_ex(const ma_backend backends[], ma_uint32 backendCount, backendsToIterateCount = ma_countof(defaultBackends); } - result = MA_NO_BACKEND; + MA_ASSERT(pBackendsToIterate != NULL); + + for (iBackend = 0; iBackend < backendsToIterateCount; ++iBackend) { + ma_backend backend = pBackendsToIterate[iBackend]; + + result = MA_NO_BACKEND; + switch (backend) { + #ifdef MA_HAS_WASAPI + case ma_backend_wasapi: + { + result = ma_context_init__wasapi(&config, pContext); + } break; + #endif + #ifdef MA_HAS_DSOUND + case ma_backend_dsound: + { + result = ma_context_init__dsound(&config, pContext); + } break; + #endif + #ifdef MA_HAS_WINMM + case ma_backend_winmm: + { + result = ma_context_init__winmm(&config, pContext); + } break; + #endif + #ifdef MA_HAS_ALSA + case ma_backend_alsa: + { + result = ma_context_init__alsa(&config, pContext); + } break; + #endif + #ifdef MA_HAS_PULSEAUDIO + case ma_backend_pulseaudio: + { + result = ma_context_init__pulse(&config, pContext); + } break; + #endif + #ifdef MA_HAS_JACK + case ma_backend_jack: + { + result = ma_context_init__jack(&config, pContext); + } break; + #endif + #ifdef MA_HAS_COREAUDIO + case ma_backend_coreaudio: + { + result = ma_context_init__coreaudio(&config, pContext); + } break; + #endif + #ifdef MA_HAS_SNDIO + case ma_backend_sndio: + { + result = ma_context_init__sndio(&config, pContext); + } break; + #endif + #ifdef MA_HAS_AUDIO4 + case ma_backend_audio4: + { + result = ma_context_init__audio4(&config, pContext); + } break; + #endif + #ifdef MA_HAS_OSS + case ma_backend_oss: + { + result = ma_context_init__oss(&config, pContext); + } break; + #endif + #ifdef MA_HAS_AAUDIO + case ma_backend_aaudio: + { + result = ma_context_init__aaudio(&config, pContext); + } break; + #endif + #ifdef MA_HAS_OPENSL + case ma_backend_opensl: + { + result = ma_context_init__opensl(&config, pContext); + } break; + #endif + #ifdef MA_HAS_WEBAUDIO + case ma_backend_webaudio: + { + result = ma_context_init__webaudio(&config, pContext); + } break; + #endif + #ifdef MA_HAS_NULL + case ma_backend_null: + { + result = ma_context_init__null(&config, pContext); + } break; + #endif + + default: break; + } - for (iBackend = 0; iBackend < backendsToIterateCount; ++iBackend) { - result = ma_context_init(&pBackendsToIterate[iBackend], 1, pContextConfig, pContext); + /* If this iteration was successful, return. */ if (result == MA_SUCCESS) { - result = ma_device_init(pContext, pConfig, pDevice); - if (result == MA_SUCCESS) { - break; /* Success. */ - } else { - ma_context_uninit(pContext); /* Failure. */ + result = ma_mutex_init(pContext, &pContext->deviceEnumLock); + if (result != MA_SUCCESS) { + ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_WARNING, "Failed to initialize mutex for device enumeration. ma_context_get_devices() is not thread safe.", MA_FAILED_TO_CREATE_MUTEX); + } + result = ma_mutex_init(pContext, &pContext->deviceInfoLock); + if (result != MA_SUCCESS) { + ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_WARNING, "Failed to initialize mutex for device info retrieval. ma_context_get_device_info() is not thread safe.", MA_FAILED_TO_CREATE_MUTEX); } - } - } - if (result != MA_SUCCESS) { - ma_free(pContext); - return result; +#ifdef MA_DEBUG_OUTPUT + printf("[miniaudio] Endian: %s\n", ma_is_little_endian() ? "LE" : "BE"); + printf("[miniaudio] SSE2: %s\n", ma_has_sse2() ? "YES" : "NO"); + printf("[miniaudio] AVX2: %s\n", ma_has_avx2() ? "YES" : "NO"); + printf("[miniaudio] AVX512F: %s\n", ma_has_avx512f() ? "YES" : "NO"); + printf("[miniaudio] NEON: %s\n", ma_has_neon() ? "YES" : "NO"); +#endif + + pContext->backend = backend; + return result; + } } - pDevice->isOwnerOfContext = MA_TRUE; - return result; + /* If we get here it means an error occurred. */ + MA_ZERO_OBJECT(pContext); /* Safety. */ + return MA_NO_BACKEND; } -void ma_device_uninit(ma_device* pDevice) +ma_result ma_context_uninit(ma_context* pContext) { - if (!ma_device__is_initialized(pDevice)) { - return; - } - - /* Make sure the device is stopped first. The backends will probably handle this naturally, but I like to do it explicitly for my own sanity. */ - if (ma_device_is_started(pDevice)) { - ma_device_stop(pDevice); - } - - /* Putting the device into an uninitialized state will make the worker thread return. */ - ma_device__set_state(pDevice, MA_STATE_UNINITIALIZED); - - /* Wake up the worker thread and wait for it to properly terminate. */ - if (!ma_context_is_backend_asynchronous(pDevice->pContext)) { - ma_event_signal(&pDevice->wakeupEvent); - ma_thread_wait(&pDevice->thread); + if (pContext == NULL) { + return MA_INVALID_ARGS; } - pDevice->pContext->onDeviceUninit(pDevice); - - ma_event_uninit(&pDevice->stopEvent); - ma_event_uninit(&pDevice->startEvent); - ma_event_uninit(&pDevice->wakeupEvent); - ma_mutex_uninit(&pDevice->lock); + pContext->onUninit(pContext); - if (pDevice->isOwnerOfContext) { - ma_context_uninit(pDevice->pContext); - ma_free(pDevice->pContext); - } + ma_mutex_uninit(&pContext->deviceEnumLock); + ma_mutex_uninit(&pContext->deviceInfoLock); + ma__free_from_callbacks(pContext->pDeviceInfos, &pContext->allocationCallbacks); + ma_context_uninit_backend_apis(pContext); - ma_zero_object(pDevice); + return MA_SUCCESS; } -void ma_device_set_stop_callback(ma_device* pDevice, ma_stop_proc proc) -{ - if (pDevice == NULL) { - return; - } - - ma_atomic_exchange_ptr(&pDevice->onStop, proc); -} -ma_result ma_device_start(ma_device* pDevice) +ma_result ma_context_enumerate_devices(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData) { ma_result result; - if (pDevice == NULL) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "ma_device_start() called with invalid arguments (pDevice == NULL).", MA_INVALID_ARGS); - } - - if (ma_device__get_state(pDevice) == MA_STATE_UNINITIALIZED) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "ma_device_start() called for an uninitialized device.", MA_DEVICE_NOT_INITIALIZED); - } - - if (ma_device__get_state(pDevice) == MA_STATE_STARTED) { - return ma_post_error(pDevice, MA_LOG_LEVEL_WARNING, "ma_device_start() called when the device is already started.", MA_INVALID_OPERATION); /* Already started. Returning an error to let the application know because it probably means they're doing something wrong. */ + if (pContext == NULL || pContext->onEnumDevices == NULL || callback == NULL) { + return MA_INVALID_ARGS; } - result = MA_ERROR; - ma_mutex_lock(&pDevice->lock); + ma_mutex_lock(&pContext->deviceEnumLock); { - /* Starting and stopping are wrapped in a mutex which means we can assert that the device is in a stopped or paused state. */ - ma_assert(ma_device__get_state(pDevice) == MA_STATE_STOPPED); - - ma_device__set_state(pDevice, MA_STATE_STARTING); - - /* Asynchronous backends need to be handled differently. */ - if (ma_context_is_backend_asynchronous(pDevice->pContext)) { - result = pDevice->pContext->onDeviceStart(pDevice); - if (result == MA_SUCCESS) { - ma_device__set_state(pDevice, MA_STATE_STARTED); - } - } else { - /* - Synchronous backends are started by signaling an event that's being waited on in the worker thread. We first wake up the - thread and then wait for the start event. - */ - ma_event_signal(&pDevice->wakeupEvent); - - /* - Wait for the worker thread to finish starting the device. Note that the worker thread will be the one who puts the device - into the started state. Don't call ma_device__set_state() here. - */ - ma_event_wait(&pDevice->startEvent); - result = pDevice->workResult; - } + result = pContext->onEnumDevices(pContext, callback, pUserData); } - ma_mutex_unlock(&pDevice->lock); + ma_mutex_unlock(&pContext->deviceEnumLock); return result; } -ma_result ma_device_stop(ma_device* pDevice) + +static ma_bool32 ma_context_get_devices__enum_callback(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pInfo, void* pUserData) { - ma_result result; + /* + We need to insert the device info into our main internal buffer. Where it goes depends on the device type. If it's a capture device + it's just appended to the end. If it's a playback device it's inserted just before the first capture device. + */ - if (pDevice == NULL) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "ma_device_stop() called with invalid arguments (pDevice == NULL).", MA_INVALID_ARGS); - } + /* + First make sure we have room. Since the number of devices we add to the list is usually relatively small I've decided to use a + simple fixed size increment for buffer expansion. + */ + const ma_uint32 bufferExpansionCount = 2; + const ma_uint32 totalDeviceInfoCount = pContext->playbackDeviceInfoCount + pContext->captureDeviceInfoCount; - if (ma_device__get_state(pDevice) == MA_STATE_UNINITIALIZED) { - return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "ma_device_stop() called for an uninitialized device.", MA_DEVICE_NOT_INITIALIZED); - } + if (pContext->deviceInfoCapacity >= totalDeviceInfoCount) { + ma_uint32 oldCapacity = pContext->deviceInfoCapacity; + ma_uint32 newCapacity = oldCapacity + bufferExpansionCount; + ma_device_info* pNewInfos = (ma_device_info*)ma__realloc_from_callbacks(pContext->pDeviceInfos, sizeof(*pContext->pDeviceInfos)*newCapacity, sizeof(*pContext->pDeviceInfos)*oldCapacity, &pContext->allocationCallbacks); + if (pNewInfos == NULL) { + return MA_FALSE; /* Out of memory. */ + } - if (ma_device__get_state(pDevice) == MA_STATE_STOPPED) { - return ma_post_error(pDevice, MA_LOG_LEVEL_WARNING, "ma_device_stop() called when the device is already stopped.", MA_INVALID_OPERATION); /* Already stopped. Returning an error to let the application know because it probably means they're doing something wrong. */ + pContext->pDeviceInfos = pNewInfos; + pContext->deviceInfoCapacity = newCapacity; } - result = MA_ERROR; - ma_mutex_lock(&pDevice->lock); - { - /* Starting and stopping are wrapped in a mutex which means we can assert that the device is in a started or paused state. */ - ma_assert(ma_device__get_state(pDevice) == MA_STATE_STARTED); - - ma_device__set_state(pDevice, MA_STATE_STOPPING); - - /* There's no need to wake up the thread like we do when starting. */ + if (deviceType == ma_device_type_playback) { + /* Playback. Insert just before the first capture device. */ - if (pDevice->pContext->onDeviceStop) { - result = pDevice->pContext->onDeviceStop(pDevice); - } else { - result = MA_SUCCESS; + /* The first thing to do is move all of the capture devices down a slot. */ + ma_uint32 iFirstCaptureDevice = pContext->playbackDeviceInfoCount; + size_t iCaptureDevice; + for (iCaptureDevice = totalDeviceInfoCount; iCaptureDevice > iFirstCaptureDevice; --iCaptureDevice) { + pContext->pDeviceInfos[iCaptureDevice] = pContext->pDeviceInfos[iCaptureDevice-1]; } - /* Asynchronous backends need to be handled differently. */ - if (ma_context_is_backend_asynchronous(pDevice->pContext)) { - ma_device__set_state(pDevice, MA_STATE_STOPPED); - } else { - /* Synchronous backends. */ - - /* - We need to wait for the worker thread to become available for work before returning. Note that the worker thread will be - the one who puts the device into the stopped state. Don't call ma_device__set_state() here. - */ - ma_event_wait(&pDevice->stopEvent); - result = MA_SUCCESS; - } + /* Now just insert where the first capture device was before moving it down a slot. */ + pContext->pDeviceInfos[iFirstCaptureDevice] = *pInfo; + pContext->playbackDeviceInfoCount += 1; + } else { + /* Capture. Insert at the end. */ + pContext->pDeviceInfos[totalDeviceInfoCount] = *pInfo; + pContext->captureDeviceInfoCount += 1; } - ma_mutex_unlock(&pDevice->lock); - return result; + (void)pUserData; + return MA_TRUE; } -ma_bool32 ma_device_is_started(ma_device* pDevice) +ma_result ma_context_get_devices(ma_context* pContext, ma_device_info** ppPlaybackDeviceInfos, ma_uint32* pPlaybackDeviceCount, ma_device_info** ppCaptureDeviceInfos, ma_uint32* pCaptureDeviceCount) { - if (pDevice == NULL) { - return MA_FALSE; - } - - return ma_device__get_state(pDevice) == MA_STATE_STARTED; -} + ma_result result; -ma_result ma_device_set_master_volume(ma_device* pDevice, float volume) -{ - if (pDevice == NULL) { - return MA_INVALID_ARGS; - } + /* Safety. */ + if (ppPlaybackDeviceInfos != NULL) *ppPlaybackDeviceInfos = NULL; + if (pPlaybackDeviceCount != NULL) *pPlaybackDeviceCount = 0; + if (ppCaptureDeviceInfos != NULL) *ppCaptureDeviceInfos = NULL; + if (pCaptureDeviceCount != NULL) *pCaptureDeviceCount = 0; - if (volume < 0.0f || volume > 1.0f) { + if (pContext == NULL || pContext->onEnumDevices == NULL) { return MA_INVALID_ARGS; } - pDevice->masterVolumeFactor = volume; + /* Note that we don't use ma_context_enumerate_devices() here because we want to do locking at a higher level. */ + ma_mutex_lock(&pContext->deviceEnumLock); + { + /* Reset everything first. */ + pContext->playbackDeviceInfoCount = 0; + pContext->captureDeviceInfoCount = 0; - return MA_SUCCESS; -} + /* Now enumerate over available devices. */ + result = pContext->onEnumDevices(pContext, ma_context_get_devices__enum_callback, NULL); + if (result == MA_SUCCESS) { + /* Playback devices. */ + if (ppPlaybackDeviceInfos != NULL) { + *ppPlaybackDeviceInfos = pContext->pDeviceInfos; + } + if (pPlaybackDeviceCount != NULL) { + *pPlaybackDeviceCount = pContext->playbackDeviceInfoCount; + } -ma_result ma_device_get_master_volume(ma_device* pDevice, float* pVolume) -{ - if (pDevice == NULL || pVolume == NULL) { - return MA_INVALID_ARGS; + /* Capture devices. */ + if (ppCaptureDeviceInfos != NULL) { + *ppCaptureDeviceInfos = pContext->pDeviceInfos + pContext->playbackDeviceInfoCount; /* Capture devices come after playback devices. */ + } + if (pCaptureDeviceCount != NULL) { + *pCaptureDeviceCount = pContext->captureDeviceInfoCount; + } + } } + ma_mutex_unlock(&pContext->deviceEnumLock); - *pVolume = pDevice->masterVolumeFactor; - - return MA_SUCCESS; + return result; } -ma_result ma_device_set_master_gain_db(ma_device* pDevice, float gainDB) +ma_result ma_context_get_device_info(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo) { - if (gainDB > 0) { + ma_device_info deviceInfo; + + /* NOTE: Do not clear pDeviceInfo on entry. The reason is the pDeviceID may actually point to pDeviceInfo->id which will break things. */ + if (pContext == NULL || pDeviceInfo == NULL) { return MA_INVALID_ARGS; } - return ma_device_set_master_volume(pDevice, ma_gain_db_to_factor(gainDB)); -} - -ma_result ma_device_get_master_gain_db(ma_device* pDevice, float* pGainDB) -{ - float factor; - ma_result result; + MA_ZERO_OBJECT(&deviceInfo); - if (pGainDB == NULL) { - return MA_INVALID_ARGS; + /* Help the backend out by copying over the device ID if we have one. */ + if (pDeviceID != NULL) { + MA_COPY_MEMORY(&deviceInfo.id, pDeviceID, sizeof(*pDeviceID)); } - result = ma_device_get_master_volume(pDevice, &factor); - if (result != MA_SUCCESS) { + /* The backend may have an optimized device info retrieval function. If so, try that first. */ + if (pContext->onGetDeviceInfo != NULL) { + ma_result result; + ma_mutex_lock(&pContext->deviceInfoLock); + { + result = pContext->onGetDeviceInfo(pContext, deviceType, pDeviceID, shareMode, &deviceInfo); + } + ma_mutex_unlock(&pContext->deviceInfoLock); + + /* Clamp ranges. */ + deviceInfo.minChannels = ma_max(deviceInfo.minChannels, MA_MIN_CHANNELS); + deviceInfo.maxChannels = ma_min(deviceInfo.maxChannels, MA_MAX_CHANNELS); + deviceInfo.minSampleRate = ma_max(deviceInfo.minSampleRate, MA_MIN_SAMPLE_RATE); + deviceInfo.maxSampleRate = ma_min(deviceInfo.maxSampleRate, MA_MAX_SAMPLE_RATE); + + *pDeviceInfo = deviceInfo; return result; } - *pGainDB = ma_factor_to_gain_db(factor); - - return MA_SUCCESS; + /* Getting here means onGetDeviceInfo has not been set. */ + return MA_ERROR; } - -ma_context_config ma_context_config_init() +ma_bool32 ma_context_is_loopback_supported(ma_context* pContext) { - ma_context_config config; - ma_zero_object(&config); + if (pContext == NULL) { + return MA_FALSE; + } - return config; + return ma_is_loopback_supported(pContext->backend); } + ma_device_config ma_device_config_init(ma_device_type deviceType) { ma_device_config config; - ma_zero_object(&config); + MA_ZERO_OBJECT(&config); config.deviceType = deviceType; + /* Resampling defaults. We must never use the Speex backend by default because it uses licensed third party code. */ + config.resampling.algorithm = ma_resample_algorithm_linear; + config.resampling.linear.lpfCount = ma_min(MA_DEFAULT_RESAMPLER_LPF_FILTERS, MA_MAX_RESAMPLER_LPF_FILTERS); + config.resampling.speex.quality = 3; + return config; } -#endif /* MA_NO_DEVICE_IO */ - -void ma_get_standard_channel_map_microsoft(ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) +ma_result ma_device_init(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice) { - /* Based off the speaker configurations mentioned here: https://docs.microsoft.com/en-us/windows-hardware/drivers/ddi/content/ksmedia/ns-ksmedia-ksaudio_channel_config */ - switch (channels) - { - case 1: - { - channelMap[0] = MA_CHANNEL_MONO; - } break; - - case 2: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - } break; - - case 3: /* Not defined, but best guess. */ - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_FRONT_CENTER; - } break; - - case 4: - { -#ifndef MA_USE_QUAD_MICROSOFT_CHANNEL_MAP - /* Surround. Using the Surround profile has the advantage of the 3rd channel (MA_CHANNEL_FRONT_CENTER) mapping nicely with higher channel counts. */ - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_FRONT_CENTER; - channelMap[3] = MA_CHANNEL_BACK_CENTER; -#else - /* Quad. */ - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_BACK_LEFT; - channelMap[3] = MA_CHANNEL_BACK_RIGHT; -#endif - } break; + ma_result result; + ma_device_config config; - case 5: /* Not defined, but best guess. */ - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_FRONT_CENTER; - channelMap[3] = MA_CHANNEL_BACK_LEFT; - channelMap[4] = MA_CHANNEL_BACK_RIGHT; - } break; + if (pContext == NULL) { + return ma_device_init_ex(NULL, 0, NULL, pConfig, pDevice); + } + if (pDevice == NULL) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with invalid arguments (pDevice == NULL).", MA_INVALID_ARGS); + } + if (pConfig == NULL) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with invalid arguments (pConfig == NULL).", MA_INVALID_ARGS); + } - case 6: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_FRONT_CENTER; - channelMap[3] = MA_CHANNEL_LFE; - channelMap[4] = MA_CHANNEL_SIDE_LEFT; - channelMap[5] = MA_CHANNEL_SIDE_RIGHT; - } break; + /* We need to make a copy of the config so we can set default values if they were left unset in the input config. */ + config = *pConfig; - case 7: /* Not defined, but best guess. */ - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_FRONT_CENTER; - channelMap[3] = MA_CHANNEL_LFE; - channelMap[4] = MA_CHANNEL_BACK_CENTER; - channelMap[5] = MA_CHANNEL_SIDE_LEFT; - channelMap[6] = MA_CHANNEL_SIDE_RIGHT; - } break; + /* Basic config validation. */ + if (config.deviceType != ma_device_type_playback && config.deviceType != ma_device_type_capture && config.deviceType != ma_device_type_duplex && config.deviceType != ma_device_type_loopback) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with an invalid config. Device type is invalid. Make sure the device type has been set in the config.", MA_INVALID_DEVICE_CONFIG); + } - case 8: - default: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_FRONT_CENTER; - channelMap[3] = MA_CHANNEL_LFE; - channelMap[4] = MA_CHANNEL_BACK_LEFT; - channelMap[5] = MA_CHANNEL_BACK_RIGHT; - channelMap[6] = MA_CHANNEL_SIDE_LEFT; - channelMap[7] = MA_CHANNEL_SIDE_RIGHT; - } break; + if (config.deviceType == ma_device_type_capture || config.deviceType == ma_device_type_duplex) { + if (config.capture.channels > MA_MAX_CHANNELS) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with an invalid config. Capture channel count cannot exceed 32.", MA_INVALID_DEVICE_CONFIG); + } + if (!ma__is_channel_map_valid(config.capture.channelMap, config.capture.channels)) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with invalid config. Capture channel map is invalid.", MA_INVALID_DEVICE_CONFIG); + } } - /* Remainder. */ - if (channels > 8) { - ma_uint32 iChannel; - for (iChannel = 8; iChannel < MA_MAX_CHANNELS; ++iChannel) { - channelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-8)); + if (config.deviceType == ma_device_type_playback || config.deviceType == ma_device_type_duplex || config.deviceType == ma_device_type_loopback) { + if (config.playback.channels > MA_MAX_CHANNELS) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with an invalid config. Playback channel count cannot exceed 32.", MA_INVALID_DEVICE_CONFIG); + } + if (!ma__is_channel_map_valid(config.playback.channelMap, config.playback.channels)) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with invalid config. Playback channel map is invalid.", MA_INVALID_DEVICE_CONFIG); } } -} -void ma_get_standard_channel_map_alsa(ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) -{ - switch (channels) - { - case 1: - { - channelMap[0] = MA_CHANNEL_MONO; - } break; - case 2: - { - channelMap[0] = MA_CHANNEL_LEFT; - channelMap[1] = MA_CHANNEL_RIGHT; - } break; + MA_ZERO_OBJECT(pDevice); + pDevice->pContext = pContext; - case 3: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_FRONT_CENTER; - } break; + /* Set the user data and log callback ASAP to ensure it is available for the entire initialization process. */ + pDevice->pUserData = config.pUserData; + pDevice->onData = config.dataCallback; + pDevice->onStop = config.stopCallback; - case 4: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_BACK_LEFT; - channelMap[3] = MA_CHANNEL_BACK_RIGHT; - } break; + if (((ma_uintptr)pDevice % sizeof(pDevice)) != 0) { + if (pContext->logCallback) { + pContext->logCallback(pContext, pDevice, MA_LOG_LEVEL_WARNING, "WARNING: ma_device_init() called for a device that is not properly aligned. Thread safety is not supported."); + } + } - case 5: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_BACK_LEFT; - channelMap[3] = MA_CHANNEL_BACK_RIGHT; - channelMap[4] = MA_CHANNEL_FRONT_CENTER; - } break; + pDevice->noPreZeroedOutputBuffer = config.noPreZeroedOutputBuffer; + pDevice->noClip = config.noClip; + pDevice->masterVolumeFactor = 1; - case 6: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_BACK_LEFT; - channelMap[3] = MA_CHANNEL_BACK_RIGHT; - channelMap[4] = MA_CHANNEL_FRONT_CENTER; - channelMap[5] = MA_CHANNEL_LFE; - } break; + /* + When passing in 0 for the format/channels/rate/chmap it means the device will be using whatever is chosen by the backend. If everything is set + to defaults it means the format conversion pipeline will run on a fast path where data transfer is just passed straight through to the backend. + */ + if (config.sampleRate == 0) { + config.sampleRate = MA_DEFAULT_SAMPLE_RATE; + pDevice->usingDefaultSampleRate = MA_TRUE; + } + + if (config.capture.format == ma_format_unknown) { + config.capture.format = MA_DEFAULT_FORMAT; + pDevice->capture.usingDefaultFormat = MA_TRUE; + } + if (config.capture.channels == 0) { + config.capture.channels = MA_DEFAULT_CHANNELS; + pDevice->capture.usingDefaultChannels = MA_TRUE; + } + if (config.capture.channelMap[0] == MA_CHANNEL_NONE) { + pDevice->capture.usingDefaultChannelMap = MA_TRUE; + } + + if (config.playback.format == ma_format_unknown) { + config.playback.format = MA_DEFAULT_FORMAT; + pDevice->playback.usingDefaultFormat = MA_TRUE; + } + if (config.playback.channels == 0) { + config.playback.channels = MA_DEFAULT_CHANNELS; + pDevice->playback.usingDefaultChannels = MA_TRUE; + } + if (config.playback.channelMap[0] == MA_CHANNEL_NONE) { + pDevice->playback.usingDefaultChannelMap = MA_TRUE; + } - case 7: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_BACK_LEFT; - channelMap[3] = MA_CHANNEL_BACK_RIGHT; - channelMap[4] = MA_CHANNEL_FRONT_CENTER; - channelMap[5] = MA_CHANNEL_LFE; - channelMap[6] = MA_CHANNEL_BACK_CENTER; - } break; - case 8: - default: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_BACK_LEFT; - channelMap[3] = MA_CHANNEL_BACK_RIGHT; - channelMap[4] = MA_CHANNEL_FRONT_CENTER; - channelMap[5] = MA_CHANNEL_LFE; - channelMap[6] = MA_CHANNEL_SIDE_LEFT; - channelMap[7] = MA_CHANNEL_SIDE_RIGHT; - } break; + /* Default periods. */ + if (config.periods == 0) { + config.periods = MA_DEFAULT_PERIODS; + pDevice->usingDefaultPeriods = MA_TRUE; } - /* Remainder. */ - if (channels > 8) { - ma_uint32 iChannel; - for (iChannel = 8; iChannel < MA_MAX_CHANNELS; ++iChannel) { - channelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-8)); - } + /* + Must have at least 3 periods for full-duplex mode. The idea is that the playback and capture positions hang out in the middle period, with the surrounding + periods acting as a buffer in case the capture and playback devices get's slightly out of sync. + */ + if (config.deviceType == ma_device_type_duplex && config.periods < 3) { + config.periods = 3; } -} -void ma_get_standard_channel_map_rfc3551(ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) -{ - switch (channels) - { - case 1: - { - channelMap[0] = MA_CHANNEL_MONO; - } break; + /* Default buffer size. */ + if (config.periodSizeInMilliseconds == 0 && config.periodSizeInFrames == 0) { + config.periodSizeInMilliseconds = (config.performanceProfile == ma_performance_profile_low_latency) ? MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY : MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE; + pDevice->usingDefaultBufferSize = MA_TRUE; + } + - case 2: - { - channelMap[0] = MA_CHANNEL_LEFT; - channelMap[1] = MA_CHANNEL_RIGHT; - } break; - case 3: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_FRONT_CENTER; - } break; + pDevice->type = config.deviceType; + pDevice->sampleRate = config.sampleRate; + pDevice->resampling.algorithm = config.resampling.algorithm; + pDevice->resampling.linear.lpfCount = config.resampling.linear.lpfCount; + pDevice->resampling.speex.quality = config.resampling.speex.quality; - case 4: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_CENTER; - channelMap[2] = MA_CHANNEL_FRONT_RIGHT; - channelMap[3] = MA_CHANNEL_BACK_CENTER; - } break; + pDevice->capture.shareMode = config.capture.shareMode; + pDevice->capture.format = config.capture.format; + pDevice->capture.channels = config.capture.channels; + ma_channel_map_copy(pDevice->capture.channelMap, config.capture.channelMap, config.capture.channels); - case 5: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_FRONT_CENTER; - channelMap[3] = MA_CHANNEL_BACK_LEFT; - channelMap[4] = MA_CHANNEL_BACK_RIGHT; - } break; + pDevice->playback.shareMode = config.playback.shareMode; + pDevice->playback.format = config.playback.format; + pDevice->playback.channels = config.playback.channels; + ma_channel_map_copy(pDevice->playback.channelMap, config.playback.channelMap, config.playback.channels); - case 6: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_SIDE_LEFT; - channelMap[2] = MA_CHANNEL_FRONT_CENTER; - channelMap[3] = MA_CHANNEL_FRONT_RIGHT; - channelMap[4] = MA_CHANNEL_SIDE_RIGHT; - channelMap[5] = MA_CHANNEL_BACK_CENTER; - } break; + + /* The internal format, channel count and sample rate can be modified by the backend. */ + pDevice->capture.internalFormat = pDevice->capture.format; + pDevice->capture.internalChannels = pDevice->capture.channels; + pDevice->capture.internalSampleRate = pDevice->sampleRate; + ma_channel_map_copy(pDevice->capture.internalChannelMap, pDevice->capture.channelMap, pDevice->capture.channels); + + pDevice->playback.internalFormat = pDevice->playback.format; + pDevice->playback.internalChannels = pDevice->playback.channels; + pDevice->playback.internalSampleRate = pDevice->sampleRate; + ma_channel_map_copy(pDevice->playback.internalChannelMap, pDevice->playback.channelMap, pDevice->playback.channels); + + + if (ma_mutex_init(pContext, &pDevice->lock) != MA_SUCCESS) { + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "Failed to create mutex.", MA_FAILED_TO_CREATE_MUTEX); } - /* Remainder. */ - if (channels > 8) { - ma_uint32 iChannel; - for (iChannel = 6; iChannel < MA_MAX_CHANNELS; ++iChannel) { - channelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-6)); - } + /* + When the device is started, the worker thread is the one that does the actual startup of the backend device. We + use a semaphore to wait for the background thread to finish the work. The same applies for stopping the device. + + Each of these semaphores is released internally by the worker thread when the work is completed. The start + semaphore is also used to wake up the worker thread. + */ + if (ma_event_init(pContext, &pDevice->wakeupEvent) != MA_SUCCESS) { + ma_mutex_uninit(&pDevice->lock); + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "Failed to create worker thread wakeup event.", MA_FAILED_TO_CREATE_EVENT); + } + if (ma_event_init(pContext, &pDevice->startEvent) != MA_SUCCESS) { + ma_event_uninit(&pDevice->wakeupEvent); + ma_mutex_uninit(&pDevice->lock); + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "Failed to create worker thread start event.", MA_FAILED_TO_CREATE_EVENT); + } + if (ma_event_init(pContext, &pDevice->stopEvent) != MA_SUCCESS) { + ma_event_uninit(&pDevice->startEvent); + ma_event_uninit(&pDevice->wakeupEvent); + ma_mutex_uninit(&pDevice->lock); + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "Failed to create worker thread stop event.", MA_FAILED_TO_CREATE_EVENT); } -} -void ma_get_standard_channel_map_flac(ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) -{ - switch (channels) - { - case 1: - { - channelMap[0] = MA_CHANNEL_MONO; - } break; - case 2: - { - channelMap[0] = MA_CHANNEL_LEFT; - channelMap[1] = MA_CHANNEL_RIGHT; - } break; + result = pContext->onDeviceInit(pContext, &config, pDevice); + if (result != MA_SUCCESS) { + return result; + } - case 3: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_FRONT_CENTER; - } break; + ma_device__post_init_setup(pDevice, pConfig->deviceType); - case 4: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_BACK_LEFT; - channelMap[3] = MA_CHANNEL_BACK_RIGHT; - } break; - case 5: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_FRONT_CENTER; - channelMap[3] = MA_CHANNEL_BACK_LEFT; - channelMap[4] = MA_CHANNEL_BACK_RIGHT; - } break; + /* If the backend did not fill out a name for the device, try a generic method. */ + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + if (pDevice->capture.name[0] == '\0') { + if (ma_context__try_get_device_name_by_id(pContext, ma_device_type_capture, config.capture.pDeviceID, pDevice->capture.name, sizeof(pDevice->capture.name)) != MA_SUCCESS) { + ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), (config.capture.pDeviceID == NULL) ? MA_DEFAULT_CAPTURE_DEVICE_NAME : "Capture Device", (size_t)-1); + } + } + } + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) { + if (pDevice->playback.name[0] == '\0') { + if (ma_context__try_get_device_name_by_id(pContext, ma_device_type_playback, config.playback.pDeviceID, pDevice->playback.name, sizeof(pDevice->playback.name)) != MA_SUCCESS) { + ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), (config.playback.pDeviceID == NULL) ? MA_DEFAULT_PLAYBACK_DEVICE_NAME : "Playback Device", (size_t)-1); + } + } + } - case 6: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_FRONT_CENTER; - channelMap[3] = MA_CHANNEL_LFE; - channelMap[4] = MA_CHANNEL_BACK_LEFT; - channelMap[5] = MA_CHANNEL_BACK_RIGHT; - } break; - case 7: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_FRONT_CENTER; - channelMap[3] = MA_CHANNEL_LFE; - channelMap[4] = MA_CHANNEL_BACK_CENTER; - channelMap[5] = MA_CHANNEL_SIDE_LEFT; - channelMap[6] = MA_CHANNEL_SIDE_RIGHT; - } break; + /* Some backends don't require the worker thread. */ + if (!ma_context_is_backend_asynchronous(pContext)) { + /* The worker thread. */ + if (ma_thread_create(pContext, &pDevice->thread, ma_worker_thread, pDevice) != MA_SUCCESS) { + ma_device_uninit(pDevice); + return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "Failed to create worker thread.", MA_FAILED_TO_CREATE_THREAD); + } - case 8: - default: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_FRONT_CENTER; - channelMap[3] = MA_CHANNEL_LFE; - channelMap[4] = MA_CHANNEL_BACK_LEFT; - channelMap[5] = MA_CHANNEL_BACK_RIGHT; - channelMap[6] = MA_CHANNEL_SIDE_LEFT; - channelMap[7] = MA_CHANNEL_SIDE_RIGHT; - } break; + /* Wait for the worker thread to put the device into it's stopped state for real. */ + ma_event_wait(&pDevice->stopEvent); + } else { + ma_device__set_state(pDevice, MA_STATE_STOPPED); } - /* Remainder. */ - if (channels > 8) { - ma_uint32 iChannel; - for (iChannel = 8; iChannel < MA_MAX_CHANNELS; ++iChannel) { - channelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-8)); - } + +#ifdef MA_DEBUG_OUTPUT + printf("[%s]\n", ma_get_backend_name(pDevice->pContext->backend)); + if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) { + printf(" %s (%s)\n", pDevice->capture.name, "Capture"); + printf(" Format: %s -> %s\n", ma_get_format_name(pDevice->capture.format), ma_get_format_name(pDevice->capture.internalFormat)); + printf(" Channels: %d -> %d\n", pDevice->capture.channels, pDevice->capture.internalChannels); + printf(" Sample Rate: %d -> %d\n", pDevice->sampleRate, pDevice->capture.internalSampleRate); + printf(" Buffer Size: %d*%d (%d)\n", pDevice->capture.internalPeriodSizeInFrames, pDevice->capture.internalPeriods, (pDevice->capture.internalPeriodSizeInFrames * pDevice->capture.internalPeriods)); + printf(" Conversion:\n"); + printf(" Pre Format Conversion: %s\n", pDevice->capture.converter.hasPreFormatConversion ? "YES" : "NO"); + printf(" Post Format Conversion: %s\n", pDevice->capture.converter.hasPostFormatConversion ? "YES" : "NO"); + printf(" Channel Routing: %s\n", pDevice->capture.converter.hasChannelConverter ? "YES" : "NO"); + printf(" Resampling: %s\n", pDevice->capture.converter.hasResampler ? "YES" : "NO"); + printf(" Passthrough: %s\n", pDevice->capture.converter.isPassthrough ? "YES" : "NO"); + } + if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) { + printf(" %s (%s)\n", pDevice->playback.name, "Playback"); + printf(" Format: %s -> %s\n", ma_get_format_name(pDevice->playback.format), ma_get_format_name(pDevice->playback.internalFormat)); + printf(" Channels: %d -> %d\n", pDevice->playback.channels, pDevice->playback.internalChannels); + printf(" Sample Rate: %d -> %d\n", pDevice->sampleRate, pDevice->playback.internalSampleRate); + printf(" Buffer Size: %d*%d (%d)\n", pDevice->playback.internalPeriodSizeInFrames, pDevice->playback.internalPeriods, (pDevice->playback.internalPeriodSizeInFrames * pDevice->playback.internalPeriods)); + printf(" Conversion:\n"); + printf(" Pre Format Conversion: %s\n", pDevice->playback.converter.hasPreFormatConversion ? "YES" : "NO"); + printf(" Post Format Conversion: %s\n", pDevice->playback.converter.hasPostFormatConversion ? "YES" : "NO"); + printf(" Channel Routing: %s\n", pDevice->playback.converter.hasChannelConverter ? "YES" : "NO"); + printf(" Resampling: %s\n", pDevice->playback.converter.hasResampler ? "YES" : "NO"); + printf(" Passthrough: %s\n", pDevice->playback.converter.isPassthrough ? "YES" : "NO"); } +#endif + + + MA_ASSERT(ma_device__get_state(pDevice) == MA_STATE_STOPPED); + return MA_SUCCESS; } -void ma_get_standard_channel_map_vorbis(ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) +ma_result ma_device_init_ex(const ma_backend backends[], ma_uint32 backendCount, const ma_context_config* pContextConfig, const ma_device_config* pConfig, ma_device* pDevice) { - /* In Vorbis' type 0 channel mapping, the first two channels are not always the standard left/right - it will have the center speaker where the right usually goes. Why?! */ - switch (channels) - { - case 1: - { - channelMap[0] = MA_CHANNEL_MONO; - } break; + ma_result result; + ma_context* pContext; + ma_backend defaultBackends[ma_backend_null+1]; + ma_uint32 iBackend; + ma_backend* pBackendsToIterate; + ma_uint32 backendsToIterateCount; + ma_allocation_callbacks allocationCallbacks; - case 2: - { - channelMap[0] = MA_CHANNEL_LEFT; - channelMap[1] = MA_CHANNEL_RIGHT; - } break; + if (pConfig == NULL) { + return MA_INVALID_ARGS; + } - case 3: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_CENTER; - channelMap[2] = MA_CHANNEL_FRONT_RIGHT; - } break; + if (pContextConfig != NULL) { + result = ma_allocation_callbacks_init_copy(&allocationCallbacks, &pContextConfig->allocationCallbacks); + if (result != MA_SUCCESS) { + return result; + } + } else { + allocationCallbacks = ma_allocation_callbacks_init_default(); + } + - case 4: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_BACK_LEFT; - channelMap[3] = MA_CHANNEL_BACK_RIGHT; - } break; + pContext = (ma_context*)ma__malloc_from_callbacks(sizeof(*pContext), &allocationCallbacks); + if (pContext == NULL) { + return MA_OUT_OF_MEMORY; + } - case 5: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_CENTER; - channelMap[2] = MA_CHANNEL_FRONT_RIGHT; - channelMap[3] = MA_CHANNEL_BACK_LEFT; - channelMap[4] = MA_CHANNEL_BACK_RIGHT; - } break; + for (iBackend = 0; iBackend <= ma_backend_null; ++iBackend) { + defaultBackends[iBackend] = (ma_backend)iBackend; + } - case 6: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_CENTER; - channelMap[2] = MA_CHANNEL_FRONT_RIGHT; - channelMap[3] = MA_CHANNEL_BACK_LEFT; - channelMap[4] = MA_CHANNEL_BACK_RIGHT; - channelMap[5] = MA_CHANNEL_LFE; - } break; + pBackendsToIterate = (ma_backend*)backends; + backendsToIterateCount = backendCount; + if (pBackendsToIterate == NULL) { + pBackendsToIterate = (ma_backend*)defaultBackends; + backendsToIterateCount = ma_countof(defaultBackends); + } - case 7: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_CENTER; - channelMap[2] = MA_CHANNEL_FRONT_RIGHT; - channelMap[3] = MA_CHANNEL_SIDE_LEFT; - channelMap[4] = MA_CHANNEL_SIDE_RIGHT; - channelMap[5] = MA_CHANNEL_BACK_CENTER; - channelMap[6] = MA_CHANNEL_LFE; - } break; + result = MA_NO_BACKEND; - case 8: - default: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_CENTER; - channelMap[2] = MA_CHANNEL_FRONT_RIGHT; - channelMap[3] = MA_CHANNEL_SIDE_LEFT; - channelMap[4] = MA_CHANNEL_SIDE_RIGHT; - channelMap[5] = MA_CHANNEL_BACK_LEFT; - channelMap[6] = MA_CHANNEL_BACK_RIGHT; - channelMap[7] = MA_CHANNEL_LFE; - } break; + for (iBackend = 0; iBackend < backendsToIterateCount; ++iBackend) { + result = ma_context_init(&pBackendsToIterate[iBackend], 1, pContextConfig, pContext); + if (result == MA_SUCCESS) { + result = ma_device_init(pContext, pConfig, pDevice); + if (result == MA_SUCCESS) { + break; /* Success. */ + } else { + ma_context_uninit(pContext); /* Failure. */ + } + } } - /* Remainder. */ - if (channels > 8) { - ma_uint32 iChannel; - for (iChannel = 8; iChannel < MA_MAX_CHANNELS; ++iChannel) { - channelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-8)); - } + if (result != MA_SUCCESS) { + ma__free_from_callbacks(pContext, &allocationCallbacks); + return result; } + + pDevice->isOwnerOfContext = MA_TRUE; + return result; } -void ma_get_standard_channel_map_sound4(ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) +void ma_device_uninit(ma_device* pDevice) { - switch (channels) - { - case 1: - { - channelMap[0] = MA_CHANNEL_MONO; - } break; + if (!ma_device__is_initialized(pDevice)) { + return; + } - case 2: - { - channelMap[0] = MA_CHANNEL_LEFT; - channelMap[1] = MA_CHANNEL_RIGHT; - } break; + /* Make sure the device is stopped first. The backends will probably handle this naturally, but I like to do it explicitly for my own sanity. */ + if (ma_device_is_started(pDevice)) { + ma_device_stop(pDevice); + } - case 3: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_BACK_CENTER; - } break; + /* Putting the device into an uninitialized state will make the worker thread return. */ + ma_device__set_state(pDevice, MA_STATE_UNINITIALIZED); - case 4: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_BACK_LEFT; - channelMap[3] = MA_CHANNEL_BACK_RIGHT; - } break; + /* Wake up the worker thread and wait for it to properly terminate. */ + if (!ma_context_is_backend_asynchronous(pDevice->pContext)) { + ma_event_signal(&pDevice->wakeupEvent); + ma_thread_wait(&pDevice->thread); + } - case 5: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_BACK_LEFT; - channelMap[3] = MA_CHANNEL_BACK_RIGHT; - channelMap[4] = MA_CHANNEL_FRONT_CENTER; - } break; + pDevice->pContext->onDeviceUninit(pDevice); - case 6: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_BACK_LEFT; - channelMap[3] = MA_CHANNEL_BACK_RIGHT; - channelMap[4] = MA_CHANNEL_FRONT_CENTER; - channelMap[5] = MA_CHANNEL_LFE; - } break; + ma_event_uninit(&pDevice->stopEvent); + ma_event_uninit(&pDevice->startEvent); + ma_event_uninit(&pDevice->wakeupEvent); + ma_mutex_uninit(&pDevice->lock); - case 7: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_BACK_LEFT; - channelMap[3] = MA_CHANNEL_BACK_RIGHT; - channelMap[4] = MA_CHANNEL_FRONT_CENTER; - channelMap[5] = MA_CHANNEL_BACK_CENTER; - channelMap[6] = MA_CHANNEL_LFE; - } break; + if (pDevice->isOwnerOfContext) { + ma_allocation_callbacks allocationCallbacks = pDevice->pContext->allocationCallbacks; - case 8: - default: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_BACK_LEFT; - channelMap[3] = MA_CHANNEL_BACK_RIGHT; - channelMap[4] = MA_CHANNEL_FRONT_CENTER; - channelMap[5] = MA_CHANNEL_LFE; - channelMap[6] = MA_CHANNEL_SIDE_LEFT; - channelMap[7] = MA_CHANNEL_SIDE_RIGHT; - } break; + ma_context_uninit(pDevice->pContext); + ma__free_from_callbacks(pDevice->pContext, &allocationCallbacks); } - /* Remainder. */ - if (channels > 8) { - ma_uint32 iChannel; - for (iChannel = 8; iChannel < MA_MAX_CHANNELS; ++iChannel) { - channelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-8)); - } - } + MA_ZERO_OBJECT(pDevice); } -void ma_get_standard_channel_map_sndio(ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) +ma_result ma_device_start(ma_device* pDevice) { - switch (channels) - { - case 1: - { - channelMap[0] = MA_CHANNEL_MONO; - } break; - - case 2: - { - channelMap[0] = MA_CHANNEL_LEFT; - channelMap[1] = MA_CHANNEL_RIGHT; - } break; - - case 3: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_FRONT_CENTER; - } break; + ma_result result; - case 4: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_BACK_LEFT; - channelMap[3] = MA_CHANNEL_BACK_RIGHT; - } break; + if (pDevice == NULL) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "ma_device_start() called with invalid arguments (pDevice == NULL).", MA_INVALID_ARGS); + } - case 5: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_BACK_LEFT; - channelMap[3] = MA_CHANNEL_BACK_RIGHT; - channelMap[4] = MA_CHANNEL_FRONT_CENTER; - } break; + if (ma_device__get_state(pDevice) == MA_STATE_UNINITIALIZED) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "ma_device_start() called for an uninitialized device.", MA_DEVICE_NOT_INITIALIZED); + } - case 6: - default: - { - channelMap[0] = MA_CHANNEL_FRONT_LEFT; - channelMap[1] = MA_CHANNEL_FRONT_RIGHT; - channelMap[2] = MA_CHANNEL_BACK_LEFT; - channelMap[3] = MA_CHANNEL_BACK_RIGHT; - channelMap[4] = MA_CHANNEL_FRONT_CENTER; - channelMap[5] = MA_CHANNEL_LFE; - } break; + if (ma_device__get_state(pDevice) == MA_STATE_STARTED) { + return ma_post_error(pDevice, MA_LOG_LEVEL_WARNING, "ma_device_start() called when the device is already started.", MA_INVALID_OPERATION); /* Already started. Returning an error to let the application know because it probably means they're doing something wrong. */ } - /* Remainder. */ - if (channels > 6) { - ma_uint32 iChannel; - for (iChannel = 6; iChannel < MA_MAX_CHANNELS; ++iChannel) { - channelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-6)); + result = MA_ERROR; + ma_mutex_lock(&pDevice->lock); + { + /* Starting and stopping are wrapped in a mutex which means we can assert that the device is in a stopped or paused state. */ + MA_ASSERT(ma_device__get_state(pDevice) == MA_STATE_STOPPED); + + ma_device__set_state(pDevice, MA_STATE_STARTING); + + /* Asynchronous backends need to be handled differently. */ + if (ma_context_is_backend_asynchronous(pDevice->pContext)) { + result = pDevice->pContext->onDeviceStart(pDevice); + if (result == MA_SUCCESS) { + ma_device__set_state(pDevice, MA_STATE_STARTED); + } + } else { + /* + Synchronous backends are started by signaling an event that's being waited on in the worker thread. We first wake up the + thread and then wait for the start event. + */ + ma_event_signal(&pDevice->wakeupEvent); + + /* + Wait for the worker thread to finish starting the device. Note that the worker thread will be the one who puts the device + into the started state. Don't call ma_device__set_state() here. + */ + ma_event_wait(&pDevice->startEvent); + result = pDevice->workResult; } } + ma_mutex_unlock(&pDevice->lock); + + return result; } -void ma_get_standard_channel_map(ma_standard_channel_map standardChannelMap, ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) +ma_result ma_device_stop(ma_device* pDevice) { - switch (standardChannelMap) + ma_result result; + + if (pDevice == NULL) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "ma_device_stop() called with invalid arguments (pDevice == NULL).", MA_INVALID_ARGS); + } + + if (ma_device__get_state(pDevice) == MA_STATE_UNINITIALIZED) { + return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "ma_device_stop() called for an uninitialized device.", MA_DEVICE_NOT_INITIALIZED); + } + + if (ma_device__get_state(pDevice) == MA_STATE_STOPPED) { + return ma_post_error(pDevice, MA_LOG_LEVEL_WARNING, "ma_device_stop() called when the device is already stopped.", MA_INVALID_OPERATION); /* Already stopped. Returning an error to let the application know because it probably means they're doing something wrong. */ + } + + result = MA_ERROR; + ma_mutex_lock(&pDevice->lock); { - case ma_standard_channel_map_alsa: - { - ma_get_standard_channel_map_alsa(channels, channelMap); - } break; + /* Starting and stopping are wrapped in a mutex which means we can assert that the device is in a started or paused state. */ + MA_ASSERT(ma_device__get_state(pDevice) == MA_STATE_STARTED); - case ma_standard_channel_map_rfc3551: - { - ma_get_standard_channel_map_rfc3551(channels, channelMap); - } break; + ma_device__set_state(pDevice, MA_STATE_STOPPING); - case ma_standard_channel_map_flac: - { - ma_get_standard_channel_map_flac(channels, channelMap); - } break; + /* There's no need to wake up the thread like we do when starting. */ - case ma_standard_channel_map_vorbis: - { - ma_get_standard_channel_map_vorbis(channels, channelMap); - } break; + if (pDevice->pContext->onDeviceStop) { + result = pDevice->pContext->onDeviceStop(pDevice); + } else { + result = MA_SUCCESS; + } - case ma_standard_channel_map_sound4: - { - ma_get_standard_channel_map_sound4(channels, channelMap); - } break; - - case ma_standard_channel_map_sndio: - { - ma_get_standard_channel_map_sndio(channels, channelMap); - } break; + /* Asynchronous backends need to be handled differently. */ + if (ma_context_is_backend_asynchronous(pDevice->pContext)) { + ma_device__set_state(pDevice, MA_STATE_STOPPED); + } else { + /* Synchronous backends. */ - case ma_standard_channel_map_microsoft: - default: - { - ma_get_standard_channel_map_microsoft(channels, channelMap); - } break; + /* + We need to wait for the worker thread to become available for work before returning. Note that the worker thread will be + the one who puts the device into the stopped state. Don't call ma_device__set_state() here. + */ + ma_event_wait(&pDevice->stopEvent); + result = MA_SUCCESS; + } } + ma_mutex_unlock(&pDevice->lock); + + return result; } -void ma_channel_map_copy(ma_channel* pOut, const ma_channel* pIn, ma_uint32 channels) +ma_bool32 ma_device_is_started(ma_device* pDevice) { - if (pOut != NULL && pIn != NULL && channels > 0) { - ma_copy_memory(pOut, pIn, sizeof(*pOut) * channels); + if (pDevice == NULL) { + return MA_FALSE; } + + return ma_device__get_state(pDevice) == MA_STATE_STARTED; } -ma_bool32 ma_channel_map_valid(ma_uint32 channels, const ma_channel channelMap[MA_MAX_CHANNELS]) +ma_result ma_device_set_master_volume(ma_device* pDevice, float volume) { - if (channelMap == NULL) { - return MA_FALSE; + if (pDevice == NULL) { + return MA_INVALID_ARGS; } - /* A channel count of 0 is invalid. */ - if (channels == 0) { - return MA_FALSE; + if (volume < 0.0f || volume > 1.0f) { + return MA_INVALID_ARGS; } - /* It does not make sense to have a mono channel when there is more than 1 channel. */ - if (channels > 1) { - ma_uint32 iChannel; - for (iChannel = 0; iChannel < channels; ++iChannel) { - if (channelMap[iChannel] == MA_CHANNEL_MONO) { - return MA_FALSE; - } - } - } + pDevice->masterVolumeFactor = volume; - return MA_TRUE; + return MA_SUCCESS; } -ma_bool32 ma_channel_map_equal(ma_uint32 channels, const ma_channel channelMapA[MA_MAX_CHANNELS], const ma_channel channelMapB[MA_MAX_CHANNELS]) +ma_result ma_device_get_master_volume(ma_device* pDevice, float* pVolume) { - ma_uint32 iChannel; - - if (channelMapA == channelMapB) { - return MA_FALSE; + if (pVolume == NULL) { + return MA_INVALID_ARGS; } - if (channels == 0 || channels > MA_MAX_CHANNELS) { - return MA_FALSE; + if (pDevice == NULL) { + *pVolume = 0; + return MA_INVALID_ARGS; } - for (iChannel = 0; iChannel < channels; ++iChannel) { - if (channelMapA[iChannel] != channelMapB[iChannel]) { - return MA_FALSE; - } - } + *pVolume = pDevice->masterVolumeFactor; - return MA_TRUE; + return MA_SUCCESS; } -ma_bool32 ma_channel_map_blank(ma_uint32 channels, const ma_channel channelMap[MA_MAX_CHANNELS]) +ma_result ma_device_set_master_gain_db(ma_device* pDevice, float gainDB) { - ma_uint32 iChannel; - - for (iChannel = 0; iChannel < channels; ++iChannel) { - if (channelMap[iChannel] != MA_CHANNEL_NONE) { - return MA_FALSE; - } + if (gainDB > 0) { + return MA_INVALID_ARGS; } - return MA_TRUE; + return ma_device_set_master_volume(pDevice, ma_gain_db_to_factor(gainDB)); } -ma_bool32 ma_channel_map_contains_channel_position(ma_uint32 channels, const ma_channel channelMap[MA_MAX_CHANNELS], ma_channel channelPosition) +ma_result ma_device_get_master_gain_db(ma_device* pDevice, float* pGainDB) { - ma_uint32 iChannel; - for (iChannel = 0; iChannel < channels; ++iChannel) { - if (channelMap[iChannel] == channelPosition) { - return MA_TRUE; - } + float factor; + ma_result result; + + if (pGainDB == NULL) { + return MA_INVALID_ARGS; } - return MA_FALSE; -} + result = ma_device_get_master_volume(pDevice, &factor); + if (result != MA_SUCCESS) { + *pGainDB = 0; + return result; + } + *pGainDB = ma_factor_to_gain_db(factor); + return MA_SUCCESS; +} +#endif /* MA_NO_DEVICE_IO */ /************************************************************************************************************************************************************** -Format Conversion. +Biquad Filter **************************************************************************************************************************************************************/ -void ma_copy_memory_64(void* dst, const void* src, ma_uint64 sizeInBytes) +#ifndef MA_BIQUAD_FIXED_POINT_SHIFT +#define MA_BIQUAD_FIXED_POINT_SHIFT 14 +#endif + +static ma_int32 ma_biquad_float_to_fp(double x) { -#if 0xFFFFFFFFFFFFFFFF <= MA_SIZE_MAX - ma_copy_memory(dst, src, (size_t)sizeInBytes); -#else - while (sizeInBytes > 0) { - ma_uint64 bytesToCopyNow = sizeInBytes; - if (bytesToCopyNow > MA_SIZE_MAX) { - bytesToCopyNow = MA_SIZE_MAX; - } + return (ma_int32)(x * (1 << MA_BIQUAD_FIXED_POINT_SHIFT)); +} - ma_copy_memory(dst, src, (size_t)bytesToCopyNow); /* Safe cast to size_t. */ +ma_biquad_config ma_biquad_config_init(ma_format format, ma_uint32 channels, double b0, double b1, double b2, double a0, double a1, double a2) +{ + ma_biquad_config config; - sizeInBytes -= bytesToCopyNow; - dst = ( void*)(( ma_uint8*)dst + bytesToCopyNow); - src = (const void*)((const ma_uint8*)src + bytesToCopyNow); - } -#endif + MA_ZERO_OBJECT(&config); + config.format = format; + config.channels = channels; + config.b0 = b0; + config.b1 = b1; + config.b2 = b2; + config.a0 = a0; + config.a1 = a1; + config.a2 = a2; + + return config; } -void ma_zero_memory_64(void* dst, ma_uint64 sizeInBytes) +ma_result ma_biquad_init(const ma_biquad_config* pConfig, ma_biquad* pBQ) { -#if 0xFFFFFFFFFFFFFFFF <= MA_SIZE_MAX - ma_zero_memory(dst, (size_t)sizeInBytes); -#else - while (sizeInBytes > 0) { - ma_uint64 bytesToZeroNow = sizeInBytes; - if (bytesToZeroNow > MA_SIZE_MAX) { - bytesToZeroNow = MA_SIZE_MAX; - } + if (pBQ == NULL) { + return MA_INVALID_ARGS; + } - ma_zero_memory(dst, (size_t)bytesToZeroNow); /* Safe cast to size_t. */ + MA_ZERO_OBJECT(pBQ); - sizeInBytes -= bytesToZeroNow; - dst = (void*)((ma_uint8*)dst + bytesToZeroNow); + if (pConfig == NULL) { + return MA_INVALID_ARGS; } -#endif -} + return ma_biquad_reinit(pConfig, pBQ); +} -/* u8 */ -void ma_pcm_u8_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +ma_result ma_biquad_reinit(const ma_biquad_config* pConfig, ma_biquad* pBQ) { - (void)ditherMode; - ma_copy_memory_64(dst, src, count * sizeof(ma_uint8)); -} + if (pBQ == NULL || pConfig == NULL) { + return MA_INVALID_ARGS; + } + if (pConfig->a0 == 0) { + return MA_INVALID_ARGS; /* Division by zero. */ + } + + /* Only supporting f32 and s16. */ + if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) { + return MA_INVALID_ARGS; + } + + /* The format cannot be changed after initialization. */ + if (pBQ->format != ma_format_unknown && pBQ->format != pConfig->format) { + return MA_INVALID_OPERATION; + } + + /* The channel count cannot be changed after initialization. */ + if (pBQ->channels != 0 && pBQ->channels != pConfig->channels) { + return MA_INVALID_OPERATION; + } -void ma_pcm_u8_to_s16__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_int16* dst_s16 = (ma_int16*)dst; - const ma_uint8* src_u8 = (const ma_uint8*)src; - ma_uint64 i; - for (i = 0; i < count; i += 1) { - ma_int16 x = src_u8[i]; - x = x - 128; - x = x << 8; - dst_s16[i] = x; + pBQ->format = pConfig->format; + pBQ->channels = pConfig->channels; + + /* Normalize. */ + if (pConfig->format == ma_format_f32) { + pBQ->b0.f32 = (float)(pConfig->b0 / pConfig->a0); + pBQ->b1.f32 = (float)(pConfig->b1 / pConfig->a0); + pBQ->b2.f32 = (float)(pConfig->b2 / pConfig->a0); + pBQ->a1.f32 = (float)(pConfig->a1 / pConfig->a0); + pBQ->a2.f32 = (float)(pConfig->a2 / pConfig->a0); + } else { + pBQ->b0.s32 = ma_biquad_float_to_fp(pConfig->b0 / pConfig->a0); + pBQ->b1.s32 = ma_biquad_float_to_fp(pConfig->b1 / pConfig->a0); + pBQ->b2.s32 = ma_biquad_float_to_fp(pConfig->b2 / pConfig->a0); + pBQ->a1.s32 = ma_biquad_float_to_fp(pConfig->a1 / pConfig->a0); + pBQ->a2.s32 = ma_biquad_float_to_fp(pConfig->a2 / pConfig->a0); } - (void)ditherMode; + return MA_SUCCESS; } -void ma_pcm_u8_to_s16__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +static MA_INLINE void ma_biquad_process_pcm_frame_f32__direct_form_2_transposed(ma_biquad* pBQ, float* pY, const float* pX) { - ma_pcm_u8_to_s16__reference(dst, src, count, ditherMode); -} + ma_uint32 c; + const float b0 = pBQ->b0.f32; + const float b1 = pBQ->b1.f32; + const float b2 = pBQ->b2.f32; + const float a1 = pBQ->a1.f32; + const float a2 = pBQ->a2.f32; + + for (c = 0; c < pBQ->channels; c += 1) { + float r1 = pBQ->r1[c].f32; + float r2 = pBQ->r2[c].f32; + float x = pX[c]; + float y; -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_u8_to_s16__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_u8_to_s16__optimized(dst, src, count, ditherMode); + y = b0*x + r1; + r1 = b1*x - a1*y + r2; + r2 = b2*x - a2*y; + + pY[c] = y; + pBQ->r1[c].f32 = r1; + pBQ->r2[c].f32 = r2; + } } -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_u8_to_s16__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +static MA_INLINE void ma_biquad_process_pcm_frame_f32(ma_biquad* pBQ, float* pY, const float* pX) { - ma_pcm_u8_to_s16__optimized(dst, src, count, ditherMode); + ma_biquad_process_pcm_frame_f32__direct_form_2_transposed(pBQ, pY, pX); } -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_u8_to_s16__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +static MA_INLINE void ma_biquad_process_pcm_frame_s16__direct_form_2_transposed(ma_biquad* pBQ, ma_int16* pY, const ma_int16* pX) { - ma_pcm_u8_to_s16__avx2(dst, src, count, ditherMode); + ma_uint32 c; + const ma_int32 b0 = pBQ->b0.s32; + const ma_int32 b1 = pBQ->b1.s32; + const ma_int32 b2 = pBQ->b2.s32; + const ma_int32 a1 = pBQ->a1.s32; + const ma_int32 a2 = pBQ->a2.s32; + + for (c = 0; c < pBQ->channels; c += 1) { + ma_int32 r1 = pBQ->r1[c].s32; + ma_int32 r2 = pBQ->r2[c].s32; + ma_int32 x = pX[c]; + ma_int32 y; + + y = (b0*x + r1) >> MA_BIQUAD_FIXED_POINT_SHIFT; + r1 = (b1*x - a1*y + r2); + r2 = (b2*x - a2*y); + + pY[c] = (ma_int16)ma_clamp(y, -32768, 32767); + pBQ->r1[c].s32 = r1; + pBQ->r2[c].s32 = r2; + } } -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_u8_to_s16__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +static MA_INLINE void ma_biquad_process_pcm_frame_s16(ma_biquad* pBQ, ma_int16* pY, const ma_int16* pX) { - ma_pcm_u8_to_s16__optimized(dst, src, count, ditherMode); + ma_biquad_process_pcm_frame_s16__direct_form_2_transposed(pBQ, pY, pX); } -#endif -void ma_pcm_u8_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +ma_result ma_biquad_process_pcm_frames(ma_biquad* pBQ, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount) { -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_u8_to_s16__reference(dst, src, count, ditherMode); -#else - ma_pcm_u8_to_s16__optimized(dst, src, count, ditherMode); -#endif -} + ma_uint32 n; + if (pBQ == NULL || pFramesOut == NULL || pFramesIn == NULL) { + return MA_INVALID_ARGS; + } -void ma_pcm_u8_to_s24__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_uint8* dst_s24 = (ma_uint8*)dst; - const ma_uint8* src_u8 = (const ma_uint8*)src; + /* Note that the logic below needs to support in-place filtering. That is, it must support the case where pFramesOut and pFramesIn are the same. */ - ma_uint64 i; - for (i = 0; i < count; i += 1) { - ma_int16 x = src_u8[i]; - x = x - 128; + if (pBQ->format == ma_format_f32) { + /* */ float* pY = ( float*)pFramesOut; + const float* pX = (const float*)pFramesIn; - dst_s24[i*3+0] = 0; - dst_s24[i*3+1] = 0; - dst_s24[i*3+2] = (ma_uint8)((ma_int8)x); + for (n = 0; n < frameCount; n += 1) { + ma_biquad_process_pcm_frame_f32__direct_form_2_transposed(pBQ, pY, pX); + pY += pBQ->channels; + pX += pBQ->channels; + } + } else if (pBQ->format == ma_format_s16) { + /* */ ma_int16* pY = ( ma_int16*)pFramesOut; + const ma_int16* pX = (const ma_int16*)pFramesIn; + + for (n = 0; n < frameCount; n += 1) { + ma_biquad_process_pcm_frame_s16__direct_form_2_transposed(pBQ, pY, pX); + pY += pBQ->channels; + pX += pBQ->channels; + } + } else { + MA_ASSERT(MA_FALSE); + return MA_INVALID_ARGS; /* Format not supported. Should never hit this because it's checked in ma_biquad_init() and ma_biquad_reinit(). */ } - (void)ditherMode; + return MA_SUCCESS; } -void ma_pcm_u8_to_s24__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +ma_uint32 ma_biquad_get_latency(ma_biquad* pBQ) { - ma_pcm_u8_to_s24__reference(dst, src, count, ditherMode); -} + if (pBQ == NULL) { + return 0; + } -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_u8_to_s24__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_u8_to_s24__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_u8_to_s24__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_u8_to_s24__optimized(dst, src, count, ditherMode); + return 2; } -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_u8_to_s24__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + + +/************************************************************************************************************************************************************** + +Low-Pass Filter + +**************************************************************************************************************************************************************/ +ma_lpf_config ma_lpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency) { - ma_pcm_u8_to_s24__avx2(dst, src, count, ditherMode); + ma_lpf_config config; + + MA_ZERO_OBJECT(&config); + config.format = format; + config.channels = channels; + config.sampleRate = sampleRate; + config.cutoffFrequency = cutoffFrequency; + + return config; } -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_u8_to_s24__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +static MA_INLINE ma_biquad_config ma_lpf__get_biquad_config(const ma_lpf_config* pConfig) { - ma_pcm_u8_to_s24__optimized(dst, src, count, ditherMode); + ma_biquad_config bqConfig; + double q; + double w; + double s; + double c; + double a; + + MA_ASSERT(pConfig != NULL); + + q = 0.707107; + w = 2 * MA_PI_D * pConfig->cutoffFrequency / pConfig->sampleRate; + s = ma_sin(w); + c = ma_cos(w); + a = s / (2*q); + + bqConfig.b0 = (1 - c) / 2; + bqConfig.b1 = 1 - c; + bqConfig.b2 = (1 - c) / 2; + bqConfig.a0 = 1 + a; + bqConfig.a1 = -2 * c; + bqConfig.a2 = 1 - a; + + bqConfig.format = pConfig->format; + bqConfig.channels = pConfig->channels; + + return bqConfig; } -#endif -void ma_pcm_u8_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +ma_result ma_lpf_init(const ma_lpf_config* pConfig, ma_lpf* pLPF) { -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_u8_to_s24__reference(dst, src, count, ditherMode); -#else - ma_pcm_u8_to_s24__optimized(dst, src, count, ditherMode); -#endif -} + ma_result result; + ma_biquad_config bqConfig; + if (pLPF == NULL) { + return MA_INVALID_ARGS; + } -void ma_pcm_u8_to_s32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_int32* dst_s32 = (ma_int32*)dst; - const ma_uint8* src_u8 = (const ma_uint8*)src; + MA_ZERO_OBJECT(pLPF); - ma_uint64 i; - for (i = 0; i < count; i += 1) { - ma_int32 x = src_u8[i]; - x = x - 128; - x = x << 24; - dst_s32[i] = x; + if (pConfig == NULL) { + return MA_INVALID_ARGS; } - (void)ditherMode; + bqConfig = ma_lpf__get_biquad_config(pConfig); + result = ma_biquad_init(&bqConfig, &pLPF->bq); + if (result != MA_SUCCESS) { + return result; + } + + return MA_SUCCESS; } -void ma_pcm_u8_to_s32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +ma_result ma_lpf_reinit(const ma_lpf_config* pConfig, ma_lpf* pLPF) { - ma_pcm_u8_to_s32__reference(dst, src, count, ditherMode); + ma_result result; + ma_biquad_config bqConfig; + + if (pLPF == NULL || pConfig == NULL) { + return MA_INVALID_ARGS; + } + + bqConfig = ma_lpf__get_biquad_config(pConfig); + result = ma_biquad_reinit(&bqConfig, &pLPF->bq); + if (result != MA_SUCCESS) { + return result; + } + + return MA_SUCCESS; } -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_u8_to_s32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +static MA_INLINE void ma_lpf_process_pcm_frame_s16(ma_lpf* pLPF, ma_int16* pFrameOut, const ma_int16* pFrameIn) { - ma_pcm_u8_to_s32__optimized(dst, src, count, ditherMode); + ma_biquad_process_pcm_frame_s16(&pLPF->bq, pFrameOut, pFrameIn); } -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_u8_to_s32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +static MA_INLINE void ma_lpf_process_pcm_frame_f32(ma_lpf* pLPF, float* pFrameOut, const float* pFrameIn) { - ma_pcm_u8_to_s32__optimized(dst, src, count, ditherMode); + ma_biquad_process_pcm_frame_f32(&pLPF->bq, pFrameOut, pFrameIn); } -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_u8_to_s32__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +ma_result ma_lpf_process_pcm_frames(ma_lpf* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount) { - ma_pcm_u8_to_s32__avx2(dst, src, count, ditherMode); + if (pLPF == NULL) { + return MA_INVALID_ARGS; + } + + return ma_biquad_process_pcm_frames(&pLPF->bq, pFramesOut, pFramesIn, frameCount); } -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_u8_to_s32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +ma_uint32 ma_lpf_get_latency(ma_lpf* pLPF) { - ma_pcm_u8_to_s32__optimized(dst, src, count, ditherMode); + if (pLPF == NULL) { + return 0; + } + + return ma_biquad_get_latency(&pLPF->bq); } -#endif -void ma_pcm_u8_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +/************************************************************************************************************************************************************** + +High-Pass Filtering + +**************************************************************************************************************************************************************/ +ma_hpf_config ma_hpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency) { -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_u8_to_s32__reference(dst, src, count, ditherMode); -#else - ma_pcm_u8_to_s32__optimized(dst, src, count, ditherMode); -#endif + ma_hpf_config config; + + MA_ZERO_OBJECT(&config); + config.format = format; + config.channels = channels; + config.sampleRate = sampleRate; + config.cutoffFrequency = cutoffFrequency; + + return config; } +static MA_INLINE ma_biquad_config ma_hpf__get_biquad_config(const ma_hpf_config* pConfig) +{ + ma_biquad_config bqConfig; + double q; + double w; + double s; + double c; + double a; + + MA_ASSERT(pConfig != NULL); -void ma_pcm_u8_to_f32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + q = 0.707107; + w = 2 * MA_PI_D * pConfig->cutoffFrequency / pConfig->sampleRate; + s = ma_sin(w); + c = ma_cos(w); + a = s / (2*q); + + bqConfig.b0 = (1 + c) / 2; + bqConfig.b1 = -(1 + c); + bqConfig.b2 = (1 + c) / 2; + bqConfig.a0 = 1 + a; + bqConfig.a1 = -2 * c; + bqConfig.a2 = 1 - a; + + bqConfig.format = pConfig->format; + bqConfig.channels = pConfig->channels; + + return bqConfig; +} + +ma_result ma_hpf_init(const ma_hpf_config* pConfig, ma_hpf* pHPF) { - float* dst_f32 = (float*)dst; - const ma_uint8* src_u8 = (const ma_uint8*)src; + ma_result result; + ma_biquad_config bqConfig; - ma_uint64 i; - for (i = 0; i < count; i += 1) { - float x = (float)src_u8[i]; - x = x * 0.00784313725490196078f; /* 0..255 to 0..2 */ - x = x - 1; /* 0..2 to -1..1 */ + if (pHPF == NULL) { + return MA_INVALID_ARGS; + } - dst_f32[i] = x; + MA_ZERO_OBJECT(pHPF); + + if (pConfig == NULL) { + return MA_INVALID_ARGS; } - (void)ditherMode; + bqConfig = ma_hpf__get_biquad_config(pConfig); + result = ma_biquad_init(&bqConfig, &pHPF->bq); + if (result != MA_SUCCESS) { + return result; + } + + return MA_SUCCESS; } -void ma_pcm_u8_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +ma_result ma_hpf_reinit(const ma_hpf_config* pConfig, ma_hpf* pHPF) { - ma_pcm_u8_to_f32__reference(dst, src, count, ditherMode); + ma_result result; + ma_biquad_config bqConfig; + + if (pHPF == NULL || pConfig == NULL) { + return MA_INVALID_ARGS; + } + + bqConfig = ma_hpf__get_biquad_config(pConfig); + result = ma_biquad_reinit(&bqConfig, &pHPF->bq); + if (result != MA_SUCCESS) { + return result; + } + + return MA_SUCCESS; } -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_u8_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +ma_result ma_hpf_process_pcm_frames(ma_hpf* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount) { - ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode); + if (pHPF == NULL) { + return MA_INVALID_ARGS; + } + + return ma_biquad_process_pcm_frames(&pHPF->bq, pFramesOut, pFramesIn, frameCount); } -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_u8_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +ma_uint32 ma_hpf_get_latency(ma_hpf* pHPF) { - ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode); + if (pHPF == NULL) { + return 0; + } + + return ma_biquad_get_latency(&pHPF->bq); } -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_u8_to_f32__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + + +/************************************************************************************************************************************************************** + +Band-Pass Filtering + +**************************************************************************************************************************************************************/ +ma_bpf_config ma_bpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency) { - ma_pcm_u8_to_f32__avx2(dst, src, count, ditherMode); + ma_bpf_config config; + + MA_ZERO_OBJECT(&config); + config.format = format; + config.channels = channels; + config.sampleRate = sampleRate; + config.cutoffFrequency = cutoffFrequency; + + return config; } -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_u8_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + + +static MA_INLINE ma_biquad_config ma_bpf__get_biquad_config(const ma_bpf_config* pConfig) { - ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode); + ma_biquad_config bqConfig; + double q; + double w; + double s; + double c; + double a; + + MA_ASSERT(pConfig != NULL); + + q = 0.707107; + w = 2 * MA_PI_D * pConfig->cutoffFrequency / pConfig->sampleRate; + s = ma_sin(w); + c = ma_cos(w); + a = s / (2*q); + + bqConfig.b0 = q * a; + bqConfig.b1 = 0; + bqConfig.b2 = -q * a; + bqConfig.a0 = 1 + a; + bqConfig.a1 = -2 * c; + bqConfig.a2 = 1 - a; + + bqConfig.format = pConfig->format; + bqConfig.channels = pConfig->channels; + + return bqConfig; } -#endif -void ma_pcm_u8_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +ma_result ma_bpf_init(const ma_bpf_config* pConfig, ma_bpf* pBPF) { -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_u8_to_f32__reference(dst, src, count, ditherMode); -#else - ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode); -#endif -} + ma_result result; + ma_biquad_config bqConfig; + + if (pBPF == NULL) { + return MA_INVALID_ARGS; + } + + MA_ZERO_OBJECT(pBPF); + + if (pConfig == NULL) { + return MA_INVALID_ARGS; + } + bqConfig = ma_bpf__get_biquad_config(pConfig); + result = ma_biquad_init(&bqConfig, &pBPF->bq); + if (result != MA_SUCCESS) { + return result; + } + return MA_SUCCESS; +} -void ma_pcm_interleave_u8__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) +ma_result ma_bpf_reinit(const ma_bpf_config* pConfig, ma_bpf* pBPF) { - ma_uint8* dst_u8 = (ma_uint8*)dst; - const ma_uint8** src_u8 = (const ma_uint8**)src; + ma_result result; + ma_biquad_config bqConfig; - ma_uint64 iFrame; - for (iFrame = 0; iFrame < frameCount; iFrame += 1) { - ma_uint32 iChannel; - for (iChannel = 0; iChannel < channels; iChannel += 1) { - dst_u8[iFrame*channels + iChannel] = src_u8[iChannel][iFrame]; - } + if (pBPF == NULL || pConfig == NULL) { + return MA_INVALID_ARGS; + } + + bqConfig = ma_bpf__get_biquad_config(pConfig); + result = ma_biquad_reinit(&bqConfig, &pBPF->bq); + if (result != MA_SUCCESS) { + return result; } + + return MA_SUCCESS; } -void ma_pcm_interleave_u8__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) +ma_result ma_bpf_process_pcm_frames(ma_bpf* pBPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount) { - ma_uint8* dst_u8 = (ma_uint8*)dst; - const ma_uint8** src_u8 = (const ma_uint8**)src; - - if (channels == 1) { - ma_copy_memory_64(dst, src[0], frameCount * sizeof(ma_uint8)); - } else if (channels == 2) { - ma_uint64 iFrame; - for (iFrame = 0; iFrame < frameCount; iFrame += 1) { - dst_u8[iFrame*2 + 0] = src_u8[0][iFrame]; - dst_u8[iFrame*2 + 1] = src_u8[1][iFrame]; - } - } else { - ma_uint64 iFrame; - for (iFrame = 0; iFrame < frameCount; iFrame += 1) { - ma_uint32 iChannel; - for (iChannel = 0; iChannel < channels; iChannel += 1) { - dst_u8[iFrame*channels + iChannel] = src_u8[iChannel][iFrame]; - } - } + if (pBPF == NULL) { + return MA_INVALID_ARGS; } + + return ma_biquad_process_pcm_frames(&pBPF->bq, pFramesOut, pFramesIn, frameCount); } -void ma_pcm_interleave_u8(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) +ma_uint32 ma_bpf_get_latency(ma_bpf* pBPF) { -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_interleave_u8__reference(dst, src, frameCount, channels); -#else - ma_pcm_interleave_u8__optimized(dst, src, frameCount, channels); -#endif + if (pBPF == NULL) { + return 0; + } + + return ma_biquad_get_latency(&pBPF->bq); } -void ma_pcm_deinterleave_u8__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) -{ - ma_uint8** dst_u8 = (ma_uint8**)dst; - const ma_uint8* src_u8 = (const ma_uint8*)src; - ma_uint64 iFrame; - for (iFrame = 0; iFrame < frameCount; iFrame += 1) { - ma_uint32 iChannel; - for (iChannel = 0; iChannel < channels; iChannel += 1) { - dst_u8[iChannel][iFrame] = src_u8[iFrame*channels + iChannel]; - } - } -} +/************************************************************************************************************************************************************** -void ma_pcm_deinterleave_u8__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) +Resampling + +**************************************************************************************************************************************************************/ +ma_linear_resampler_config ma_linear_resampler_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut) { - ma_pcm_deinterleave_u8__reference(dst, src, frameCount, channels); + ma_linear_resampler_config config; + MA_ZERO_OBJECT(&config); + config.format = format; + config.channels = channels; + config.sampleRateIn = sampleRateIn; + config.sampleRateOut = sampleRateOut; + config.lpfCount = 1; + config.lpfNyquistFactor = 1; + + return config; } -void ma_pcm_deinterleave_u8(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) +static ma_result ma_linear_resampler_set_rate_internal(ma_linear_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut, ma_bool32 isResamplerAlreadyInitialized) { -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_deinterleave_u8__reference(dst, src, frameCount, channels); -#else - ma_pcm_deinterleave_u8__optimized(dst, src, frameCount, channels); -#endif -} + ma_uint32 gcf; + if (pResampler == NULL) { + return MA_INVALID_ARGS; + } -/* s16 */ -void ma_pcm_s16_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_uint8* dst_u8 = (ma_uint8*)dst; - const ma_int16* src_s16 = (const ma_int16*)src; + if (sampleRateIn == 0 || sampleRateOut == 0) { + return MA_INVALID_ARGS; + } - if (ditherMode == ma_dither_mode_none) { - ma_uint64 i; - for (i = 0; i < count; i += 1) { - ma_int16 x = src_s16[i]; - x = x >> 8; - x = x + 128; - dst_u8[i] = (ma_uint8)x; + /* Simplify the sample rate. */ + gcf = ma_gcf_u32(pResampler->config.sampleRateIn, pResampler->config.sampleRateOut); + pResampler->config.sampleRateIn /= gcf; + pResampler->config.sampleRateOut /= gcf; + + if (pResampler->config.lpfCount > 0) { + ma_result result; + ma_uint32 iFilter; + ma_uint32 lpfSampleRate; + double lpfCutoffFrequency; + ma_lpf_config lpfConfig; + + if (pResampler->config.lpfCount > MA_MAX_RESAMPLER_LPF_FILTERS) { + return MA_INVALID_ARGS; } - } else { - ma_uint64 i; - for (i = 0; i < count; i += 1) { - ma_int16 x = src_s16[i]; - /* Dither. Don't overflow. */ - ma_int32 dither = ma_dither_s32(ditherMode, -0x80, 0x7F); - if ((x + dither) <= 0x7FFF) { - x = (ma_int16)(x + dither); + lpfSampleRate = (ma_uint32)(ma_max(pResampler->config.sampleRateIn, pResampler->config.sampleRateOut)); + lpfCutoffFrequency = ( double)(ma_min(pResampler->config.sampleRateIn, pResampler->config.sampleRateOut) * 0.5 * pResampler->config.lpfNyquistFactor); + + lpfConfig = ma_lpf_config_init(pResampler->config.format, pResampler->config.channels, lpfSampleRate, lpfCutoffFrequency); + + /* + If the resampler is alreay initialized we don't want to do a fresh initialization of the low-pass filter because it will result in the cached frames + getting cleared. Instead we re-initialize the filter which will maintain any cached frames. + */ + result = MA_SUCCESS; + for (iFilter = 0; iFilter < pResampler->config.lpfCount; iFilter += 1) { + if (isResamplerAlreadyInitialized) { + result = ma_lpf_reinit(&lpfConfig, &pResampler->lpf[iFilter]); } else { - x = 0x7FFF; + result = ma_lpf_init(&lpfConfig, &pResampler->lpf[iFilter]); } - x = x >> 8; - x = x + 128; - dst_u8[i] = (ma_uint8)x; + if (result != MA_SUCCESS) { + break; + } + } + + if (result != MA_SUCCESS) { + return result; /* Failed to initialize the low-pass filter. */ } } -} -void ma_pcm_s16_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s16_to_u8__reference(dst, src, count, ditherMode); -} + pResampler->inAdvanceInt = pResampler->config.sampleRateIn / pResampler->config.sampleRateOut; + pResampler->inAdvanceFrac = pResampler->config.sampleRateIn % pResampler->config.sampleRateOut; -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_s16_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_s16_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_s16_to_u8__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s16_to_u8__avx2(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_s16_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode); + /* Make sure the fractional part is less than the output sample rate. */ + pResampler->inTimeInt += pResampler->inTimeFrac / pResampler->config.sampleRateOut; + pResampler->inTimeFrac = pResampler->inTimeFrac % pResampler->config.sampleRateOut; + + return MA_SUCCESS; } -#endif -void ma_pcm_s16_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +ma_result ma_linear_resampler_init(const ma_linear_resampler_config* pConfig, ma_linear_resampler* pResampler) { -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_s16_to_u8__reference(dst, src, count, ditherMode); -#else - ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode); -#endif -} + ma_result result; + if (pResampler == NULL) { + return MA_INVALID_ARGS; + } -void ma_pcm_s16_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - (void)ditherMode; - ma_copy_memory_64(dst, src, count * sizeof(ma_int16)); -} + MA_ZERO_OBJECT(pResampler); + if (pConfig == NULL) { + return MA_INVALID_ARGS; + } -void ma_pcm_s16_to_s24__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_uint8* dst_s24 = (ma_uint8*)dst; - const ma_int16* src_s16 = (const ma_int16*)src; + pResampler->config = *pConfig; - ma_uint64 i; - for (i = 0; i < count; i += 1) { - dst_s24[i*3+0] = 0; - dst_s24[i*3+1] = (ma_uint8)(src_s16[i] & 0xFF); - dst_s24[i*3+2] = (ma_uint8)(src_s16[i] >> 8); + /* Setting the rate will set up the filter and time advances for us. */ + result = ma_linear_resampler_set_rate_internal(pResampler, pConfig->sampleRateIn, pConfig->sampleRateOut, /* isResamplerAlreadyInitialized = */ MA_FALSE); + if (result != MA_SUCCESS) { + return result; } - (void)ditherMode; + pResampler->inTimeInt = 1; /* Set this to one to force an input sample to always be loaded for the first output frame. */ + pResampler->inTimeFrac = 0; + + return MA_SUCCESS; } -void ma_pcm_s16_to_s24__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +void ma_linear_resampler_uninit(ma_linear_resampler* pResampler) { - ma_pcm_s16_to_s24__reference(dst, src, count, ditherMode); + if (pResampler == NULL) { + return; + } } -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_s16_to_s24__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +static MA_INLINE ma_int16 ma_linear_resampler_mix_s16(ma_int16 x, ma_int16 y, ma_int32 a, const ma_int32 shift) { - ma_pcm_s16_to_s24__optimized(dst, src, count, ditherMode); + ma_int32 b; + ma_int32 c; + ma_int32 r; + + MA_ASSERT(a <= (1<> shift); } -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_s16_to_s24__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +static void ma_linear_resampler_interpolate_frame_s16(ma_linear_resampler* pResampler, ma_int16* pFrameOut) { - ma_pcm_s16_to_s24__optimized(dst, src, count, ditherMode); + ma_uint32 c; + ma_uint32 a; + const ma_uint32 shift = 12; + + MA_ASSERT(pResampler != NULL); + MA_ASSERT(pFrameOut != NULL); + + a = (pResampler->inTimeFrac << shift) / pResampler->config.sampleRateOut; + + for (c = 0; c < pResampler->config.channels; c += 1) { + ma_int16 s = ma_linear_resampler_mix_s16(pResampler->x0.s16[c], pResampler->x1.s16[c], a, shift); + pFrameOut[c] = s; + } } -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_s16_to_s24__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + + +static void ma_linear_resampler_interpolate_frame_f32(ma_linear_resampler* pResampler, float* pFrameOut) { - ma_pcm_s16_to_s24__avx2(dst, src, count, ditherMode); + ma_uint32 c; + float a; + + MA_ASSERT(pResampler != NULL); + MA_ASSERT(pFrameOut != NULL); + + a = (float)pResampler->inTimeFrac / pResampler->config.sampleRateOut; + + for (c = 0; c < pResampler->config.channels; c += 1) { + float s = ma_mix_f32_fast(pResampler->x0.f32[c], pResampler->x1.f32[c], a); + pFrameOut[c] = s; + } } -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_s16_to_s24__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +static ma_result ma_linear_resampler_process_pcm_frames_s16_downsample(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) { - ma_pcm_s16_to_s24__optimized(dst, src, count, ditherMode); + const ma_int16* pFramesInS16; + /* */ ma_int16* pFramesOutS16; + ma_uint64 frameCountIn; + ma_uint64 frameCountOut; + ma_uint64 framesProcessedIn; + ma_uint64 framesProcessedOut; + + MA_ASSERT(pResampler != NULL); + MA_ASSERT(pFrameCountIn != NULL); + MA_ASSERT(pFrameCountOut != NULL); + + pFramesInS16 = (const ma_int16*)pFramesIn; + pFramesOutS16 = ( ma_int16*)pFramesOut; + frameCountIn = *pFrameCountIn; + frameCountOut = *pFrameCountOut; + framesProcessedIn = 0; + framesProcessedOut = 0; + + for (;;) { + if (framesProcessedOut >= frameCountOut) { + break; + } + + /* Before interpolating we need to load the buffers. When doing this we need to ensure we run every input sample through the filter. */ + while (pResampler->inTimeInt > 0 && frameCountIn > 0) { + ma_uint32 iFilter; + ma_uint32 iChannel; + + if (pFramesInS16 != NULL) { + for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) { + pResampler->x0.s16[iChannel] = pResampler->x1.s16[iChannel]; + pResampler->x1.s16[iChannel] = pFramesInS16[iChannel]; + } + pFramesInS16 += pResampler->config.channels; + } else { + for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) { + pResampler->x0.s16[iChannel] = pResampler->x1.s16[iChannel]; + pResampler->x1.s16[iChannel] = 0; + } + } + + /* Filter. */ + for (iFilter = 0; iFilter < pResampler->config.lpfCount; iFilter += 1) { + ma_lpf_process_pcm_frame_s16(&pResampler->lpf[iFilter], pResampler->x1.s16, pResampler->x1.s16); + } + + frameCountIn -= 1; + framesProcessedIn += 1; + pResampler->inTimeInt -= 1; + } + + if (pResampler->inTimeInt > 0) { + break; /* Ran out of input data. */ + } + + /* Getting here means the frames have been loaded and filtered and we can generate the next output frame. */ + if (pFramesOutS16 != NULL) { + MA_ASSERT(pResampler->inTimeInt == 0); + ma_linear_resampler_interpolate_frame_s16(pResampler, pFramesOutS16); + + pFramesOutS16 += pResampler->config.channels; + } + + framesProcessedOut += 1; + + /* Advance time forward. */ + pResampler->inTimeInt += pResampler->inAdvanceInt; + pResampler->inTimeFrac += pResampler->inAdvanceFrac; + if (pResampler->inTimeFrac >= pResampler->config.sampleRateOut) { + pResampler->inTimeFrac -= pResampler->config.sampleRateOut; + pResampler->inTimeInt += 1; + } + } + + *pFrameCountIn = framesProcessedIn; + *pFrameCountOut = framesProcessedOut; + + return MA_SUCCESS; } -#endif -void ma_pcm_s16_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +static ma_result ma_linear_resampler_process_pcm_frames_s16_upsample(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) { -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_s16_to_s24__reference(dst, src, count, ditherMode); -#else - ma_pcm_s16_to_s24__optimized(dst, src, count, ditherMode); -#endif -} + const ma_int16* pFramesInS16; + /* */ ma_int16* pFramesOutS16; + ma_uint64 frameCountIn; + ma_uint64 frameCountOut; + ma_uint64 framesProcessedIn; + ma_uint64 framesProcessedOut; + + MA_ASSERT(pResampler != NULL); + MA_ASSERT(pFrameCountIn != NULL); + MA_ASSERT(pFrameCountOut != NULL); + + pFramesInS16 = (const ma_int16*)pFramesIn; + pFramesOutS16 = ( ma_int16*)pFramesOut; + frameCountIn = *pFrameCountIn; + frameCountOut = *pFrameCountOut; + framesProcessedIn = 0; + framesProcessedOut = 0; + + for (;;) { + ma_uint32 iFilter; + + if (framesProcessedOut >= frameCountOut) { + break; + } + + /* Before interpolating we need to load the buffers. */ + while (pResampler->inTimeInt > 0 && frameCountIn > 0) { + ma_uint32 iChannel; + + if (pFramesInS16 != NULL) { + for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) { + pResampler->x0.s16[iChannel] = pResampler->x1.s16[iChannel]; + pResampler->x1.s16[iChannel] = pFramesInS16[iChannel]; + } + pFramesInS16 += pResampler->config.channels; + } else { + for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) { + pResampler->x0.s16[iChannel] = pResampler->x1.s16[iChannel]; + pResampler->x1.s16[iChannel] = 0; + } + } + + frameCountIn -= 1; + framesProcessedIn += 1; + pResampler->inTimeInt -= 1; + } + + if (pResampler->inTimeInt > 0) { + break; /* Ran out of input data. */ + } + /* Getting here means the frames have been loaded and we can generate the next output frame. */ + if (pFramesOutS16 != NULL) { + MA_ASSERT(pResampler->inTimeInt == 0); + ma_linear_resampler_interpolate_frame_s16(pResampler, pFramesOutS16); -void ma_pcm_s16_to_s32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_int32* dst_s32 = (ma_int32*)dst; - const ma_int16* src_s16 = (const ma_int16*)src; + /* Filter. */ + for (iFilter = 0; iFilter < pResampler->config.lpfCount; iFilter += 1) { + ma_lpf_process_pcm_frame_s16(&pResampler->lpf[iFilter], pFramesOutS16, pFramesOutS16); + } - ma_uint64 i; - for (i = 0; i < count; i += 1) { - dst_s32[i] = src_s16[i] << 16; + pFramesOutS16 += pResampler->config.channels; + } + + framesProcessedOut += 1; + + /* Advance time forward. */ + pResampler->inTimeInt += pResampler->inAdvanceInt; + pResampler->inTimeFrac += pResampler->inAdvanceFrac; + if (pResampler->inTimeFrac >= pResampler->config.sampleRateOut) { + pResampler->inTimeFrac -= pResampler->config.sampleRateOut; + pResampler->inTimeInt += 1; + } } - (void)ditherMode; -} + *pFrameCountIn = framesProcessedIn; + *pFrameCountOut = framesProcessedOut; -void ma_pcm_s16_to_s32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s16_to_s32__reference(dst, src, count, ditherMode); + return MA_SUCCESS; } -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_s16_to_s32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s16_to_s32__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_s16_to_s32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s16_to_s32__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_s16_to_s32__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s16_to_s32__avx2(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_s16_to_s32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +static ma_result ma_linear_resampler_process_pcm_frames_s16(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) { - ma_pcm_s16_to_s32__optimized(dst, src, count, ditherMode); -} -#endif + MA_ASSERT(pResampler != NULL); -void ma_pcm_s16_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_s16_to_s32__reference(dst, src, count, ditherMode); -#else - ma_pcm_s16_to_s32__optimized(dst, src, count, ditherMode); -#endif + if (pResampler->config.sampleRateIn > pResampler->config.sampleRateOut) { + return ma_linear_resampler_process_pcm_frames_s16_downsample(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); + } else { + return ma_linear_resampler_process_pcm_frames_s16_upsample(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); + } } -void ma_pcm_s16_to_f32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +static ma_result ma_linear_resampler_process_pcm_frames_f32_downsample(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) { - float* dst_f32 = (float*)dst; - const ma_int16* src_s16 = (const ma_int16*)src; + const float* pFramesInF32; + /* */ float* pFramesOutF32; + ma_uint64 frameCountIn; + ma_uint64 frameCountOut; + ma_uint64 framesProcessedIn; + ma_uint64 framesProcessedOut; - ma_uint64 i; - for (i = 0; i < count; i += 1) { - float x = (float)src_s16[i]; + MA_ASSERT(pResampler != NULL); + MA_ASSERT(pFrameCountIn != NULL); + MA_ASSERT(pFrameCountOut != NULL); -#if 0 - /* The accurate way. */ - x = x + 32768.0f; /* -32768..32767 to 0..65535 */ - x = x * 0.00003051804379339284f; /* 0..65536 to 0..2 */ - x = x - 1; /* 0..2 to -1..1 */ -#else - /* The fast way. */ - x = x * 0.000030517578125f; /* -32768..32767 to -1..0.999969482421875 */ -#endif + pFramesInF32 = (const float*)pFramesIn; + pFramesOutF32 = ( float*)pFramesOut; + frameCountIn = *pFrameCountIn; + frameCountOut = *pFrameCountOut; + framesProcessedIn = 0; + framesProcessedOut = 0; - dst_f32[i] = x; - } + for (;;) { + if (framesProcessedOut >= frameCountOut) { + break; + } - (void)ditherMode; -} + /* Before interpolating we need to load the buffers. When doing this we need to ensure we run every input sample through the filter. */ + while (pResampler->inTimeInt > 0 && frameCountIn > 0) { + ma_uint32 iFilter; + ma_uint32 iChannel; -void ma_pcm_s16_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s16_to_f32__reference(dst, src, count, ditherMode); -} + if (pFramesInF32 != NULL) { + for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) { + pResampler->x0.f32[iChannel] = pResampler->x1.f32[iChannel]; + pResampler->x1.f32[iChannel] = pFramesInF32[iChannel]; + } + pFramesInF32 += pResampler->config.channels; + } else { + for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) { + pResampler->x0.f32[iChannel] = pResampler->x1.f32[iChannel]; + pResampler->x1.f32[iChannel] = 0; + } + } -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_s16_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_s16_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_s16_to_f32__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s16_to_f32__avx2(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_s16_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode); -} -#endif + /* Filter. */ + for (iFilter = 0; iFilter < pResampler->config.lpfCount; iFilter += 1) { + ma_lpf_process_pcm_frame_f32(&pResampler->lpf[iFilter], pResampler->x1.f32, pResampler->x1.f32); + } -void ma_pcm_s16_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_s16_to_f32__reference(dst, src, count, ditherMode); -#else - ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode); -#endif -} + frameCountIn -= 1; + framesProcessedIn += 1; + pResampler->inTimeInt -= 1; + } + if (pResampler->inTimeInt > 0) { + break; /* Ran out of input data. */ + } -void ma_pcm_interleave_s16__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) -{ - ma_int16* dst_s16 = (ma_int16*)dst; - const ma_int16** src_s16 = (const ma_int16**)src; + /* Getting here means the frames have been loaded and filtered and we can generate the next output frame. */ + if (pFramesOutF32 != NULL) { + MA_ASSERT(pResampler->inTimeInt == 0); + ma_linear_resampler_interpolate_frame_f32(pResampler, pFramesOutF32); - ma_uint64 iFrame; - for (iFrame = 0; iFrame < frameCount; iFrame += 1) { - ma_uint32 iChannel; - for (iChannel = 0; iChannel < channels; iChannel += 1) { - dst_s16[iFrame*channels + iChannel] = src_s16[iChannel][iFrame]; + pFramesOutF32 += pResampler->config.channels; + } + + framesProcessedOut += 1; + + /* Advance time forward. */ + pResampler->inTimeInt += pResampler->inAdvanceInt; + pResampler->inTimeFrac += pResampler->inAdvanceFrac; + if (pResampler->inTimeFrac >= pResampler->config.sampleRateOut) { + pResampler->inTimeFrac -= pResampler->config.sampleRateOut; + pResampler->inTimeInt += 1; } } -} -void ma_pcm_interleave_s16__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) -{ - ma_pcm_interleave_s16__reference(dst, src, frameCount, channels); + *pFrameCountIn = framesProcessedIn; + *pFrameCountOut = framesProcessedOut; + + return MA_SUCCESS; } -void ma_pcm_interleave_s16(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) +static ma_result ma_linear_resampler_process_pcm_frames_f32_upsample(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) { -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_interleave_s16__reference(dst, src, frameCount, channels); -#else - ma_pcm_interleave_s16__optimized(dst, src, frameCount, channels); -#endif -} + const float* pFramesInF32; + /* */ float* pFramesOutF32; + ma_uint64 frameCountIn; + ma_uint64 frameCountOut; + ma_uint64 framesProcessedIn; + ma_uint64 framesProcessedOut; + MA_ASSERT(pResampler != NULL); + MA_ASSERT(pFrameCountIn != NULL); + MA_ASSERT(pFrameCountOut != NULL); -void ma_pcm_deinterleave_s16__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) -{ - ma_int16** dst_s16 = (ma_int16**)dst; - const ma_int16* src_s16 = (const ma_int16*)src; + pFramesInF32 = (const float*)pFramesIn; + pFramesOutF32 = ( float*)pFramesOut; + frameCountIn = *pFrameCountIn; + frameCountOut = *pFrameCountOut; + framesProcessedIn = 0; + framesProcessedOut = 0; - ma_uint64 iFrame; - for (iFrame = 0; iFrame < frameCount; iFrame += 1) { - ma_uint32 iChannel; - for (iChannel = 0; iChannel < channels; iChannel += 1) { - dst_s16[iChannel][iFrame] = src_s16[iFrame*channels + iChannel]; - } - } -} + for (;;) { + ma_uint32 iFilter; -void ma_pcm_deinterleave_s16__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) -{ - ma_pcm_deinterleave_s16__reference(dst, src, frameCount, channels); -} + if (framesProcessedOut >= frameCountOut) { + break; + } -void ma_pcm_deinterleave_s16(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) -{ -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_deinterleave_s16__reference(dst, src, frameCount, channels); -#else - ma_pcm_deinterleave_s16__optimized(dst, src, frameCount, channels); -#endif -} + /* Before interpolating we need to load the buffers. */ + while (pResampler->inTimeInt > 0 && frameCountIn > 0) { + ma_uint32 iChannel; + if (pFramesInF32 != NULL) { + for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) { + pResampler->x0.f32[iChannel] = pResampler->x1.f32[iChannel]; + pResampler->x1.f32[iChannel] = pFramesInF32[iChannel]; + } + pFramesInF32 += pResampler->config.channels; + } else { + for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) { + pResampler->x0.f32[iChannel] = pResampler->x1.f32[iChannel]; + pResampler->x1.f32[iChannel] = 0; + } + } -/* s24 */ -void ma_pcm_s24_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_uint8* dst_u8 = (ma_uint8*)dst; - const ma_uint8* src_s24 = (const ma_uint8*)src; + frameCountIn -= 1; + framesProcessedIn += 1; + pResampler->inTimeInt -= 1; + } - if (ditherMode == ma_dither_mode_none) { - ma_uint64 i; - for (i = 0; i < count; i += 1) { - ma_int8 x = (ma_int8)src_s24[i*3 + 2] + 128; - dst_u8[i] = (ma_uint8)x; + if (pResampler->inTimeInt > 0) { + break; /* Ran out of input data. */ } - } else { - ma_uint64 i; - for (i = 0; i < count; i += 1) { - ma_int32 x = (ma_int32)(((ma_uint32)(src_s24[i*3+0]) << 8) | ((ma_uint32)(src_s24[i*3+1]) << 16) | ((ma_uint32)(src_s24[i*3+2])) << 24); - /* Dither. Don't overflow. */ - ma_int32 dither = ma_dither_s32(ditherMode, -0x800000, 0x7FFFFF); - if ((ma_int64)x + dither <= 0x7FFFFFFF) { - x = x + dither; - } else { - x = 0x7FFFFFFF; + /* Getting here means the frames have been loaded and we can generate the next output frame. */ + if (pFramesOutF32 != NULL) { + MA_ASSERT(pResampler->inTimeInt == 0); + ma_linear_resampler_interpolate_frame_f32(pResampler, pFramesOutF32); + + /* Filter. */ + for (iFilter = 0; iFilter < pResampler->config.lpfCount; iFilter += 1) { + ma_lpf_process_pcm_frame_f32(&pResampler->lpf[iFilter], pFramesOutF32, pFramesOutF32); } - - x = x >> 24; - x = x + 128; - dst_u8[i] = (ma_uint8)x; + + pFramesOutF32 += pResampler->config.channels; + } + + framesProcessedOut += 1; + + /* Advance time forward. */ + pResampler->inTimeInt += pResampler->inAdvanceInt; + pResampler->inTimeFrac += pResampler->inAdvanceFrac; + if (pResampler->inTimeFrac >= pResampler->config.sampleRateOut) { + pResampler->inTimeFrac -= pResampler->config.sampleRateOut; + pResampler->inTimeInt += 1; } } + + *pFrameCountIn = framesProcessedIn; + *pFrameCountOut = framesProcessedOut; + + return MA_SUCCESS; } -void ma_pcm_s24_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +static ma_result ma_linear_resampler_process_pcm_frames_f32(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) { - ma_pcm_s24_to_u8__reference(dst, src, count, ditherMode); + MA_ASSERT(pResampler != NULL); + + if (pResampler->config.sampleRateIn > pResampler->config.sampleRateOut) { + return ma_linear_resampler_process_pcm_frames_f32_downsample(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); + } else { + return ma_linear_resampler_process_pcm_frames_f32_upsample(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); + } } -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_s24_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +ma_result ma_linear_resampler_process_pcm_frames(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) { - ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode); + if (pResampler == NULL) { + return MA_INVALID_ARGS; + } + + /* */ if (pResampler->config.format == ma_format_s16) { + return ma_linear_resampler_process_pcm_frames_s16(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); + } else if (pResampler->config.format == ma_format_f32) { + return ma_linear_resampler_process_pcm_frames_f32(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); + } else { + /* Should never get here. Getting here means the format is not supported and you didn't check the return value of ma_linear_resampler_init(). */ + MA_ASSERT(MA_FALSE); + return MA_INVALID_ARGS; + } } -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_s24_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + + +ma_result ma_linear_resampler_set_rate(ma_linear_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut) { - ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode); + return ma_linear_resampler_set_rate_internal(pResampler, sampleRateIn, sampleRateOut, /* isResamplerAlreadyInitialized = */ MA_TRUE); } -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_s24_to_u8__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +ma_result ma_linear_resampler_set_rate_ratio(ma_linear_resampler* pResampler, float ratioInOut) { - ma_pcm_s24_to_u8__avx2(dst, src, count, ditherMode); + ma_uint32 n; + ma_uint32 d; + + d = 1000000; /* We use up to 6 decimal places. */ + n = (ma_uint32)(ratioInOut * d); + + if (n == 0) { + return MA_INVALID_ARGS; /* Ratio too small. */ + } + + MA_ASSERT(n != 0); + + return ma_linear_resampler_set_rate(pResampler, n, d); } -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_s24_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + + +ma_uint64 ma_linear_resampler_get_required_input_frame_count(ma_linear_resampler* pResampler, ma_uint64 outputFrameCount) { - ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode); + ma_uint64 count; + + if (pResampler == NULL) { + return 0; + } + + if (outputFrameCount == 0) { + return 0; + } + + /* Any whole input frames are consumed before the first output frame is generated. */ + count = pResampler->inTimeInt; + outputFrameCount -= 1; + + /* The rest of the output frames can be calculated in constant time. */ + count += outputFrameCount * pResampler->inAdvanceInt; + count += (pResampler->inTimeFrac + (outputFrameCount * pResampler->inAdvanceFrac)) / pResampler->config.sampleRateOut; + + return count; } -#endif -void ma_pcm_s24_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +ma_uint64 ma_linear_resampler_get_expected_output_frame_count(ma_linear_resampler* pResampler, ma_uint64 inputFrameCount) { -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_s24_to_u8__reference(dst, src, count, ditherMode); -#else - ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode); -#endif -} + ma_uint64 outputFrameCount; + ma_uint64 inTimeInt; + ma_uint64 inTimeFrac; + if (pResampler == NULL) { + return 0; + } -void ma_pcm_s24_to_s16__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_int16* dst_s16 = (ma_int16*)dst; - const ma_uint8* src_s24 = (const ma_uint8*)src; + /* TODO: Try making this run in constant time. */ - if (ditherMode == ma_dither_mode_none) { - ma_uint64 i; - for (i = 0; i < count; i += 1) { - ma_uint16 dst_lo = ((ma_uint16)src_s24[i*3 + 1]); - ma_uint16 dst_hi = ((ma_uint16)src_s24[i*3 + 2]) << 8; - dst_s16[i] = (ma_int16)dst_lo | dst_hi; + outputFrameCount = 0; + inTimeInt = pResampler->inTimeInt; + inTimeFrac = pResampler->inTimeFrac; + + for (;;) { + while (inTimeInt > 0 && inputFrameCount > 0) { + inputFrameCount -= 1; + inTimeInt -= 1; + } + + if (inTimeInt > 0) { + break; } - } else { - ma_uint64 i; - for (i = 0; i < count; i += 1) { - ma_int32 x = (ma_int32)(((ma_uint32)(src_s24[i*3+0]) << 8) | ((ma_uint32)(src_s24[i*3+1]) << 16) | ((ma_uint32)(src_s24[i*3+2])) << 24); - /* Dither. Don't overflow. */ - ma_int32 dither = ma_dither_s32(ditherMode, -0x8000, 0x7FFF); - if ((ma_int64)x + dither <= 0x7FFFFFFF) { - x = x + dither; - } else { - x = 0x7FFFFFFF; - } + outputFrameCount += 1; - x = x >> 16; - dst_s16[i] = (ma_int16)x; + /* Advance time forward. */ + inTimeInt += pResampler->inAdvanceInt; + inTimeFrac += pResampler->inAdvanceFrac; + if (inTimeFrac >= pResampler->config.sampleRateOut) { + inTimeFrac -= pResampler->config.sampleRateOut; + inTimeInt += 1; } } -} -void ma_pcm_s24_to_s16__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s24_to_s16__reference(dst, src, count, ditherMode); + return outputFrameCount; } -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_s24_to_s16__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +ma_uint64 ma_linear_resampler_get_input_latency(ma_linear_resampler* pResampler) { - ma_pcm_s24_to_s16__optimized(dst, src, count, ditherMode); + ma_uint32 latency; + ma_uint32 iFilter; + + if (pResampler == NULL) { + return 0; + } + + latency = 1; + for (iFilter = 0; iFilter < pResampler->config.lpfCount; iFilter += 1) { + latency += ma_lpf_get_latency(&pResampler->lpf[iFilter]); + } + + return latency; } -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_s24_to_s16__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +ma_uint64 ma_linear_resampler_get_output_latency(ma_linear_resampler* pResampler) { - ma_pcm_s24_to_s16__optimized(dst, src, count, ditherMode); + if (pResampler == NULL) { + return 0; + } + + return ma_linear_resampler_get_input_latency(pResampler) * pResampler->config.sampleRateOut / pResampler->config.sampleRateIn; } -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_s24_to_s16__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + + +#if defined(ma_speex_resampler_h) +#define MA_HAS_SPEEX_RESAMPLER + +static ma_result ma_result_from_speex_err(int err) { - ma_pcm_s24_to_s16__avx2(dst, src, count, ditherMode); + switch (err) + { + case RESAMPLER_ERR_SUCCESS: return MA_SUCCESS; + case RESAMPLER_ERR_ALLOC_FAILED: return MA_OUT_OF_MEMORY; + case RESAMPLER_ERR_BAD_STATE: return MA_ERROR; + case RESAMPLER_ERR_INVALID_ARG: return MA_INVALID_ARGS; + case RESAMPLER_ERR_PTR_OVERLAP: return MA_INVALID_ARGS; + case RESAMPLER_ERR_OVERFLOW: return MA_ERROR; + default: return MA_ERROR; + } } -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_s24_to_s16__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +#endif /* ma_speex_resampler_h */ + +ma_resampler_config ma_resampler_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut, ma_resample_algorithm algorithm) { - ma_pcm_s24_to_s16__optimized(dst, src, count, ditherMode); + ma_resampler_config config; + + MA_ZERO_OBJECT(&config); + config.format = format; + config.channels = channels; + config.sampleRateIn = sampleRateIn; + config.sampleRateOut = sampleRateOut; + config.algorithm = algorithm; + + /* Linear. */ + config.linear.lpfCount = 1; + config.linear.lpfNyquistFactor = 1; + + /* Speex. */ + config.speex.quality = 3; /* Cannot leave this as 0 as that is actually a valid value for Speex resampling quality. */ + + return config; } -#endif -void ma_pcm_s24_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +ma_result ma_resampler_init(const ma_resampler_config* pConfig, ma_resampler* pResampler) { -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_s24_to_s16__reference(dst, src, count, ditherMode); -#else - ma_pcm_s24_to_s16__optimized(dst, src, count, ditherMode); -#endif -} + ma_result result; + if (pResampler == NULL) { + return MA_INVALID_ARGS; + } -void ma_pcm_s24_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - (void)ditherMode; + MA_ZERO_OBJECT(pResampler); - ma_copy_memory_64(dst, src, count * 3); -} + if (pConfig == NULL) { + return MA_INVALID_ARGS; + } + if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) { + return MA_INVALID_ARGS; + } -void ma_pcm_s24_to_s32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_int32* dst_s32 = (ma_int32*)dst; - const ma_uint8* src_s24 = (const ma_uint8*)src; + pResampler->config = *pConfig; - ma_uint64 i; - for (i = 0; i < count; i += 1) { - dst_s32[i] = (ma_int32)(((ma_uint32)(src_s24[i*3+0]) << 8) | ((ma_uint32)(src_s24[i*3+1]) << 16) | ((ma_uint32)(src_s24[i*3+2])) << 24); + switch (pConfig->algorithm) + { + case ma_resample_algorithm_linear: + { + ma_linear_resampler_config linearConfig; + linearConfig = ma_linear_resampler_config_init(pConfig->format, pConfig->channels, pConfig->sampleRateIn, pConfig->sampleRateOut); + linearConfig.lpfCount = pConfig->linear.lpfCount; + linearConfig.lpfNyquistFactor = pConfig->linear.lpfNyquistFactor; + + result = ma_linear_resampler_init(&linearConfig, &pResampler->state.linear); + if (result != MA_SUCCESS) { + return result; + } + } break; + + case ma_resample_algorithm_speex: + { + #if defined(MA_HAS_SPEEX_RESAMPLER) + int speexErr; + pResampler->state.speex.pSpeexResamplerState = speex_resampler_init(pConfig->channels, pConfig->sampleRateIn, pConfig->sampleRateOut, pConfig->speex.quality, &speexErr); + if (pResampler->state.speex.pSpeexResamplerState == NULL) { + return ma_result_from_speex_err(speexErr); + } + #else + /* Speex resampler not available. */ + return MA_NO_BACKEND; + #endif + } break; + + default: return MA_INVALID_ARGS; } - (void)ditherMode; + return MA_SUCCESS; } -void ma_pcm_s24_to_s32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +void ma_resampler_uninit(ma_resampler* pResampler) { - ma_pcm_s24_to_s32__reference(dst, src, count, ditherMode); -} + if (pResampler == NULL) { + return; + } -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_s24_to_s32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s24_to_s32__optimized(dst, src, count, ditherMode); -} + if (pResampler->config.algorithm == ma_resample_algorithm_linear) { + ma_linear_resampler_uninit(&pResampler->state.linear); + } + +#if defined(MA_HAS_SPEEX_RESAMPLER) + if (pResampler->config.algorithm == ma_resample_algorithm_speex) { + speex_resampler_destroy((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState); + } #endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_s24_to_s32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s24_to_s32__optimized(dst, src, count, ditherMode); } -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_s24_to_s32__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +static ma_result ma_resampler_process_pcm_frames__read__linear(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) { - ma_pcm_s24_to_s32__avx2(dst, src, count, ditherMode); + return ma_linear_resampler_process_pcm_frames(&pResampler->state.linear, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); } -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_s24_to_s32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +#if defined(MA_HAS_SPEEX_RESAMPLER) +static ma_result ma_resampler_process_pcm_frames__read__speex(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) { - ma_pcm_s24_to_s32__optimized(dst, src, count, ditherMode); + int speexErr; + ma_uint64 frameCountOut; + ma_uint64 frameCountIn; + ma_uint64 framesProcessedOut; + ma_uint64 framesProcessedIn; + unsigned int framesPerIteration = UINT_MAX; + + MA_ASSERT(pResampler != NULL); + MA_ASSERT(pFramesOut != NULL); + MA_ASSERT(pFrameCountOut != NULL); + MA_ASSERT(pFrameCountIn != NULL); + + /* + Reading from the Speex resampler requires a bit of dancing around for a few reasons. The first thing is that it's frame counts + are in unsigned int's whereas ours is in ma_uint64. We therefore need to run the conversion in a loop. The other, more complicated + problem, is that we need to keep track of the input time, similar to what we do with the linear resampler. The reason we need to + do this is for ma_resampler_get_required_input_frame_count() and ma_resampler_get_expected_output_frame_count(). + */ + frameCountOut = *pFrameCountOut; + frameCountIn = *pFrameCountIn; + framesProcessedOut = 0; + framesProcessedIn = 0; + + while (framesProcessedOut < frameCountOut && framesProcessedIn < frameCountIn) { + unsigned int frameCountInThisIteration; + unsigned int frameCountOutThisIteration; + const void* pFramesInThisIteration; + void* pFramesOutThisIteration; + + frameCountInThisIteration = framesPerIteration; + if ((ma_uint64)frameCountInThisIteration > (frameCountIn - framesProcessedIn)) { + frameCountInThisIteration = (unsigned int)(frameCountIn - framesProcessedIn); + } + + frameCountOutThisIteration = framesPerIteration; + if ((ma_uint64)frameCountOutThisIteration > (frameCountOut - framesProcessedOut)) { + frameCountOutThisIteration = (unsigned int)(frameCountOut - framesProcessedOut); + } + + pFramesInThisIteration = ma_offset_ptr(pFramesIn, framesProcessedIn * ma_get_bytes_per_frame(pResampler->config.format, pResampler->config.channels)); + pFramesOutThisIteration = ma_offset_ptr(pFramesOut, framesProcessedOut * ma_get_bytes_per_frame(pResampler->config.format, pResampler->config.channels)); + + if (pResampler->config.format == ma_format_f32) { + speexErr = speex_resampler_process_interleaved_float((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState, (const float*)pFramesInThisIteration, &frameCountInThisIteration, (float*)pFramesOutThisIteration, &frameCountOutThisIteration); + } else if (pResampler->config.format == ma_format_s16) { + speexErr = speex_resampler_process_interleaved_int((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState, (const spx_int16_t*)pFramesInThisIteration, &frameCountInThisIteration, (spx_int16_t*)pFramesOutThisIteration, &frameCountOutThisIteration); + } else { + /* Format not supported. Should never get here. */ + MA_ASSERT(MA_FALSE); + return MA_INVALID_OPERATION; + } + + if (speexErr != RESAMPLER_ERR_SUCCESS) { + return ma_result_from_speex_err(speexErr); + } + + framesProcessedIn += frameCountInThisIteration; + framesProcessedOut += frameCountOutThisIteration; + } + + *pFrameCountOut = framesProcessedOut; + *pFrameCountIn = framesProcessedIn; + + return MA_SUCCESS; } #endif -void ma_pcm_s24_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +static ma_result ma_resampler_process_pcm_frames__read(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) { -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_s24_to_s32__reference(dst, src, count, ditherMode); -#else - ma_pcm_s24_to_s32__optimized(dst, src, count, ditherMode); -#endif -} - + MA_ASSERT(pResampler != NULL); + MA_ASSERT(pFramesOut != NULL); -void ma_pcm_s24_to_f32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - float* dst_f32 = (float*)dst; - const ma_uint8* src_s24 = (const ma_uint8*)src; + /* pFramesOut is not NULL, which means we must have a capacity. */ + if (pFrameCountOut == NULL) { + return MA_INVALID_ARGS; + } - ma_uint64 i; - for (i = 0; i < count; i += 1) { - float x = (float)(((ma_int32)(((ma_uint32)(src_s24[i*3+0]) << 8) | ((ma_uint32)(src_s24[i*3+1]) << 16) | ((ma_uint32)(src_s24[i*3+2])) << 24)) >> 8); + /* It doesn't make sense to not have any input frames to process. */ + if (pFrameCountIn == NULL || pFramesIn == NULL) { + return MA_INVALID_ARGS; + } -#if 0 - /* The accurate way. */ - x = x + 8388608.0f; /* -8388608..8388607 to 0..16777215 */ - x = x * 0.00000011920929665621f; /* 0..16777215 to 0..2 */ - x = x - 1; /* 0..2 to -1..1 */ -#else - /* The fast way. */ - x = x * 0.00000011920928955078125f; /* -8388608..8388607 to -1..0.999969482421875 */ -#endif + switch (pResampler->config.algorithm) + { + case ma_resample_algorithm_linear: + { + return ma_resampler_process_pcm_frames__read__linear(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); + } - dst_f32[i] = x; + case ma_resample_algorithm_speex: + { + #if defined(MA_HAS_SPEEX_RESAMPLER) + return ma_resampler_process_pcm_frames__read__speex(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); + #else + break; + #endif + } + + default: break; } - (void)ditherMode; + /* Should never get here. */ + MA_ASSERT(MA_FALSE); + return MA_INVALID_ARGS; } -void ma_pcm_s24_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +static ma_result ma_resampler_process_pcm_frames__seek__linear(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, ma_uint64* pFrameCountOut) { - ma_pcm_s24_to_f32__reference(dst, src, count, ditherMode); + MA_ASSERT(pResampler != NULL); + + /* Seeking is supported natively by the linear resampler. */ + return ma_linear_resampler_process_pcm_frames(&pResampler->state.linear, pFramesIn, pFrameCountIn, NULL, pFrameCountOut); } -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_s24_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +#if defined(MA_HAS_SPEEX_RESAMPLER) +static ma_result ma_resampler_process_pcm_frames__seek__speex(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, ma_uint64* pFrameCountOut) { - ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode); + /* The generic seek method is implemented in on top of ma_resampler_process_pcm_frames__read() by just processing into a dummy buffer. */ + float devnull[8192]; + ma_uint64 totalOutputFramesToProcess; + ma_uint64 totalOutputFramesProcessed; + ma_uint64 totalInputFramesProcessed; + ma_uint32 bpf; + ma_result result; + + MA_ASSERT(pResampler != NULL); + + totalOutputFramesProcessed = 0; + totalInputFramesProcessed = 0; + bpf = ma_get_bytes_per_frame(pResampler->config.format, pResampler->config.channels); + + if (pFrameCountOut != NULL) { + /* Seek by output frames. */ + totalOutputFramesToProcess = *pFrameCountOut; + } else { + /* Seek by input frames. */ + MA_ASSERT(pFrameCountIn != NULL); + totalOutputFramesToProcess = ma_resampler_get_expected_output_frame_count(pResampler, *pFrameCountIn); + } + + if (pFramesIn != NULL) { + /* Process input data. */ + MA_ASSERT(pFrameCountIn != NULL); + while (totalOutputFramesProcessed < totalOutputFramesToProcess && totalInputFramesProcessed < *pFrameCountIn) { + ma_uint64 inputFramesToProcessThisIteration = (*pFrameCountIn - totalInputFramesProcessed); + ma_uint64 outputFramesToProcessThisIteration = (totalOutputFramesToProcess - totalOutputFramesProcessed); + if (outputFramesToProcessThisIteration > sizeof(devnull) / bpf) { + outputFramesToProcessThisIteration = sizeof(devnull) / bpf; + } + + result = ma_resampler_process_pcm_frames__read(pResampler, ma_offset_ptr(pFramesIn, totalInputFramesProcessed*bpf), &inputFramesToProcessThisIteration, ma_offset_ptr(devnull, totalOutputFramesProcessed*bpf), &outputFramesToProcessThisIteration); + if (result != MA_SUCCESS) { + return result; + } + + totalOutputFramesProcessed += outputFramesToProcessThisIteration; + totalInputFramesProcessed += inputFramesToProcessThisIteration; + } + } else { + /* Don't process input data - just update timing and filter state as if zeroes were passed in. */ + while (totalOutputFramesProcessed < totalOutputFramesToProcess) { + ma_uint64 inputFramesToProcessThisIteration = 16384; + ma_uint64 outputFramesToProcessThisIteration = (totalOutputFramesToProcess - totalOutputFramesProcessed); + if (outputFramesToProcessThisIteration > sizeof(devnull) / bpf) { + outputFramesToProcessThisIteration = sizeof(devnull) / bpf; + } + + result = ma_resampler_process_pcm_frames__read(pResampler, NULL, &inputFramesToProcessThisIteration, ma_offset_ptr(devnull, totalOutputFramesProcessed*bpf), &outputFramesToProcessThisIteration); + if (result != MA_SUCCESS) { + return result; + } + + totalOutputFramesProcessed += outputFramesToProcessThisIteration; + totalInputFramesProcessed += inputFramesToProcessThisIteration; + } + } + + + if (pFrameCountIn != NULL) { + *pFrameCountIn = totalInputFramesProcessed; + } + if (pFrameCountOut != NULL) { + *pFrameCountOut = totalOutputFramesProcessed; + } + + return MA_SUCCESS; } #endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_s24_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +static ma_result ma_resampler_process_pcm_frames__seek(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, ma_uint64* pFrameCountOut) { - ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode); + MA_ASSERT(pResampler != NULL); + + switch (pResampler->config.algorithm) + { + case ma_resample_algorithm_linear: + { + return ma_resampler_process_pcm_frames__seek__linear(pResampler, pFramesIn, pFrameCountIn, pFrameCountOut); + } break; + + case ma_resample_algorithm_speex: + { + #if defined(MA_HAS_SPEEX_RESAMPLER) + return ma_resampler_process_pcm_frames__seek__speex(pResampler, pFramesIn, pFrameCountIn, pFrameCountOut); + #else + break; + #endif + }; + + default: break; + } + + /* Should never hit this. */ + MA_ASSERT(MA_FALSE); + return MA_INVALID_ARGS; } -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_s24_to_f32__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + + +ma_result ma_resampler_process_pcm_frames(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) { - ma_pcm_s24_to_f32__avx2(dst, src, count, ditherMode); + if (pResampler == NULL) { + return MA_INVALID_ARGS; + } + + if (pFrameCountOut == NULL && pFrameCountIn == NULL) { + return MA_INVALID_ARGS; + } + + if (pFramesOut != NULL) { + /* Reading. */ + return ma_resampler_process_pcm_frames__read(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); + } else { + /* Seeking. */ + return ma_resampler_process_pcm_frames__seek(pResampler, pFramesIn, pFrameCountIn, pFrameCountOut); + } } -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_s24_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +ma_result ma_resampler_set_rate(ma_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut) { - ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode); + if (pResampler == NULL) { + return MA_INVALID_ARGS; + } + + if (sampleRateIn == 0 || sampleRateOut == 0) { + return MA_INVALID_ARGS; + } + + pResampler->config.sampleRateIn = sampleRateIn; + pResampler->config.sampleRateOut = sampleRateOut; + + switch (pResampler->config.algorithm) + { + case ma_resample_algorithm_linear: + { + return ma_linear_resampler_set_rate(&pResampler->state.linear, sampleRateIn, sampleRateOut); + } break; + + case ma_resample_algorithm_speex: + { + #if defined(MA_HAS_SPEEX_RESAMPLER) + return ma_result_from_speex_err(speex_resampler_set_rate((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState, sampleRateIn, sampleRateOut)); + #else + break; + #endif + }; + + default: break; + } + + /* Should never get here. */ + MA_ASSERT(MA_FALSE); + return MA_INVALID_OPERATION; } -#endif -void ma_pcm_s24_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +ma_result ma_resampler_set_rate_ratio(ma_resampler* pResampler, float ratio) { -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_s24_to_f32__reference(dst, src, count, ditherMode); -#else - ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode); -#endif -} + if (pResampler == NULL) { + return MA_INVALID_ARGS; + } + if (pResampler->config.algorithm == ma_resample_algorithm_linear) { + return ma_linear_resampler_set_rate_ratio(&pResampler->state.linear, ratio); + } else { + /* Getting here means the backend does not have native support for setting the rate as a ratio so we just do it generically. */ + ma_uint32 n; + ma_uint32 d; -void ma_pcm_interleave_s24__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) -{ - ma_uint8* dst8 = (ma_uint8*)dst; - const ma_uint8** src8 = (const ma_uint8**)src; + d = 1000000; /* We use up to 6 decimal places. */ + n = (ma_uint32)(ratio * d); - ma_uint64 iFrame; - for (iFrame = 0; iFrame < frameCount; iFrame += 1) { - ma_uint32 iChannel; - for (iChannel = 0; iChannel < channels; iChannel += 1) { - dst8[iFrame*3*channels + iChannel*3 + 0] = src8[iChannel][iFrame*3 + 0]; - dst8[iFrame*3*channels + iChannel*3 + 1] = src8[iChannel][iFrame*3 + 1]; - dst8[iFrame*3*channels + iChannel*3 + 2] = src8[iChannel][iFrame*3 + 2]; + if (n == 0) { + return MA_INVALID_ARGS; /* Ratio too small. */ } + + MA_ASSERT(n != 0); + + return ma_resampler_set_rate(pResampler, n, d); } } -void ma_pcm_interleave_s24__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) +ma_uint64 ma_resampler_get_required_input_frame_count(ma_resampler* pResampler, ma_uint64 outputFrameCount) { - ma_pcm_interleave_s24__reference(dst, src, frameCount, channels); -} + if (pResampler == NULL) { + return 0; + } -void ma_pcm_interleave_s24(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) -{ -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_interleave_s24__reference(dst, src, frameCount, channels); -#else - ma_pcm_interleave_s24__optimized(dst, src, frameCount, channels); -#endif -} + if (outputFrameCount == 0) { + return 0; + } + switch (pResampler->config.algorithm) + { + case ma_resample_algorithm_linear: + { + return ma_linear_resampler_get_required_input_frame_count(&pResampler->state.linear, outputFrameCount); + } -void ma_pcm_deinterleave_s24__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) -{ - ma_uint8** dst8 = (ma_uint8**)dst; - const ma_uint8* src8 = (const ma_uint8*)src; + case ma_resample_algorithm_speex: + { + #if defined(MA_HAS_SPEEX_RESAMPLER) + ma_uint64 count; + int speexErr = ma_speex_resampler_get_required_input_frame_count((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState, outputFrameCount, &count); + if (speexErr != RESAMPLER_ERR_SUCCESS) { + return 0; + } - ma_uint32 iFrame; - for (iFrame = 0; iFrame < frameCount; iFrame += 1) { - ma_uint32 iChannel; - for (iChannel = 0; iChannel < channels; iChannel += 1) { - dst8[iChannel][iFrame*3 + 0] = src8[iFrame*3*channels + iChannel*3 + 0]; - dst8[iChannel][iFrame*3 + 1] = src8[iFrame*3*channels + iChannel*3 + 1]; - dst8[iChannel][iFrame*3 + 2] = src8[iFrame*3*channels + iChannel*3 + 2]; + return count; + #else + break; + #endif } + + default: break; } + + /* Should never get here. */ + MA_ASSERT(MA_FALSE); + return 0; } -void ma_pcm_deinterleave_s24__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) +ma_uint64 ma_resampler_get_expected_output_frame_count(ma_resampler* pResampler, ma_uint64 inputFrameCount) { - ma_pcm_deinterleave_s24__reference(dst, src, frameCount, channels); + if (pResampler == NULL) { + return 0; /* Invalid args. */ + } + + if (inputFrameCount == 0) { + return 0; + } + + switch (pResampler->config.algorithm) + { + case ma_resample_algorithm_linear: + { + return ma_linear_resampler_get_expected_output_frame_count(&pResampler->state.linear, inputFrameCount); + } + + case ma_resample_algorithm_speex: + { + #if defined(MA_HAS_SPEEX_RESAMPLER) + ma_uint64 count; + int speexErr = ma_speex_resampler_get_expected_output_frame_count((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState, inputFrameCount, &count); + if (speexErr != RESAMPLER_ERR_SUCCESS) { + return 0; + } + + return count; + #else + break; + #endif + } + + default: break; + } + + /* Should never get here. */ + MA_ASSERT(MA_FALSE); + return 0; } -void ma_pcm_deinterleave_s24(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) +ma_uint64 ma_resampler_get_input_latency(ma_resampler* pResampler) { -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_deinterleave_s24__reference(dst, src, frameCount, channels); -#else - ma_pcm_deinterleave_s24__optimized(dst, src, frameCount, channels); -#endif -} + if (pResampler == NULL) { + return 0; + } + + switch (pResampler->config.algorithm) + { + case ma_resample_algorithm_linear: + { + return ma_linear_resampler_get_input_latency(&pResampler->state.linear); + } + case ma_resample_algorithm_speex: + { + #if defined(MA_HAS_SPEEX_RESAMPLER) + return (ma_uint64)ma_speex_resampler_get_input_latency((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState); + #else + break; + #endif + } + default: break; + } -/* s32 */ -void ma_pcm_s32_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + /* Should never get here. */ + MA_ASSERT(MA_FALSE); + return 0; +} + +ma_uint64 ma_resampler_get_output_latency(ma_resampler* pResampler) { - ma_uint8* dst_u8 = (ma_uint8*)dst; - const ma_int32* src_s32 = (const ma_int32*)src; + if (pResampler == NULL) { + return 0; + } - if (ditherMode == ma_dither_mode_none) { - ma_uint64 i; - for (i = 0; i < count; i += 1) { - ma_int32 x = src_s32[i]; - x = x >> 24; - x = x + 128; - dst_u8[i] = (ma_uint8)x; + switch (pResampler->config.algorithm) + { + case ma_resample_algorithm_linear: + { + return ma_linear_resampler_get_output_latency(&pResampler->state.linear); } - } else { - ma_uint64 i; - for (i = 0; i < count; i += 1) { - ma_int32 x = src_s32[i]; - /* Dither. Don't overflow. */ - ma_int32 dither = ma_dither_s32(ditherMode, -0x800000, 0x7FFFFF); - if ((ma_int64)x + dither <= 0x7FFFFFFF) { - x = x + dither; - } else { - x = 0x7FFFFFFF; - } - - x = x >> 24; - x = x + 128; - dst_u8[i] = (ma_uint8)x; + case ma_resample_algorithm_speex: + { + #if defined(MA_HAS_SPEEX_RESAMPLER) + return (ma_uint64)ma_speex_resampler_get_output_latency((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState); + #else + break; + #endif } + + default: break; } + + /* Should never get here. */ + MA_ASSERT(MA_FALSE); + return 0; } -void ma_pcm_s32_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s32_to_u8__reference(dst, src, count, ditherMode); +/************************************************************************************************************************************************************** + +Channel Conversion + +**************************************************************************************************************************************************************/ +#ifndef MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT +#define MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT 12 +#endif + +#define MA_PLANE_LEFT 0 +#define MA_PLANE_RIGHT 1 +#define MA_PLANE_FRONT 2 +#define MA_PLANE_BACK 3 +#define MA_PLANE_BOTTOM 4 +#define MA_PLANE_TOP 5 + +float g_maChannelPlaneRatios[MA_CHANNEL_POSITION_COUNT][6] = { + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_NONE */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_MONO */ + { 0.5f, 0.0f, 0.5f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_LEFT */ + { 0.0f, 0.5f, 0.5f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_RIGHT */ + { 0.0f, 0.0f, 1.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_CENTER */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_LFE */ + { 0.5f, 0.0f, 0.0f, 0.5f, 0.0f, 0.0f}, /* MA_CHANNEL_BACK_LEFT */ + { 0.0f, 0.5f, 0.0f, 0.5f, 0.0f, 0.0f}, /* MA_CHANNEL_BACK_RIGHT */ + { 0.25f, 0.0f, 0.75f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_LEFT_CENTER */ + { 0.0f, 0.25f, 0.75f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_RIGHT_CENTER */ + { 0.0f, 0.0f, 0.0f, 1.0f, 0.0f, 0.0f}, /* MA_CHANNEL_BACK_CENTER */ + { 1.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_SIDE_LEFT */ + { 0.0f, 1.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_SIDE_RIGHT */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 1.0f}, /* MA_CHANNEL_TOP_CENTER */ + { 0.33f, 0.0f, 0.33f, 0.0f, 0.0f, 0.34f}, /* MA_CHANNEL_TOP_FRONT_LEFT */ + { 0.0f, 0.0f, 0.5f, 0.0f, 0.0f, 0.5f}, /* MA_CHANNEL_TOP_FRONT_CENTER */ + { 0.0f, 0.33f, 0.33f, 0.0f, 0.0f, 0.34f}, /* MA_CHANNEL_TOP_FRONT_RIGHT */ + { 0.33f, 0.0f, 0.0f, 0.33f, 0.0f, 0.34f}, /* MA_CHANNEL_TOP_BACK_LEFT */ + { 0.0f, 0.0f, 0.0f, 0.5f, 0.0f, 0.5f}, /* MA_CHANNEL_TOP_BACK_CENTER */ + { 0.0f, 0.33f, 0.0f, 0.33f, 0.0f, 0.34f}, /* MA_CHANNEL_TOP_BACK_RIGHT */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_0 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_1 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_2 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_3 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_4 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_5 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_6 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_7 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_8 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_9 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_10 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_11 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_12 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_13 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_14 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_15 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_16 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_17 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_18 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_19 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_20 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_21 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_22 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_23 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_24 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_25 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_26 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_27 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_28 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_29 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_30 */ + { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_31 */ +}; + +float ma_calculate_channel_position_rectangular_weight(ma_channel channelPositionA, ma_channel channelPositionB) +{ + /* + Imagine the following simplified example: You have a single input speaker which is the front/left speaker which you want to convert to + the following output configuration: + + - front/left + - side/left + - back/left + + The front/left output is easy - it the same speaker position so it receives the full contribution of the front/left input. The amount + of contribution to apply to the side/left and back/left speakers, however, is a bit more complicated. + + Imagine the front/left speaker as emitting audio from two planes - the front plane and the left plane. You can think of the front/left + speaker emitting half of it's total volume from the front, and the other half from the left. Since part of it's volume is being emitted + from the left side, and the side/left and back/left channels also emit audio from the left plane, one would expect that they would + receive some amount of contribution from front/left speaker. The amount of contribution depends on how many planes are shared between + the two speakers. Note that in the examples below I've added a top/front/left speaker as an example just to show how the math works + across 3 spatial dimensions. + + The first thing to do is figure out how each speaker's volume is spread over each of plane: + - front/left: 2 planes (front and left) = 1/2 = half it's total volume on each plane + - side/left: 1 plane (left only) = 1/1 = entire volume from left plane + - back/left: 2 planes (back and left) = 1/2 = half it's total volume on each plane + - top/front/left: 3 planes (top, front and left) = 1/3 = one third it's total volume on each plane + + The amount of volume each channel contributes to each of it's planes is what controls how much it is willing to given and take to other + channels on the same plane. The volume that is willing to the given by one channel is multiplied by the volume that is willing to be + taken by the other to produce the final contribution. + */ + + /* Contribution = Sum(Volume to Give * Volume to Take) */ + float contribution = + g_maChannelPlaneRatios[channelPositionA][0] * g_maChannelPlaneRatios[channelPositionB][0] + + g_maChannelPlaneRatios[channelPositionA][1] * g_maChannelPlaneRatios[channelPositionB][1] + + g_maChannelPlaneRatios[channelPositionA][2] * g_maChannelPlaneRatios[channelPositionB][2] + + g_maChannelPlaneRatios[channelPositionA][3] * g_maChannelPlaneRatios[channelPositionB][3] + + g_maChannelPlaneRatios[channelPositionA][4] * g_maChannelPlaneRatios[channelPositionB][4] + + g_maChannelPlaneRatios[channelPositionA][5] * g_maChannelPlaneRatios[channelPositionB][5]; + + return contribution; } -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_s32_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_s32_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_s32_to_u8__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s32_to_u8__avx2(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_s32_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +ma_channel_converter_config ma_channel_converter_config_init(ma_format format, ma_uint32 channelsIn, const ma_channel channelMapIn[MA_MAX_CHANNELS], ma_uint32 channelsOut, const ma_channel channelMapOut[MA_MAX_CHANNELS], ma_channel_mix_mode mixingMode) { - ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode); + ma_channel_converter_config config; + MA_ZERO_OBJECT(&config); + config.format = format; + config.channelsIn = channelsIn; + config.channelsOut = channelsOut; + ma_channel_map_copy(config.channelMapIn, channelMapIn, channelsIn); + ma_channel_map_copy(config.channelMapOut, channelMapOut, channelsOut); + config.mixingMode = mixingMode; + + return config; } -#endif -void ma_pcm_s32_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +static ma_int32 ma_channel_converter_float_to_fp(float x) { -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_s32_to_u8__reference(dst, src, count, ditherMode); -#else - ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode); -#endif + return (ma_int32)(x * (1<> 16; - dst_s16[i] = (ma_int16)x; - } - } else { - ma_uint64 i; - for (i = 0; i < count; i += 1) { - ma_int32 x = src_s32[i]; + if (channelPosition == MA_CHANNEL_NONE || channelPosition == MA_CHANNEL_MONO || channelPosition == MA_CHANNEL_LFE) { + return MA_FALSE; + } - /* Dither. Don't overflow. */ - ma_int32 dither = ma_dither_s32(ditherMode, -0x8000, 0x7FFF); - if ((ma_int64)x + dither <= 0x7FFFFFFF) { - x = x + dither; - } else { - x = 0x7FFFFFFF; - } - - x = x >> 16; - dst_s16[i] = (ma_int16)x; + for (i = 0; i < 6; ++i) { /* Each side of a cube. */ + if (g_maChannelPlaneRatios[channelPosition][i] != 0) { + return MA_TRUE; } } -} -void ma_pcm_s32_to_s16__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s32_to_s16__reference(dst, src, count, ditherMode); + return MA_FALSE; } -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_s32_to_s16__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s32_to_s16__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_s32_to_s16__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s32_to_s16__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_s32_to_s16__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s32_to_s16__avx2(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_s32_to_s16__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +ma_result ma_channel_converter_init(const ma_channel_converter_config* pConfig, ma_channel_converter* pConverter) { - ma_pcm_s32_to_s16__optimized(dst, src, count, ditherMode); -} -#endif + ma_uint32 iChannelIn; + ma_uint32 iChannelOut; -void ma_pcm_s32_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_s32_to_s16__reference(dst, src, count, ditherMode); -#else - ma_pcm_s32_to_s16__optimized(dst, src, count, ditherMode); -#endif -} + if (pConverter == NULL) { + return MA_INVALID_ARGS; + } + MA_ZERO_OBJECT(pConverter); -void ma_pcm_s32_to_s24__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_uint8* dst_s24 = (ma_uint8*)dst; - const ma_int32* src_s32 = (const ma_int32*)src; + if (pConfig == NULL) { + return MA_INVALID_ARGS; + } - ma_uint64 i; - for (i = 0; i < count; i += 1) { - ma_uint32 x = (ma_uint32)src_s32[i]; - dst_s24[i*3+0] = (ma_uint8)((x & 0x0000FF00) >> 8); - dst_s24[i*3+1] = (ma_uint8)((x & 0x00FF0000) >> 16); - dst_s24[i*3+2] = (ma_uint8)((x & 0xFF000000) >> 24); + if (!ma_channel_map_valid(pConfig->channelsIn, pConfig->channelMapIn)) { + return MA_INVALID_ARGS; /* Invalid input channel map. */ + } + if (!ma_channel_map_valid(pConfig->channelsOut, pConfig->channelMapOut)) { + return MA_INVALID_ARGS; /* Invalid output channel map. */ } - (void)ditherMode; /* No dithering for s32 -> s24. */ -} + if (pConfig->format != ma_format_s16 && pConfig->format != ma_format_f32) { + return MA_INVALID_ARGS; /* Invalid format. */ + } -void ma_pcm_s32_to_s24__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s32_to_s24__reference(dst, src, count, ditherMode); -} + pConverter->format = pConfig->format; + pConverter->channelsIn = pConfig->channelsIn; + pConverter->channelsOut = pConfig->channelsOut; + ma_channel_map_copy(pConverter->channelMapIn, pConfig->channelMapIn, pConfig->channelsIn); + ma_channel_map_copy(pConverter->channelMapOut, pConfig->channelMapOut, pConfig->channelsOut); + pConverter->mixingMode = pConfig->mixingMode; -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_s32_to_s24__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s32_to_s24__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_s32_to_s24__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s32_to_s24__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_s32_to_s24__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s32_to_s24__avx2(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_s32_to_s24__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s32_to_s24__optimized(dst, src, count, ditherMode); -} -#endif + for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; iChannelIn += 1) { + for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) { + if (pConverter->format == ma_format_s16) { + pConverter->weights.f32[iChannelIn][iChannelOut] = pConfig->weights[iChannelIn][iChannelOut]; + } else { + pConverter->weights.s16[iChannelIn][iChannelOut] = ma_channel_converter_float_to_fp(pConfig->weights[iChannelIn][iChannelOut]); + } + } + } + -void ma_pcm_s32_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_s32_to_s24__reference(dst, src, count, ditherMode); -#else - ma_pcm_s32_to_s24__optimized(dst, src, count, ditherMode); -#endif -} + /* If the input and output channels and channel maps are the same we should use a passthrough. */ + if (pConverter->channelsIn == pConverter->channelsOut) { + if (ma_channel_map_equal(pConverter->channelsIn, pConverter->channelMapIn, pConverter->channelMapOut)) { + pConverter->isPassthrough = MA_TRUE; + } + if (ma_channel_map_blank(pConverter->channelsIn, pConverter->channelMapIn) || ma_channel_map_blank(pConverter->channelsOut, pConverter->channelMapOut)) { + pConverter->isPassthrough = MA_TRUE; + } + } -void ma_pcm_s32_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - (void)ditherMode; - ma_copy_memory_64(dst, src, count * sizeof(ma_int32)); -} + /* + We can use a simple case for expanding the mono channel. This will used when expanding a mono input into any output so long + as no LFE is present in the output. + */ + if (!pConverter->isPassthrough) { + if (pConverter->channelsIn == 1 && pConverter->channelMapIn[0] == MA_CHANNEL_MONO) { + /* Optimal case if no LFE is in the output channel map. */ + pConverter->isSimpleMonoExpansion = MA_TRUE; + if (ma_channel_map_contains_channel_position(pConverter->channelsOut, pConverter->channelMapOut, MA_CHANNEL_LFE)) { + pConverter->isSimpleMonoExpansion = MA_FALSE; + } + } + } + /* Another optimized case is stereo to mono. */ + if (!pConverter->isPassthrough) { + if (pConverter->channelsOut == 1 && pConverter->channelMapOut[0] == MA_CHANNEL_MONO && pConverter->channelsIn == 2) { + /* Optimal case if no LFE is in the input channel map. */ + pConverter->isStereoToMono = MA_TRUE; + if (ma_channel_map_contains_channel_position(pConverter->channelsIn, pConverter->channelMapIn, MA_CHANNEL_LFE)) { + pConverter->isStereoToMono = MA_FALSE; + } + } + } -void ma_pcm_s32_to_f32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - float* dst_f32 = (float*)dst; - const ma_int32* src_s32 = (const ma_int32*)src; - ma_uint64 i; - for (i = 0; i < count; i += 1) { - double x = src_s32[i]; + /* + Here is where we do a bit of pre-processing to know how each channel should be combined to make up the output. Rules: + + 1) If it's a passthrough, do nothing - it's just a simple memcpy(). + 2) If the channel counts are the same and every channel position in the input map is present in the output map, use a + simple shuffle. An example might be different 5.1 channel layouts. + 3) Otherwise channels are blended based on spatial locality. + */ + if (!pConverter->isPassthrough) { + if (pConverter->channelsIn == pConverter->channelsOut) { + ma_bool32 areAllChannelPositionsPresent = MA_TRUE; + for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) { + ma_bool32 isInputChannelPositionInOutput = MA_FALSE; + for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) { + if (pConverter->channelMapIn[iChannelIn] == pConverter->channelMapOut[iChannelOut]) { + isInputChannelPositionInOutput = MA_TRUE; + break; + } + } -#if 0 - x = x + 2147483648.0; - x = x * 0.0000000004656612873077392578125; - x = x - 1; -#else - x = x / 2147483648.0; -#endif + if (!isInputChannelPositionInOutput) { + areAllChannelPositionsPresent = MA_FALSE; + break; + } + } - dst_f32[i] = (float)x; - } + if (areAllChannelPositionsPresent) { + pConverter->isSimpleShuffle = MA_TRUE; - (void)ditherMode; /* No dithering for s32 -> f32. */ -} + /* + All the router will be doing is rearranging channels which means all we need to do is use a shuffling table which is just + a mapping between the index of the input channel to the index of the output channel. + */ + for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) { + for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) { + if (pConverter->channelMapIn[iChannelIn] == pConverter->channelMapOut[iChannelOut]) { + pConverter->shuffleTable[iChannelIn] = (ma_uint8)iChannelOut; + break; + } + } + } + } + } + } -void ma_pcm_s32_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s32_to_f32__reference(dst, src, count, ditherMode); -} -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_s32_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_s32_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_s32_to_f32__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s32_to_f32__avx2(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_s32_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode); -} -#endif + /* + Here is where weights are calculated. Note that we calculate the weights at all times, even when using a passthrough and simple + shuffling. We use different algorithms for calculating weights depending on our mixing mode. + + In simple mode we don't do any blending (except for converting between mono, which is done in a later step). Instead we just + map 1:1 matching channels. In this mode, if no channels in the input channel map correspond to anything in the output channel + map, nothing will be heard! + */ -void ma_pcm_s32_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_s32_to_f32__reference(dst, src, count, ditherMode); -#else - ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode); -#endif -} + /* In all cases we need to make sure all channels that are present in both channel maps have a 1:1 mapping. */ + for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) { + ma_channel channelPosIn = pConverter->channelMapIn[iChannelIn]; + for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) { + ma_channel channelPosOut = pConverter->channelMapOut[iChannelOut]; -void ma_pcm_interleave_s32__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) -{ - ma_int32* dst_s32 = (ma_int32*)dst; - const ma_int32** src_s32 = (const ma_int32**)src; + if (channelPosIn == channelPosOut) { + if (pConverter->format == ma_format_s16) { + pConverter->weights.s16[iChannelIn][iChannelOut] = (1 << MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT); + } else { + pConverter->weights.f32[iChannelIn][iChannelOut] = 1; + } + } + } + } - ma_uint64 iFrame; - for (iFrame = 0; iFrame < frameCount; iFrame += 1) { - ma_uint32 iChannel; - for (iChannel = 0; iChannel < channels; iChannel += 1) { - dst_s32[iFrame*channels + iChannel] = src_s32[iChannel][iFrame]; + /* + The mono channel is accumulated on all other channels, except LFE. Make sure in this loop we exclude output mono channels since + they were handled in the pass above. + */ + for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) { + ma_channel channelPosIn = pConverter->channelMapIn[iChannelIn]; + + if (channelPosIn == MA_CHANNEL_MONO) { + for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) { + ma_channel channelPosOut = pConverter->channelMapOut[iChannelOut]; + + if (channelPosOut != MA_CHANNEL_NONE && channelPosOut != MA_CHANNEL_MONO && channelPosOut != MA_CHANNEL_LFE) { + if (pConverter->format == ma_format_s16) { + pConverter->weights.s16[iChannelIn][iChannelOut] = (1 << MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT); + } else { + pConverter->weights.f32[iChannelIn][iChannelOut] = 1; + } + } + } } } -} -void ma_pcm_interleave_s32__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) -{ - ma_pcm_interleave_s32__reference(dst, src, frameCount, channels); -} + /* The output mono channel is the average of all non-none, non-mono and non-lfe input channels. */ + { + ma_uint32 len = 0; + for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) { + ma_channel channelPosIn = pConverter->channelMapIn[iChannelIn]; -void ma_pcm_interleave_s32(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) -{ -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_interleave_s32__reference(dst, src, frameCount, channels); -#else - ma_pcm_interleave_s32__optimized(dst, src, frameCount, channels); -#endif -} + if (channelPosIn != MA_CHANNEL_NONE && channelPosIn != MA_CHANNEL_MONO && channelPosIn != MA_CHANNEL_LFE) { + len += 1; + } + } + if (len > 0) { + float monoWeight = 1.0f / len; -void ma_pcm_deinterleave_s32__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) -{ - ma_int32** dst_s32 = (ma_int32**)dst; - const ma_int32* src_s32 = (const ma_int32*)src; + for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) { + ma_channel channelPosOut = pConverter->channelMapOut[iChannelOut]; - ma_uint64 iFrame; - for (iFrame = 0; iFrame < frameCount; iFrame += 1) { - ma_uint32 iChannel; - for (iChannel = 0; iChannel < channels; iChannel += 1) { - dst_s32[iChannel][iFrame] = src_s32[iFrame*channels + iChannel]; + if (channelPosOut == MA_CHANNEL_MONO) { + for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) { + ma_channel channelPosIn = pConverter->channelMapIn[iChannelIn]; + + if (channelPosIn != MA_CHANNEL_NONE && channelPosIn != MA_CHANNEL_MONO && channelPosIn != MA_CHANNEL_LFE) { + if (pConverter->format == ma_format_s16) { + pConverter->weights.s16[iChannelIn][iChannelOut] = ma_channel_converter_float_to_fp(monoWeight); + } else { + pConverter->weights.f32[iChannelIn][iChannelOut] = monoWeight; + } + } + } + } + } } } -} -void ma_pcm_deinterleave_s32__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) -{ - ma_pcm_deinterleave_s32__reference(dst, src, frameCount, channels); -} -void ma_pcm_deinterleave_s32(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) -{ -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_deinterleave_s32__reference(dst, src, frameCount, channels); -#else - ma_pcm_deinterleave_s32__optimized(dst, src, frameCount, channels); -#endif -} + /* Input and output channels that are not present on the other side need to be blended in based on spatial locality. */ + switch (pConverter->mixingMode) + { + case ma_channel_mix_mode_rectangular: + { + /* Unmapped input channels. */ + for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) { + ma_channel channelPosIn = pConverter->channelMapIn[iChannelIn]; + if (ma_is_spatial_channel_position(channelPosIn)) { + if (!ma_channel_map_contains_channel_position(pConverter->channelsOut, pConverter->channelMapOut, channelPosIn)) { + for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) { + ma_channel channelPosOut = pConverter->channelMapOut[iChannelOut]; -/* f32 */ -void ma_pcm_f32_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_uint64 i; + if (ma_is_spatial_channel_position(channelPosOut)) { + float weight = 0; + if (pConverter->mixingMode == ma_channel_mix_mode_rectangular) { + weight = ma_calculate_channel_position_rectangular_weight(channelPosIn, channelPosOut); + } - ma_uint8* dst_u8 = (ma_uint8*)dst; - const float* src_f32 = (const float*)src; + /* Only apply the weight if we haven't already got some contribution from the respective channels. */ + if (pConverter->format == ma_format_s16) { + if (pConverter->weights.s16[iChannelIn][iChannelOut] == 0) { + pConverter->weights.s16[iChannelIn][iChannelOut] = ma_channel_converter_float_to_fp(weight); + } + } else { + if (pConverter->weights.f32[iChannelIn][iChannelOut] == 0) { + pConverter->weights.f32[iChannelIn][iChannelOut] = weight; + } + } + } + } + } + } + } - float ditherMin = 0; - float ditherMax = 0; - if (ditherMode != ma_dither_mode_none) { - ditherMin = 1.0f / -128; - ditherMax = 1.0f / 127; - } + /* Unmapped output channels. */ + for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) { + ma_channel channelPosOut = pConverter->channelMapOut[iChannelOut]; - for (i = 0; i < count; i += 1) { - float x = src_f32[i]; - x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax); - x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */ - x = x + 1; /* -1..1 to 0..2 */ - x = x * 127.5f; /* 0..2 to 0..255 */ + if (ma_is_spatial_channel_position(channelPosOut)) { + if (!ma_channel_map_contains_channel_position(pConverter->channelsIn, pConverter->channelMapIn, channelPosOut)) { + for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) { + ma_channel channelPosIn = pConverter->channelMapIn[iChannelIn]; - dst_u8[i] = (ma_uint8)x; + if (ma_is_spatial_channel_position(channelPosIn)) { + float weight = 0; + if (pConverter->mixingMode == ma_channel_mix_mode_rectangular) { + weight = ma_calculate_channel_position_rectangular_weight(channelPosIn, channelPosOut); + } + + /* Only apply the weight if we haven't already got some contribution from the respective channels. */ + if (pConverter->format == ma_format_s16) { + if (pConverter->weights.s16[iChannelIn][iChannelOut] == 0) { + pConverter->weights.s16[iChannelIn][iChannelOut] = ma_channel_converter_float_to_fp(weight); + } + } else { + if (pConverter->weights.f32[iChannelIn][iChannelOut] == 0) { + pConverter->weights.f32[iChannelIn][iChannelOut] = weight; + } + } + } + } + } + } + } + } break; + + case ma_channel_mix_mode_custom_weights: + case ma_channel_mix_mode_simple: + default: + { + /* Fallthrough. */ + } break; } -} -void ma_pcm_f32_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_f32_to_u8__reference(dst, src, count, ditherMode); -} -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_f32_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_f32_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode); + return MA_SUCCESS; } -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_f32_to_u8__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +void ma_channel_converter_uninit(ma_channel_converter* pConverter) { - ma_pcm_f32_to_u8__avx2(dst, src, count, ditherMode); + if (pConverter == NULL) { + return; + } } -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_f32_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +static ma_result ma_channel_converter_process_pcm_frames__passthrough(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount) { - ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode); + MA_ASSERT(pConverter != NULL); + MA_ASSERT(pFramesOut != NULL); + MA_ASSERT(pFramesIn != NULL); + + ma_copy_memory_64(pFramesOut, pFramesIn, frameCount * ma_get_bytes_per_frame(pConverter->format, pConverter->channelsOut)); + return MA_SUCCESS; } -#endif -void ma_pcm_f32_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +static ma_result ma_channel_converter_process_pcm_frames__simple_shuffle(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount) { -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_f32_to_u8__reference(dst, src, count, ditherMode); -#else - ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode); -#endif -} + ma_uint32 iFrame; + ma_uint32 iChannelIn; + MA_ASSERT(pConverter != NULL); + MA_ASSERT(pFramesOut != NULL); + MA_ASSERT(pFramesIn != NULL); + MA_ASSERT(pConverter->channelsIn == pConverter->channelsOut); -void ma_pcm_f32_to_s16__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_uint64 i; + if (pConverter->format == ma_format_s16) { + /* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut; + const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn; - ma_int16* dst_s16 = (ma_int16*)dst; - const float* src_f32 = (const float*)src; + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) { + pFramesOutS16[pConverter->shuffleTable[iChannelIn]] = pFramesInS16[iChannelIn]; + } + } + } else { + /* */ float* pFramesOutF32 = ( float*)pFramesOut; + const float* pFramesInF32 = (const float*)pFramesIn; - float ditherMin = 0; - float ditherMax = 0; - if (ditherMode != ma_dither_mode_none) { - ditherMin = 1.0f / -32768; - ditherMax = 1.0f / 32767; + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) { + pFramesOutF32[pConverter->shuffleTable[iChannelIn]] = pFramesInF32[iChannelIn]; + } + } } - for (i = 0; i < count; i += 1) { - float x = src_f32[i]; - x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax); - x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */ + return MA_SUCCESS; +} -#if 0 - /* The accurate way. */ - x = x + 1; /* -1..1 to 0..2 */ - x = x * 32767.5f; /* 0..2 to 0..65535 */ - x = x - 32768.0f; /* 0...65535 to -32768..32767 */ -#else - /* The fast way. */ - x = x * 32767.0f; /* -1..1 to -32767..32767 */ -#endif +static ma_result ma_channel_converter_process_pcm_frames__simple_mono_expansion(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount) +{ + ma_uint64 iFrame; - dst_s16[i] = (ma_int16)x; + MA_ASSERT(pConverter != NULL); + MA_ASSERT(pFramesOut != NULL); + MA_ASSERT(pFramesIn != NULL); + + if (pConverter->format == ma_format_s16) { + /* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut; + const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn; + + if (pConverter->channelsOut == 2) { + for (iFrame = 0; iFrame < frameCount; ++iFrame) { + pFramesOutS16[iFrame*2 + 0] = pFramesInS16[iFrame]; + pFramesOutS16[iFrame*2 + 1] = pFramesInS16[iFrame]; + } + } else { + for (iFrame = 0; iFrame < frameCount; ++iFrame) { + ma_uint32 iChannel; + for (iChannel = 0; iChannel < pConverter->channelsOut; iChannel += 1) { + pFramesOutS16[iFrame*pConverter->channelsOut + iChannel] = pFramesInS16[iFrame]; + } + } + } + } else { + /* */ float* pFramesOutF32 = ( float*)pFramesOut; + const float* pFramesInF32 = (const float*)pFramesIn; + + if (pConverter->channelsOut == 2) { + for (iFrame = 0; iFrame < frameCount; ++iFrame) { + pFramesOutF32[iFrame*2 + 0] = pFramesInF32[iFrame]; + pFramesOutF32[iFrame*2 + 1] = pFramesInF32[iFrame]; + } + } else { + for (iFrame = 0; iFrame < frameCount; ++iFrame) { + ma_uint32 iChannel; + for (iChannel = 0; iChannel < pConverter->channelsOut; iChannel += 1) { + pFramesOutF32[iFrame*pConverter->channelsOut + iChannel] = pFramesInF32[iFrame]; + } + } + } } + + return MA_SUCCESS; } -void ma_pcm_f32_to_s16__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +static ma_result ma_channel_converter_process_pcm_frames__stereo_to_mono(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount) { - ma_uint64 i; - ma_uint64 i4; - ma_uint64 count4; + ma_uint64 iFrame; - ma_int16* dst_s16 = (ma_int16*)dst; - const float* src_f32 = (const float*)src; + MA_ASSERT(pConverter != NULL); + MA_ASSERT(pFramesOut != NULL); + MA_ASSERT(pFramesIn != NULL); + MA_ASSERT(pConverter->channelsIn == 2); + MA_ASSERT(pConverter->channelsOut == 1); - float ditherMin = 0; - float ditherMax = 0; - if (ditherMode != ma_dither_mode_none) { - ditherMin = 1.0f / -32768; - ditherMax = 1.0f / 32767; + if (pConverter->format == ma_format_s16) { + /* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut; + const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn; + + for (iFrame = 0; iFrame < frameCount; ++iFrame) { + pFramesOutS16[iFrame] = (ma_int16)(((ma_int32)pFramesInS16[iFrame*2+0] + (ma_int32)pFramesInS16[iFrame*2+1]) / 2); + } + } else { + /* */ float* pFramesOutF32 = ( float*)pFramesOut; + const float* pFramesInF32 = (const float*)pFramesIn; + + for (iFrame = 0; iFrame < frameCount; ++iFrame) { + pFramesOutF32[iFrame] = (pFramesInF32[iFrame*2+0] + pFramesInF32[iFrame*2+0]) * 0.5f; + } } - /* Unrolled. */ - i = 0; - count4 = count >> 2; - for (i4 = 0; i4 < count4; i4 += 1) { - float d0 = ma_dither_f32(ditherMode, ditherMin, ditherMax); - float d1 = ma_dither_f32(ditherMode, ditherMin, ditherMax); - float d2 = ma_dither_f32(ditherMode, ditherMin, ditherMax); - float d3 = ma_dither_f32(ditherMode, ditherMin, ditherMax); - - float x0 = src_f32[i+0]; - float x1 = src_f32[i+1]; - float x2 = src_f32[i+2]; - float x3 = src_f32[i+3]; + return MA_SUCCESS; +} + +static ma_result ma_channel_converter_process_pcm_frames__weights(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount) +{ + ma_uint32 iFrame; + ma_uint32 iChannelIn; + ma_uint32 iChannelOut; - x0 = x0 + d0; - x1 = x1 + d1; - x2 = x2 + d2; - x3 = x3 + d3; + MA_ASSERT(pConverter != NULL); + MA_ASSERT(pFramesOut != NULL); + MA_ASSERT(pFramesIn != NULL); - x0 = ((x0 < -1) ? -1 : ((x0 > 1) ? 1 : x0)); - x1 = ((x1 < -1) ? -1 : ((x1 > 1) ? 1 : x1)); - x2 = ((x2 < -1) ? -1 : ((x2 > 1) ? 1 : x2)); - x3 = ((x3 < -1) ? -1 : ((x3 > 1) ? 1 : x3)); + /* This is the more complicated case. Each of the output channels is accumulated with 0 or more input channels. */ - x0 = x0 * 32767.0f; - x1 = x1 * 32767.0f; - x2 = x2 * 32767.0f; - x3 = x3 * 32767.0f; + /* Clear. */ + ma_zero_memory_64(pFramesOut, frameCount * ma_get_bytes_per_frame(pConverter->format, pConverter->channelsOut)); - dst_s16[i+0] = (ma_int16)x0; - dst_s16[i+1] = (ma_int16)x1; - dst_s16[i+2] = (ma_int16)x2; - dst_s16[i+3] = (ma_int16)x3; + /* Accumulate. */ + if (pConverter->format == ma_format_s16) { + /* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut; + const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn; - i += 4; - } + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) { + for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) { + ma_int32 s = pFramesOutS16[iFrame*pConverter->channelsOut + iChannelOut]; + s += (pFramesInS16[iFrame*pConverter->channelsIn + iChannelIn] * pConverter->weights.s16[iChannelIn][iChannelOut]) >> MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT; - /* Leftover. */ - for (; i < count; i += 1) { - float x = src_f32[i]; - x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax); - x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */ - x = x * 32767.0f; /* -1..1 to -32767..32767 */ + pFramesOutS16[iFrame*pConverter->channelsOut + iChannelOut] = (ma_int16)ma_clamp(s, -32768, 32767); + } + } + } + } else { + /* */ float* pFramesOutF32 = ( float*)pFramesOut; + const float* pFramesInF32 = (const float*)pFramesIn; - dst_s16[i] = (ma_int16)x; + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) { + for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) { + pFramesOutF32[iFrame*pConverter->channelsOut + iChannelOut] += pFramesInF32[iFrame*pConverter->channelsIn + iChannelIn] * pConverter->weights.f32[iChannelIn][iChannelOut]; + } + } + } } + + return MA_SUCCESS; } -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_f32_to_s16__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +ma_result ma_channel_converter_process_pcm_frames(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount) { - ma_uint64 i; - ma_uint64 i8; - ma_uint64 count8; - ma_int16* dst_s16; - const float* src_f32; - float ditherMin; - float ditherMax; + if (pConverter == NULL) { + return MA_INVALID_ARGS; + } - /* Both the input and output buffers need to be aligned to 16 bytes. */ - if ((((ma_uintptr)dst & 15) != 0) || (((ma_uintptr)src & 15) != 0)) { - ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode); - return; + if (pFramesOut == NULL) { + return MA_INVALID_ARGS; } - dst_s16 = (ma_int16*)dst; - src_f32 = (const float*)src; + if (pFramesIn == NULL) { + ma_zero_memory_64(pFramesOut, frameCount * ma_get_bytes_per_frame(pConverter->format, pConverter->channelsOut)); + return MA_SUCCESS; + } - ditherMin = 0; - ditherMax = 0; - if (ditherMode != ma_dither_mode_none) { - ditherMin = 1.0f / -32768; - ditherMax = 1.0f / 32767; + if (pConverter->isPassthrough) { + return ma_channel_converter_process_pcm_frames__passthrough(pConverter, pFramesOut, pFramesIn, frameCount); + } else if (pConverter->isSimpleShuffle) { + return ma_channel_converter_process_pcm_frames__simple_shuffle(pConverter, pFramesOut, pFramesIn, frameCount); + } else if (pConverter->isSimpleMonoExpansion) { + return ma_channel_converter_process_pcm_frames__simple_mono_expansion(pConverter, pFramesOut, pFramesIn, frameCount); + } else if (pConverter->isStereoToMono) { + return ma_channel_converter_process_pcm_frames__stereo_to_mono(pConverter, pFramesOut, pFramesIn, frameCount); + } else { + return ma_channel_converter_process_pcm_frames__weights(pConverter, pFramesOut, pFramesIn, frameCount); } +} - i = 0; - /* SSE2. SSE allows us to output 8 s16's at a time which means our loop is unrolled 8 times. */ - count8 = count >> 3; - for (i8 = 0; i8 < count8; i8 += 1) { - __m128 d0; - __m128 d1; - __m128 x0; - __m128 x1; +/************************************************************************************************************************************************************** - if (ditherMode == ma_dither_mode_none) { - d0 = _mm_set1_ps(0); - d1 = _mm_set1_ps(0); - } else if (ditherMode == ma_dither_mode_rectangle) { - d0 = _mm_set_ps( - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax) - ); - d1 = _mm_set_ps( - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax) - ); - } else { - d0 = _mm_set_ps( - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax) - ); - d1 = _mm_set_ps( - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax) - ); - } +Data Conversion - x0 = *((__m128*)(src_f32 + i) + 0); - x1 = *((__m128*)(src_f32 + i) + 1); +**************************************************************************************************************************************************************/ +ma_data_converter_config ma_data_converter_config_init_default() +{ + ma_data_converter_config config; + MA_ZERO_OBJECT(&config); - x0 = _mm_add_ps(x0, d0); - x1 = _mm_add_ps(x1, d1); + config.ditherMode = ma_dither_mode_none; + config.resampling.algorithm = ma_resample_algorithm_linear; + config.resampling.allowDynamicSampleRate = MA_FALSE; /* Disable dynamic sample rates by default because dynamic rate adjustments should be quite rare and it allows an optimization for cases when the in and out sample rates are the same. */ - x0 = _mm_mul_ps(x0, _mm_set1_ps(32767.0f)); - x1 = _mm_mul_ps(x1, _mm_set1_ps(32767.0f)); + /* Linear resampling defaults. */ + config.resampling.linear.lpfCount = 1; + config.resampling.linear.lpfNyquistFactor = 1; - _mm_stream_si128(((__m128i*)(dst_s16 + i)), _mm_packs_epi32(_mm_cvttps_epi32(x0), _mm_cvttps_epi32(x1))); - - i += 8; + /* Speex resampling defaults. */ + config.resampling.speex.quality = 3; + + return config; +} + +ma_data_converter_config ma_data_converter_config_init(ma_format formatIn, ma_format formatOut, ma_uint32 channelsIn, ma_uint32 channelsOut, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut) +{ + ma_data_converter_config config = ma_data_converter_config_init_default(); + config.formatIn = formatIn; + config.formatOut = formatOut; + config.channelsIn = channelsIn; + config.channelsOut = channelsOut; + config.sampleRateIn = sampleRateIn; + config.sampleRateOut = sampleRateOut; + + return config; +} + +ma_result ma_data_converter_init(const ma_data_converter_config* pConfig, ma_data_converter* pConverter) +{ + ma_result result; + ma_format midFormat; + + if (pConverter == NULL) { + return MA_INVALID_ARGS; } + MA_ZERO_OBJECT(pConverter); - /* Leftover. */ - for (; i < count; i += 1) { - float x = src_f32[i]; - x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax); - x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */ - x = x * 32767.0f; /* -1..1 to -32767..32767 */ + if (pConfig == NULL) { + return MA_INVALID_ARGS; + } - dst_s16[i] = (ma_int16)x; + pConverter->config = *pConfig; + + /* + We want to avoid as much data conversion as possible. The channel converter and resampler both support s16 and f32 natively. We need to decide + on the format to use for this stage. We call this the mid format because it's used in the middle stage of the conversion pipeline. If the output + format is either s16 or f32 we use that one. If that is not the case it will do the same thing for the input format. If it's neither we just + use f32. + */ + /* */ if (pConverter->config.formatOut == ma_format_s16 || pConverter->config.formatOut == ma_format_f32) { + midFormat = pConverter->config.formatOut; + } else if (pConverter->config.formatIn == ma_format_s16 || pConverter->config.formatIn == ma_format_f32) { + midFormat = pConverter->config.formatIn; + } else { + midFormat = ma_format_f32; } -} -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_f32_to_s16__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_uint64 i; - ma_uint64 i16; - ma_uint64 count16; - ma_int16* dst_s16; - const float* src_f32; - float ditherMin; - float ditherMax; - /* Both the input and output buffers need to be aligned to 32 bytes. */ - if ((((ma_uintptr)dst & 31) != 0) || (((ma_uintptr)src & 31) != 0)) { - ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode); - return; + if (pConverter->config.formatIn != midFormat) { + pConverter->hasPreFormatConversion = MA_TRUE; + } + if (pConverter->config.formatOut != midFormat) { + pConverter->hasPostFormatConversion = MA_TRUE; } - dst_s16 = (ma_int16*)dst; - src_f32 = (const float*)src; - ditherMin = 0; - ditherMax = 0; - if (ditherMode != ma_dither_mode_none) { - ditherMin = 1.0f / -32768; - ditherMax = 1.0f / 32767; + /* Channel converter. We always initialize this, but we check if it configures itself as a passthrough to determine whether or not it's needed. */ + { + ma_uint32 iChannelIn; + ma_uint32 iChannelOut; + ma_channel_converter_config channelConverterConfig; + + channelConverterConfig = ma_channel_converter_config_init(midFormat, pConverter->config.channelsIn, pConverter->config.channelMapIn, pConverter->config.channelsOut, pConverter->config.channelMapOut, pConverter->config.channelMixMode); + + /* Channel weights. */ + for (iChannelIn = 0; iChannelIn < pConverter->config.channelsIn; iChannelIn += 1) { + for (iChannelOut = 0; iChannelOut < pConverter->config.channelsOut; iChannelOut += 1) { + channelConverterConfig.weights[iChannelIn][iChannelOut] = pConverter->config.channelWeights[iChannelIn][iChannelOut]; + } + } + + result = ma_channel_converter_init(&channelConverterConfig, &pConverter->channelConverter); + if (result != MA_SUCCESS) { + return result; + } + + /* If the channel converter is not a passthrough we need to enable it. Otherwise we can skip it. */ + if (pConverter->channelConverter.isPassthrough == MA_FALSE) { + pConverter->hasChannelConverter = MA_TRUE; + } } - i = 0; - /* AVX2. AVX2 allows us to output 16 s16's at a time which means our loop is unrolled 16 times. */ - count16 = count >> 4; - for (i16 = 0; i16 < count16; i16 += 1) { - __m256 d0; - __m256 d1; - __m256 x0; - __m256 x1; - __m256i i0; - __m256i i1; - __m256i p0; - __m256i p1; - __m256i r; + /* Always enable dynamic sample rates if the input sample rate is different because we're always going to need a resampler in this case anyway. */ + if (pConverter->config.resampling.allowDynamicSampleRate == MA_FALSE) { + pConverter->config.resampling.allowDynamicSampleRate = pConverter->config.sampleRateIn != pConverter->config.sampleRateOut; + } - if (ditherMode == ma_dither_mode_none) { - d0 = _mm256_set1_ps(0); - d1 = _mm256_set1_ps(0); - } else if (ditherMode == ma_dither_mode_rectangle) { - d0 = _mm256_set_ps( - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax) - ); - d1 = _mm256_set_ps( - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax), - ma_dither_f32_rectangle(ditherMin, ditherMax) - ); + /* Resampler. */ + if (pConverter->config.resampling.allowDynamicSampleRate) { + ma_resampler_config resamplerConfig; + ma_uint32 resamplerChannels; + + /* The resampler is the most expensive part of the conversion process, so we need to do it at the stage where the channel count is at it's lowest. */ + if (pConverter->config.channelsIn < pConverter->config.channelsOut) { + resamplerChannels = pConverter->config.channelsIn; } else { - d0 = _mm256_set_ps( - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax) - ); - d1 = _mm256_set_ps( - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax), - ma_dither_f32_triangle(ditherMin, ditherMax) - ); + resamplerChannels = pConverter->config.channelsOut; } - x0 = *((__m256*)(src_f32 + i) + 0); - x1 = *((__m256*)(src_f32 + i) + 1); + resamplerConfig = ma_resampler_config_init(midFormat, resamplerChannels, pConverter->config.sampleRateIn, pConverter->config.sampleRateOut, pConverter->config.resampling.algorithm); + resamplerConfig.linear.lpfCount = pConverter->config.resampling.linear.lpfCount; + resamplerConfig.linear.lpfNyquistFactor = pConverter->config.resampling.linear.lpfNyquistFactor; + resamplerConfig.speex.quality = pConverter->config.resampling.speex.quality; + + result = ma_resampler_init(&resamplerConfig, &pConverter->resampler); + if (result != MA_SUCCESS) { + return result; + } + + pConverter->hasResampler = MA_TRUE; + } + + /* We can enable passthrough optimizations if applicable. Note that we'll only be able to do this if the sample rate is static. */ + if (pConverter->hasPreFormatConversion == MA_FALSE && + pConverter->hasPostFormatConversion == MA_FALSE && + pConverter->hasChannelConverter == MA_FALSE && + pConverter->hasResampler == MA_FALSE) { + pConverter->isPassthrough = MA_TRUE; + } + + return MA_SUCCESS; +} - x0 = _mm256_add_ps(x0, d0); - x1 = _mm256_add_ps(x1, d1); +void ma_data_converter_uninit(ma_data_converter* pConverter) +{ + if (pConverter == NULL) { + return; + } - x0 = _mm256_mul_ps(x0, _mm256_set1_ps(32767.0f)); - x1 = _mm256_mul_ps(x1, _mm256_set1_ps(32767.0f)); + if (pConverter->hasResampler) { + ma_resampler_uninit(&pConverter->resampler); + } +} - /* Computing the final result is a little more complicated for AVX2 than SSE2. */ - i0 = _mm256_cvttps_epi32(x0); - i1 = _mm256_cvttps_epi32(x1); - p0 = _mm256_permute2x128_si256(i0, i1, 0 | 32); - p1 = _mm256_permute2x128_si256(i0, i1, 1 | 48); - r = _mm256_packs_epi32(p0, p1); +static ma_result ma_data_converter_process_pcm_frames__passthrough(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) +{ + ma_uint64 frameCountIn; + ma_uint64 frameCountOut; + ma_uint64 frameCount; - _mm256_stream_si256(((__m256i*)(dst_s16 + i)), r); + MA_ASSERT(pConverter != NULL); + + frameCountIn = 0; + if (pFrameCountIn != NULL) { + frameCountIn = *pFrameCountIn; + } - i += 16; + frameCountOut = 0; + if (pFrameCountOut != NULL) { + frameCountOut = *pFrameCountOut; } + frameCount = ma_min(frameCountIn, frameCountOut); - /* Leftover. */ - for (; i < count; i += 1) { - float x = src_f32[i]; - x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax); - x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */ - x = x * 32767.0f; /* -1..1 to -32767..32767 */ + if (pFramesOut != NULL) { + if (pFramesIn != NULL) { + ma_copy_memory_64(pFramesOut, pFramesIn, frameCount * ma_get_bytes_per_frame(pConverter->config.formatOut, pConverter->config.channelsOut)); + } else { + ma_zero_memory_64(pFramesOut, frameCount * ma_get_bytes_per_frame(pConverter->config.formatOut, pConverter->config.channelsOut)); + } + } - dst_s16[i] = (ma_int16)x; + if (pFrameCountIn != NULL) { + *pFrameCountIn = frameCount; } + if (pFrameCountOut != NULL) { + *pFrameCountOut = frameCount; + } + + return MA_SUCCESS; } -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_f32_to_s16__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - /* TODO: Convert this from AVX to AVX-512. */ - ma_pcm_f32_to_s16__avx2(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_f32_to_s16__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +static ma_result ma_data_converter_process_pcm_frames__format_only(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) { - ma_uint64 i; - ma_uint64 i8; - ma_uint64 count8; - ma_int16* dst_s16; - const float* src_f32; - float ditherMin; - float ditherMax; + ma_uint64 frameCountIn; + ma_uint64 frameCountOut; + ma_uint64 frameCount; - /* Both the input and output buffers need to be aligned to 16 bytes. */ - if ((((ma_uintptr)dst & 15) != 0) || (((ma_uintptr)src & 15) != 0)) { - ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode); - return; + MA_ASSERT(pConverter != NULL); + + frameCountIn = 0; + if (pFrameCountIn != NULL) { + frameCountIn = *pFrameCountIn; } - dst_s16 = (ma_int16*)dst; - src_f32 = (const float*)src; + frameCountOut = 0; + if (pFrameCountOut != NULL) { + frameCountOut = *pFrameCountOut; + } - ditherMin = 0; - ditherMax = 0; - if (ditherMode != ma_dither_mode_none) { - ditherMin = 1.0f / -32768; - ditherMax = 1.0f / 32767; + frameCount = ma_min(frameCountIn, frameCountOut); + + if (pFramesOut != NULL) { + if (pFramesIn != NULL) { + ma_convert_pcm_frames_format(pFramesOut, pConverter->config.formatOut, pFramesIn, pConverter->config.formatIn, frameCount, pConverter->config.channelsIn, pConverter->config.ditherMode); + } else { + ma_zero_memory_64(pFramesOut, frameCount * ma_get_bytes_per_frame(pConverter->config.formatOut, pConverter->config.channelsOut)); + } } - i = 0; + if (pFrameCountIn != NULL) { + *pFrameCountIn = frameCount; + } + if (pFrameCountOut != NULL) { + *pFrameCountOut = frameCount; + } - /* NEON. NEON allows us to output 8 s16's at a time which means our loop is unrolled 8 times. */ - count8 = count >> 3; - for (i8 = 0; i8 < count8; i8 += 1) { - float32x4_t d0; - float32x4_t d1; - float32x4_t x0; - float32x4_t x1; - int32x4_t i0; - int32x4_t i1; + return MA_SUCCESS; +} - if (ditherMode == ma_dither_mode_none) { - d0 = vmovq_n_f32(0); - d1 = vmovq_n_f32(0); - } else if (ditherMode == ma_dither_mode_rectangle) { - float d0v[4]; - d0v[0] = ma_dither_f32_rectangle(ditherMin, ditherMax); - d0v[1] = ma_dither_f32_rectangle(ditherMin, ditherMax); - d0v[2] = ma_dither_f32_rectangle(ditherMin, ditherMax); - d0v[3] = ma_dither_f32_rectangle(ditherMin, ditherMax); - d0 = vld1q_f32(d0v); - float d1v[4]; - d1v[0] = ma_dither_f32_rectangle(ditherMin, ditherMax); - d1v[1] = ma_dither_f32_rectangle(ditherMin, ditherMax); - d1v[2] = ma_dither_f32_rectangle(ditherMin, ditherMax); - d1v[3] = ma_dither_f32_rectangle(ditherMin, ditherMax); - d1 = vld1q_f32(d1v); +static ma_result ma_data_converter_process_pcm_frames__resample_with_format_conversion(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) +{ + ma_result result = MA_SUCCESS; + ma_uint64 frameCountIn; + ma_uint64 frameCountOut; + ma_uint64 framesProcessedIn; + ma_uint64 framesProcessedOut; + + MA_ASSERT(pConverter != NULL); + + frameCountIn = 0; + if (pFrameCountIn != NULL) { + frameCountIn = *pFrameCountIn; + } + + frameCountOut = 0; + if (pFrameCountOut != NULL) { + frameCountOut = *pFrameCountOut; + } + + framesProcessedIn = 0; + framesProcessedOut = 0; + + while (framesProcessedOut < frameCountOut) { + ma_uint8 pTempBufferOut[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + const ma_uint32 tempBufferOutCap = sizeof(pTempBufferOut) / ma_get_bytes_per_frame(pConverter->resampler.config.format, pConverter->resampler.config.channels); + const void* pFramesInThisIteration; + /* */ void* pFramesOutThisIteration; + ma_uint64 frameCountInThisIteration; + ma_uint64 frameCountOutThisIteration; + + if (pFramesIn != NULL) { + pFramesInThisIteration = ma_offset_ptr(pFramesIn, framesProcessedIn * ma_get_bytes_per_frame(pConverter->config.formatIn, pConverter->config.channelsIn)); } else { - float d0v[4]; - d0v[0] = ma_dither_f32_triangle(ditherMin, ditherMax); - d0v[1] = ma_dither_f32_triangle(ditherMin, ditherMax); - d0v[2] = ma_dither_f32_triangle(ditherMin, ditherMax); - d0v[3] = ma_dither_f32_triangle(ditherMin, ditherMax); - d0 = vld1q_f32(d0v); + pFramesInThisIteration = NULL; + } - float d1v[4]; - d1v[0] = ma_dither_f32_triangle(ditherMin, ditherMax); - d1v[1] = ma_dither_f32_triangle(ditherMin, ditherMax); - d1v[2] = ma_dither_f32_triangle(ditherMin, ditherMax); - d1v[3] = ma_dither_f32_triangle(ditherMin, ditherMax); - d1 = vld1q_f32(d1v); + if (pFramesOut != NULL) { + pFramesOutThisIteration = ma_offset_ptr(pFramesOut, framesProcessedOut * ma_get_bytes_per_frame(pConverter->config.formatOut, pConverter->config.channelsOut)); + } else { + pFramesOutThisIteration = NULL; } - x0 = *((float32x4_t*)(src_f32 + i) + 0); - x1 = *((float32x4_t*)(src_f32 + i) + 1); + /* Do a pre format conversion if necessary. */ + if (pConverter->hasPreFormatConversion) { + ma_uint8 pTempBufferIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + const ma_uint32 tempBufferInCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->resampler.config.format, pConverter->resampler.config.channels); - x0 = vaddq_f32(x0, d0); - x1 = vaddq_f32(x1, d1); + frameCountInThisIteration = (frameCountIn - framesProcessedIn); + if (frameCountInThisIteration > tempBufferInCap) { + frameCountInThisIteration = tempBufferInCap; + } - x0 = vmulq_n_f32(x0, 32767.0f); - x1 = vmulq_n_f32(x1, 32767.0f); + if (pConverter->hasPostFormatConversion) { + if (frameCountInThisIteration > tempBufferOutCap) { + frameCountInThisIteration = tempBufferOutCap; + } + } - i0 = vcvtq_s32_f32(x0); - i1 = vcvtq_s32_f32(x1); - *((int16x8_t*)(dst_s16 + i)) = vcombine_s16(vqmovn_s32(i0), vqmovn_s32(i1)); + if (pFramesInThisIteration != NULL) { + ma_convert_pcm_frames_format(pTempBufferIn, pConverter->resampler.config.format, pFramesInThisIteration, pConverter->config.formatIn, frameCountInThisIteration, pConverter->config.channelsIn, pConverter->config.ditherMode); + } else { + MA_ZERO_MEMORY(pTempBufferIn, sizeof(pTempBufferIn)); + } - i += 8; - } + frameCountOutThisIteration = (frameCountOut - framesProcessedOut); + if (pConverter->hasPostFormatConversion) { + /* Both input and output conversion required. Output to the temp buffer. */ + if (frameCountOutThisIteration > tempBufferOutCap) { + frameCountOutThisIteration = tempBufferOutCap; + } - /* Leftover. */ - for (; i < count; i += 1) { - float x = src_f32[i]; - x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax); - x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */ - x = x * 32767.0f; /* -1..1 to -32767..32767 */ + result = ma_resampler_process_pcm_frames(&pConverter->resampler, pTempBufferIn, &frameCountInThisIteration, pTempBufferOut, &frameCountOutThisIteration); + } else { + /* Only pre-format required. Output straight to the output buffer. */ + result = ma_resampler_process_pcm_frames(&pConverter->resampler, pTempBufferIn, &frameCountInThisIteration, pFramesOutThisIteration, &frameCountOutThisIteration); + } - dst_s16[i] = (ma_int16)x; - } -} -#endif + if (result != MA_SUCCESS) { + break; + } + } else { + /* No pre-format required. Just read straight from the input buffer. */ + MA_ASSERT(pConverter->hasPostFormatConversion == MA_TRUE); -void ma_pcm_f32_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_f32_to_s16__reference(dst, src, count, ditherMode); -#else - ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode); -#endif -} + frameCountInThisIteration = (frameCountIn - framesProcessedIn); + frameCountOutThisIteration = (frameCountOut - framesProcessedOut); + if (frameCountOutThisIteration > tempBufferOutCap) { + frameCountOutThisIteration = tempBufferOutCap; + } + result = ma_resampler_process_pcm_frames(&pConverter->resampler, pFramesInThisIteration, &frameCountInThisIteration, pTempBufferOut, &frameCountOutThisIteration); + if (result != MA_SUCCESS) { + break; + } + } -void ma_pcm_f32_to_s24__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_uint8* dst_s24 = (ma_uint8*)dst; - const float* src_f32 = (const float*)src; + /* If we are doing a post format conversion we need to do that now. */ + if (pConverter->hasPostFormatConversion) { + if (pFramesOutThisIteration != NULL) { + ma_convert_pcm_frames_format(pFramesOutThisIteration, pConverter->config.formatOut, pTempBufferOut, pConverter->resampler.config.format, frameCountOutThisIteration, pConverter->resampler.config.channels, pConverter->config.ditherMode); + } + } - ma_uint64 i; - for (i = 0; i < count; i += 1) { - ma_int32 r; - float x = src_f32[i]; - x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */ + framesProcessedIn += frameCountInThisIteration; + framesProcessedOut += frameCountOutThisIteration; -#if 0 - /* The accurate way. */ - x = x + 1; /* -1..1 to 0..2 */ - x = x * 8388607.5f; /* 0..2 to 0..16777215 */ - x = x - 8388608.0f; /* 0..16777215 to -8388608..8388607 */ -#else - /* The fast way. */ - x = x * 8388607.0f; /* -1..1 to -8388607..8388607 */ -#endif + MA_ASSERT(framesProcessedIn <= frameCountIn); + MA_ASSERT(framesProcessedOut <= frameCountOut); - r = (ma_int32)x; - dst_s24[(i*3)+0] = (ma_uint8)((r & 0x0000FF) >> 0); - dst_s24[(i*3)+1] = (ma_uint8)((r & 0x00FF00) >> 8); - dst_s24[(i*3)+2] = (ma_uint8)((r & 0xFF0000) >> 16); + if (frameCountOutThisIteration == 0) { + break; /* Consumed all of our input data. */ + } } - (void)ditherMode; /* No dithering for f32 -> s24. */ -} + if (pFrameCountIn != NULL) { + *pFrameCountIn = framesProcessedIn; + } + if (pFrameCountOut != NULL) { + *pFrameCountOut = framesProcessedOut; + } -void ma_pcm_f32_to_s24__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_f32_to_s24__reference(dst, src, count, ditherMode); + return result; } -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_f32_to_s24__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_f32_to_s24__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_f32_to_s24__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_f32_to_s24__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_f32_to_s24__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +static ma_result ma_data_converter_process_pcm_frames__resample_only(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) { - ma_pcm_f32_to_s24__avx2(dst, src, count, ditherMode); + MA_ASSERT(pConverter != NULL); + + if (pConverter->hasPreFormatConversion == MA_FALSE && pConverter->hasPostFormatConversion == MA_FALSE) { + /* Neither pre- nor post-format required. This is simple case where only resampling is required. */ + return ma_resampler_process_pcm_frames(&pConverter->resampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); + } else { + /* Format conversion required. */ + return ma_data_converter_process_pcm_frames__resample_with_format_conversion(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); + } } -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_f32_to_s24__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) + +static ma_result ma_data_converter_process_pcm_frames__channels_only(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) { - ma_pcm_f32_to_s24__optimized(dst, src, count, ditherMode); -} -#endif + ma_result result; + ma_uint64 frameCountIn; + ma_uint64 frameCountOut; + ma_uint64 frameCount; + + MA_ASSERT(pConverter != NULL); + + frameCountIn = 0; + if (pFrameCountIn != NULL) { + frameCountIn = *pFrameCountIn; + } + + frameCountOut = 0; + if (pFrameCountOut != NULL) { + frameCountOut = *pFrameCountOut; + } + + frameCount = ma_min(frameCountIn, frameCountOut); + + if (pConverter->hasPreFormatConversion == MA_FALSE && pConverter->hasPostFormatConversion == MA_FALSE) { + /* No format conversion required. */ + result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pFramesOut, pFramesIn, frameCount); + if (result != MA_SUCCESS) { + return result; + } + } else { + /* Format conversion required. */ + ma_uint64 framesProcessed = 0; + + while (framesProcessed < frameCount) { + ma_uint8 pTempBufferOut[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + const ma_uint32 tempBufferOutCap = sizeof(pTempBufferOut) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsOut); + const void* pFramesInThisIteration; + /* */ void* pFramesOutThisIteration; + ma_uint64 frameCountThisIteration; + + if (pFramesIn != NULL) { + pFramesInThisIteration = ma_offset_ptr(pFramesIn, framesProcessed * ma_get_bytes_per_frame(pConverter->config.formatIn, pConverter->config.channelsIn)); + } else { + pFramesInThisIteration = NULL; + } + + if (pFramesOut != NULL) { + pFramesOutThisIteration = ma_offset_ptr(pFramesOut, framesProcessed * ma_get_bytes_per_frame(pConverter->config.formatOut, pConverter->config.channelsOut)); + } else { + pFramesOutThisIteration = NULL; + } + + /* Do a pre format conversion if necessary. */ + if (pConverter->hasPreFormatConversion) { + ma_uint8 pTempBufferIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; + const ma_uint32 tempBufferInCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsIn); + + frameCountThisIteration = (frameCount - framesProcessed); + if (frameCountThisIteration > tempBufferInCap) { + frameCountThisIteration = tempBufferInCap; + } + + if (pConverter->hasPostFormatConversion) { + if (frameCountThisIteration > tempBufferOutCap) { + frameCountThisIteration = tempBufferOutCap; + } + } + + if (pFramesInThisIteration != NULL) { + ma_convert_pcm_frames_format(pTempBufferIn, pConverter->channelConverter.format, pFramesInThisIteration, pConverter->config.formatIn, frameCountThisIteration, pConverter->config.channelsIn, pConverter->config.ditherMode); + } else { + MA_ZERO_MEMORY(pTempBufferIn, sizeof(pTempBufferIn)); + } + + if (pConverter->hasPostFormatConversion) { + /* Both input and output conversion required. Output to the temp buffer. */ + result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pTempBufferOut, pTempBufferIn, frameCountThisIteration); + } else { + /* Only pre-format required. Output straight to the output buffer. */ + result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pFramesOutThisIteration, pTempBufferIn, frameCountThisIteration); + } -void ma_pcm_f32_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_f32_to_s24__reference(dst, src, count, ditherMode); -#else - ma_pcm_f32_to_s24__optimized(dst, src, count, ditherMode); -#endif -} + if (result != MA_SUCCESS) { + break; + } + } else { + /* No pre-format required. Just read straight from the input buffer. */ + MA_ASSERT(pConverter->hasPostFormatConversion == MA_TRUE); + frameCountThisIteration = (frameCount - framesProcessed); + if (frameCountThisIteration > tempBufferOutCap) { + frameCountThisIteration = tempBufferOutCap; + } -void ma_pcm_f32_to_s32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_int32* dst_s32 = (ma_int32*)dst; - const float* src_f32 = (const float*)src; + result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pTempBufferOut, pFramesInThisIteration, frameCountThisIteration); + if (result != MA_SUCCESS) { + break; + } + } - ma_uint32 i; - for (i = 0; i < count; i += 1) { - double x = src_f32[i]; - x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */ + /* If we are doing a post format conversion we need to do that now. */ + if (pConverter->hasPostFormatConversion) { + if (pFramesOutThisIteration != NULL) { + ma_convert_pcm_frames_format(pFramesOutThisIteration, pConverter->config.formatOut, pTempBufferOut, pConverter->channelConverter.format, frameCountThisIteration, pConverter->channelConverter.channelsOut, pConverter->config.ditherMode); + } + } -#if 0 - /* The accurate way. */ - x = x + 1; /* -1..1 to 0..2 */ - x = x * 2147483647.5; /* 0..2 to 0..4294967295 */ - x = x - 2147483648.0; /* 0...4294967295 to -2147483648..2147483647 */ -#else - /* The fast way. */ - x = x * 2147483647.0; /* -1..1 to -2147483647..2147483647 */ -#endif + framesProcessed += frameCountThisIteration; + } + } - dst_s32[i] = (ma_int32)x; + if (pFrameCountIn != NULL) { + *pFrameCountIn = frameCount; + } + if (pFrameCountOut != NULL) { + *pFrameCountOut = frameCount; } - (void)ditherMode; /* No dithering for f32 -> s32. */ + return MA_SUCCESS; } -void ma_pcm_f32_to_s32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +static ma_result ma_data_converter_process_pcm_frames__resampling_first(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) { - ma_pcm_f32_to_s32__reference(dst, src, count, ditherMode); -} + ma_result result; + ma_uint64 frameCountIn; + ma_uint64 frameCountOut; + ma_uint64 framesProcessedIn; + ma_uint64 framesProcessedOut; + ma_uint8 pTempBufferIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In resampler format. */ + ma_uint64 tempBufferInCap; + ma_uint8 pTempBufferMid[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In resampler format, channel converter input format. */ + ma_uint64 tempBufferMidCap; + ma_uint8 pTempBufferOut[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In channel converter output format. */ + ma_uint64 tempBufferOutCap; + + MA_ASSERT(pConverter != NULL); + MA_ASSERT(pConverter->resampler.config.format == pConverter->channelConverter.format); + MA_ASSERT(pConverter->resampler.config.channels == pConverter->channelConverter.channelsIn); + MA_ASSERT(pConverter->resampler.config.channels < pConverter->channelConverter.channelsOut); + + frameCountIn = 0; + if (pFrameCountIn != NULL) { + frameCountIn = *pFrameCountIn; + } + + frameCountOut = 0; + if (pFrameCountOut != NULL) { + frameCountOut = *pFrameCountOut; + } -#if defined(MA_SUPPORT_SSE2) -void ma_pcm_f32_to_s32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_pcm_f32_to_s32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_pcm_f32_to_s32__avx512(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_f32_to_s32__avx2(dst, src, count, ditherMode); -} -#endif -#if defined(MA_SUPPORT_NEON) -void ma_pcm_f32_to_s32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode); -} -#endif + framesProcessedIn = 0; + framesProcessedOut = 0; -void ma_pcm_f32_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_f32_to_s32__reference(dst, src, count, ditherMode); -#else - ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode); -#endif -} + tempBufferInCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->resampler.config.format, pConverter->resampler.config.channels); + tempBufferMidCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->resampler.config.format, pConverter->resampler.config.channels); + tempBufferOutCap = sizeof(pTempBufferOut) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsOut); + while (framesProcessedOut < frameCountOut) { + ma_uint64 frameCountInThisIteration; + ma_uint64 frameCountOutThisIteration; + const void* pRunningFramesIn = NULL; + void* pRunningFramesOut = NULL; + const void* pResampleBufferIn; + void* pChannelsBufferOut; -void ma_pcm_f32_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) -{ - (void)ditherMode; + if (pFramesIn != NULL) { + pRunningFramesIn = ma_offset_ptr(pFramesIn, framesProcessedIn * ma_get_bytes_per_frame(pConverter->config.formatIn, pConverter->config.channelsIn)); + } + if (pFramesOut != NULL) { + pRunningFramesOut = ma_offset_ptr(pFramesOut, framesProcessedOut * ma_get_bytes_per_frame(pConverter->config.formatOut, pConverter->config.channelsOut)); + } - ma_copy_memory_64(dst, src, count * sizeof(float)); -} + /* Run input data through the resampler and output it to the temporary buffer. */ + frameCountInThisIteration = (frameCountIn - framesProcessedIn); + if (pConverter->hasPreFormatConversion) { + if (frameCountInThisIteration > tempBufferInCap) { + frameCountInThisIteration = tempBufferInCap; + } + } -void ma_pcm_interleave_f32__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) -{ - float* dst_f32 = (float*)dst; - const float** src_f32 = (const float**)src; + frameCountOutThisIteration = (frameCountOut - framesProcessedOut); + if (frameCountOutThisIteration > tempBufferMidCap) { + frameCountOutThisIteration = tempBufferMidCap; + } - ma_uint64 iFrame; - for (iFrame = 0; iFrame < frameCount; iFrame += 1) { - ma_uint32 iChannel; - for (iChannel = 0; iChannel < channels; iChannel += 1) { - dst_f32[iFrame*channels + iChannel] = src_f32[iChannel][iFrame]; + /* We can't read more frames than can fit in the output buffer. */ + if (pConverter->hasPostFormatConversion) { + if (frameCountOutThisIteration > tempBufferOutCap) { + frameCountOutThisIteration = tempBufferOutCap; + } } - } -} -void ma_pcm_interleave_f32__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) -{ - ma_pcm_interleave_f32__reference(dst, src, frameCount, channels); -} + /* We need to ensure we don't try to process too many input frames that we run out of room in the output buffer. If this happens we'll end up glitching. */ + { + ma_uint64 requiredInputFrameCount = ma_resampler_get_required_input_frame_count(&pConverter->resampler, frameCountOutThisIteration); + if (frameCountInThisIteration > requiredInputFrameCount) { + frameCountInThisIteration = requiredInputFrameCount; + } + } -void ma_pcm_interleave_f32(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) -{ -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_interleave_f32__reference(dst, src, frameCount, channels); -#else - ma_pcm_interleave_f32__optimized(dst, src, frameCount, channels); -#endif -} + if (pConverter->hasPreFormatConversion) { + if (pFramesIn != NULL) { + ma_convert_pcm_frames_format(pTempBufferIn, pConverter->resampler.config.format, pRunningFramesIn, pConverter->config.formatIn, frameCountInThisIteration, pConverter->config.channelsIn, pConverter->config.ditherMode); + pResampleBufferIn = pTempBufferIn; + } else { + pResampleBufferIn = NULL; + } + } else { + pResampleBufferIn = pRunningFramesIn; + } + result = ma_resampler_process_pcm_frames(&pConverter->resampler, pResampleBufferIn, &frameCountInThisIteration, pTempBufferMid, &frameCountOutThisIteration); + if (result != MA_SUCCESS) { + return result; + } -void ma_pcm_deinterleave_f32__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) -{ - float** dst_f32 = (float**)dst; - const float* src_f32 = (const float*)src; - ma_uint64 iFrame; - for (iFrame = 0; iFrame < frameCount; iFrame += 1) { - ma_uint32 iChannel; - for (iChannel = 0; iChannel < channels; iChannel += 1) { - dst_f32[iChannel][iFrame] = src_f32[iFrame*channels + iChannel]; + /* + The input data has been resampled so now we need to run it through the channel converter. The input data is always contained in pTempBufferMid. We only need to do + this part if we have an output buffer. + */ + if (pFramesOut != NULL) { + if (pConverter->hasPostFormatConversion) { + pChannelsBufferOut = pTempBufferOut; + } else { + pChannelsBufferOut = pRunningFramesOut; + } + + result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pChannelsBufferOut, pTempBufferMid, frameCountOutThisIteration); + if (result != MA_SUCCESS) { + return result; + } + + /* Finally we do post format conversion. */ + if (pConverter->hasPostFormatConversion) { + ma_convert_pcm_frames_format(pRunningFramesOut, pConverter->config.formatOut, pChannelsBufferOut, pConverter->channelConverter.format, frameCountOutThisIteration, pConverter->channelConverter.channelsOut, pConverter->config.ditherMode); + } + } + + + framesProcessedIn += frameCountInThisIteration; + framesProcessedOut += frameCountOutThisIteration; + + MA_ASSERT(framesProcessedIn <= frameCountIn); + MA_ASSERT(framesProcessedOut <= frameCountOut); + + if (frameCountOutThisIteration == 0) { + break; /* Consumed all of our input data. */ } } -} -void ma_pcm_deinterleave_f32__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) -{ - ma_pcm_deinterleave_f32__reference(dst, src, frameCount, channels); + if (pFrameCountIn != NULL) { + *pFrameCountIn = framesProcessedIn; + } + if (pFrameCountOut != NULL) { + *pFrameCountOut = framesProcessedOut; + } + + return MA_SUCCESS; } -void ma_pcm_deinterleave_f32(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) +static ma_result ma_data_converter_process_pcm_frames__channels_first(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) { -#ifdef MA_USE_REFERENCE_CONVERSION_APIS - ma_pcm_deinterleave_f32__reference(dst, src, frameCount, channels); -#else - ma_pcm_deinterleave_f32__optimized(dst, src, frameCount, channels); -#endif -} + ma_result result; + ma_uint64 frameCountIn; + ma_uint64 frameCountOut; + ma_uint64 framesProcessedIn; + ma_uint64 framesProcessedOut; + ma_uint8 pTempBufferIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In resampler format. */ + ma_uint64 tempBufferInCap; + ma_uint8 pTempBufferMid[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In resampler format, channel converter input format. */ + ma_uint64 tempBufferMidCap; + ma_uint8 pTempBufferOut[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In channel converter output format. */ + ma_uint64 tempBufferOutCap; + + MA_ASSERT(pConverter != NULL); + MA_ASSERT(pConverter->resampler.config.format == pConverter->channelConverter.format); + MA_ASSERT(pConverter->resampler.config.channels == pConverter->channelConverter.channelsOut); + MA_ASSERT(pConverter->resampler.config.channels < pConverter->channelConverter.channelsIn); + + frameCountIn = 0; + if (pFrameCountIn != NULL) { + frameCountIn = *pFrameCountIn; + } + + frameCountOut = 0; + if (pFrameCountOut != NULL) { + frameCountOut = *pFrameCountOut; + } + framesProcessedIn = 0; + framesProcessedOut = 0; -void ma_format_converter_init_callbacks__default(ma_format_converter* pConverter) -{ - ma_assert(pConverter != NULL); + tempBufferInCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsIn); + tempBufferMidCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsOut); + tempBufferOutCap = sizeof(pTempBufferOut) / ma_get_bytes_per_frame(pConverter->resampler.config.format, pConverter->resampler.config.channels); - switch (pConverter->config.formatIn) - { - case ma_format_u8: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_u8_to_u8; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_u8_to_s16; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_u8_to_s24; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_u8_to_s32; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_u8_to_f32; + while (framesProcessedOut < frameCountOut) { + ma_uint64 frameCountInThisIteration; + ma_uint64 frameCountOutThisIteration; + const void* pRunningFramesIn = NULL; + void* pRunningFramesOut = NULL; + const void* pChannelsBufferIn; + void* pResampleBufferOut; + + if (pFramesIn != NULL) { + pRunningFramesIn = ma_offset_ptr(pFramesIn, framesProcessedIn * ma_get_bytes_per_frame(pConverter->config.formatIn, pConverter->config.channelsIn)); + } + if (pFramesOut != NULL) { + pRunningFramesOut = ma_offset_ptr(pFramesOut, framesProcessedOut * ma_get_bytes_per_frame(pConverter->config.formatOut, pConverter->config.channelsOut)); + } + + /* Run input data through the channel converter and output it to the temporary buffer. */ + frameCountInThisIteration = (frameCountIn - framesProcessedIn); + + if (pConverter->hasPreFormatConversion) { + if (frameCountInThisIteration > tempBufferInCap) { + frameCountInThisIteration = tempBufferInCap; } - } break; - case ma_format_s16: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_s16_to_u8; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_s16_to_s16; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_s16_to_s24; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_s16_to_s32; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_s16_to_f32; + if (pRunningFramesIn != NULL) { + ma_convert_pcm_frames_format(pTempBufferIn, pConverter->channelConverter.format, pRunningFramesIn, pConverter->config.formatIn, frameCountInThisIteration, pConverter->config.channelsIn, pConverter->config.ditherMode); + pChannelsBufferIn = pTempBufferIn; + } else { + pChannelsBufferIn = NULL; } - } break; + } else { + pChannelsBufferIn = pRunningFramesIn; + } - case ma_format_s24: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_s24_to_u8; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_s24_to_s16; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_s24_to_s24; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_s24_to_s32; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_s24_to_f32; + /* + We can't convert more frames than will fit in the output buffer. We shouldn't actually need to do this check because the channel count is always reduced + in this case which means we should always have capacity, but I'm leaving it here just for safety for future maintenance. + */ + if (frameCountInThisIteration > tempBufferMidCap) { + frameCountInThisIteration = tempBufferMidCap; + } + + /* + Make sure we don't read any more input frames than we need to fill the output frame count. If we do this we will end up in a situation where we lose some + input samples and will end up glitching. + */ + frameCountOutThisIteration = (frameCountOut - framesProcessedOut); + if (frameCountOutThisIteration > tempBufferMidCap) { + frameCountOutThisIteration = tempBufferMidCap; + } + + if (pConverter->hasPostFormatConversion) { + ma_uint64 requiredInputFrameCount; + + if (frameCountOutThisIteration > tempBufferOutCap) { + frameCountOutThisIteration = tempBufferOutCap; } - } break; - case ma_format_s32: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_s32_to_u8; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_s32_to_s16; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_s32_to_s24; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_s32_to_s32; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_s32_to_f32; + requiredInputFrameCount = ma_resampler_get_required_input_frame_count(&pConverter->resampler, frameCountOutThisIteration); + if (frameCountInThisIteration > requiredInputFrameCount) { + frameCountInThisIteration = requiredInputFrameCount; } - } break; + } - case ma_format_f32: - default: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_f32_to_u8; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_f32_to_s16; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_f32_to_s24; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_f32_to_s32; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_f32_to_f32; + result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pTempBufferMid, pChannelsBufferIn, frameCountInThisIteration); + if (result != MA_SUCCESS) { + return result; + } + + + /* At this point we have converted the channels to the output channel count which we now need to resample. */ + if (pConverter->hasPostFormatConversion) { + pResampleBufferOut = pTempBufferOut; + } else { + pResampleBufferOut = pRunningFramesOut; + } + + result = ma_resampler_process_pcm_frames(&pConverter->resampler, pTempBufferMid, &frameCountInThisIteration, pResampleBufferOut, &frameCountOutThisIteration); + if (result != MA_SUCCESS) { + return result; + } + + /* Finally we can do the post format conversion. */ + if (pConverter->hasPostFormatConversion) { + if (pRunningFramesOut != NULL) { + ma_convert_pcm_frames_format(pRunningFramesOut, pConverter->config.formatOut, pResampleBufferOut, pConverter->resampler.config.format, frameCountOutThisIteration, pConverter->config.channelsOut, pConverter->config.ditherMode); } - } break; + } + + framesProcessedIn += frameCountInThisIteration; + framesProcessedOut += frameCountOutThisIteration; + + MA_ASSERT(framesProcessedIn <= frameCountIn); + MA_ASSERT(framesProcessedOut <= frameCountOut); + + if (frameCountOutThisIteration == 0) { + break; /* Consumed all of our input data. */ + } + } + + if (pFrameCountIn != NULL) { + *pFrameCountIn = framesProcessedIn; } + if (pFrameCountOut != NULL) { + *pFrameCountOut = framesProcessedOut; + } + + return MA_SUCCESS; } -#if defined(MA_SUPPORT_SSE2) -void ma_format_converter_init_callbacks__sse2(ma_format_converter* pConverter) +ma_result ma_data_converter_process_pcm_frames(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut) { - ma_assert(pConverter != NULL); - - switch (pConverter->config.formatIn) - { - case ma_format_u8: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_u8_to_u8; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_u8_to_s16__sse2; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_u8_to_s24__sse2; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_u8_to_s32__sse2; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_u8_to_f32__sse2; - } - } break; + if (pConverter == NULL) { + return MA_INVALID_ARGS; + } - case ma_format_s16: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_s16_to_u8__sse2; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_s16_to_s16; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_s16_to_s24__sse2; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_s16_to_s32__sse2; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_s16_to_f32__sse2; - } - } break; + if (pConverter->isPassthrough) { + return ma_data_converter_process_pcm_frames__passthrough(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); + } - case ma_format_s24: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_s24_to_u8__sse2; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_s24_to_s16__sse2; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_s24_to_s24; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_s24_to_s32__sse2; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_s24_to_f32__sse2; - } - } break; + /* + Here is where the real work is done. Getting here means we're not using a passthrough and we need to move the data through each of the relevant stages. The order + of our stages depends on the input and output channel count. If the input channels is less than the output channels we want to do sample rate conversion first so + that it has less work (resampling is the most expensive part of format conversion). + */ + if (pConverter->config.channelsIn < pConverter->config.channelsOut) { + /* Do resampling first, if necessary. */ + MA_ASSERT(pConverter->hasChannelConverter == MA_TRUE); - case ma_format_s32: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_s32_to_u8__sse2; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_s32_to_s16__sse2; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_s32_to_s24__sse2; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_s32_to_s32; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_s32_to_f32__sse2; + if (pConverter->hasResampler) { + /* Resampling first. */ + return ma_data_converter_process_pcm_frames__resampling_first(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); + } else { + /* Resampling not required. */ + return ma_data_converter_process_pcm_frames__channels_only(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); + } + } else { + /* Do channel conversion first, if necessary. */ + if (pConverter->hasChannelConverter) { + if (pConverter->hasResampler) { + /* Channel routing first. */ + return ma_data_converter_process_pcm_frames__channels_first(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); + } else { + /* Resampling not required. */ + return ma_data_converter_process_pcm_frames__channels_only(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); } - } break; - - case ma_format_f32: - default: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_f32_to_u8__sse2; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_f32_to_s16__sse2; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_f32_to_s24__sse2; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_f32_to_s32__sse2; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_f32_to_f32; + } else { + /* Channel routing not required. */ + if (pConverter->hasResampler) { + /* Resampling only. */ + return ma_data_converter_process_pcm_frames__resample_only(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); + } else { + /* No channel routing nor resampling required. Just format conversion. */ + return ma_data_converter_process_pcm_frames__format_only(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut); } - } break; + } } } -#endif -#if defined(MA_SUPPORT_AVX2) -void ma_format_converter_init_callbacks__avx2(ma_format_converter* pConverter) +ma_result ma_data_converter_set_rate(ma_data_converter* pConverter, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut) { - ma_assert(pConverter != NULL); - - switch (pConverter->config.formatIn) - { - case ma_format_u8: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_u8_to_u8; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_u8_to_s16__avx2; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_u8_to_s24__avx2; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_u8_to_s32__avx2; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_u8_to_f32__avx2; - } - } break; + if (pConverter == NULL) { + return MA_INVALID_ARGS; + } - case ma_format_s16: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_s16_to_u8__avx2; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_s16_to_s16; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_s16_to_s24__avx2; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_s16_to_s32__avx2; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_s16_to_f32__avx2; - } - } break; + if (pConverter->hasResampler == MA_FALSE) { + return MA_INVALID_OPERATION; /* Dynamic resampling not enabled. */ + } - case ma_format_s24: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_s24_to_u8__avx2; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_s24_to_s16__avx2; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_s24_to_s24; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_s24_to_s32__avx2; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_s24_to_f32__avx2; - } - } break; + return ma_resampler_set_rate(&pConverter->resampler, sampleRateIn, sampleRateOut); +} - case ma_format_s32: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_s32_to_u8__avx2; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_s32_to_s16__avx2; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_s32_to_s24__avx2; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_s32_to_s32; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_s32_to_f32__avx2; - } - } break; +ma_result ma_data_converter_set_rate_ratio(ma_data_converter* pConverter, float ratioInOut) +{ + if (pConverter == NULL) { + return MA_INVALID_ARGS; + } - case ma_format_f32: - default: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_f32_to_u8__avx2; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_f32_to_s16__avx2; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_f32_to_s24__avx2; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_f32_to_s32__avx2; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_f32_to_f32; - } - } break; + if (pConverter->hasResampler == MA_FALSE) { + return MA_INVALID_OPERATION; /* Dynamic resampling not enabled. */ } + + return ma_resampler_set_rate_ratio(&pConverter->resampler, ratioInOut); } -#endif -#if defined(MA_SUPPORT_AVX512) -void ma_format_converter_init_callbacks__avx512(ma_format_converter* pConverter) +ma_uint64 ma_data_converter_get_required_input_frame_count(ma_data_converter* pConverter, ma_uint64 outputFrameCount) { - ma_assert(pConverter != NULL); + if (pConverter == NULL) { + return 0; + } - switch (pConverter->config.formatIn) - { - case ma_format_u8: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_u8_to_u8; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_u8_to_s16__avx512; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_u8_to_s24__avx512; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_u8_to_s32__avx512; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_u8_to_f32__avx512; - } - } break; + if (pConverter->hasResampler) { + return ma_resampler_get_required_input_frame_count(&pConverter->resampler, outputFrameCount); + } else { + return outputFrameCount; /* 1:1 */ + } +} - case ma_format_s16: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_s16_to_u8__avx512; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_s16_to_s16; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_s16_to_s24__avx512; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_s16_to_s32__avx512; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_s16_to_f32__avx512; - } - } break; +ma_uint64 ma_data_converter_get_expected_output_frame_count(ma_data_converter* pConverter, ma_uint64 inputFrameCount) +{ + if (pConverter == NULL) { + return 0; + } - case ma_format_s24: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_s24_to_u8__avx512; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_s24_to_s16__avx512; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_s24_to_s24; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_s24_to_s32__avx512; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_s24_to_f32__avx512; - } - } break; + if (pConverter->hasResampler) { + return ma_resampler_get_expected_output_frame_count(&pConverter->resampler, inputFrameCount); + } else { + return inputFrameCount; /* 1:1 */ + } +} - case ma_format_s32: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_s32_to_u8__avx512; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_s32_to_s16__avx512; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_s32_to_s24__avx512; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_s32_to_s32; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_s32_to_f32__avx512; - } - } break; +ma_uint64 ma_data_converter_get_input_latency(ma_data_converter* pConverter) +{ + if (pConverter == NULL) { + return 0; + } - case ma_format_f32: - default: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_f32_to_u8__avx512; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_f32_to_s16__avx512; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_f32_to_s24__avx512; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_f32_to_s32__avx512; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_f32_to_f32; - } - } break; + if (pConverter->hasResampler) { + return ma_resampler_get_input_latency(&pConverter->resampler); } + + return 0; /* No latency without a resampler. */ } -#endif -#if defined(MA_SUPPORT_NEON) -void ma_format_converter_init_callbacks__neon(ma_format_converter* pConverter) +ma_uint64 ma_data_converter_get_output_latency(ma_data_converter* pConverter) { - ma_assert(pConverter != NULL); + if (pConverter == NULL) { + return 0; + } - switch (pConverter->config.formatIn) - { - case ma_format_u8: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_u8_to_u8; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_u8_to_s16__neon; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_u8_to_s24__neon; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_u8_to_s32__neon; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_u8_to_f32__neon; - } - } break; + if (pConverter->hasResampler) { + return ma_resampler_get_output_latency(&pConverter->resampler); + } - case ma_format_s16: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_s16_to_u8__neon; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_s16_to_s16; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_s16_to_s24__neon; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_s16_to_s32__neon; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_s16_to_f32__neon; - } - } break; + return 0; /* No latency without a resampler. */ +} - case ma_format_s24: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_s24_to_u8__neon; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_s24_to_s16__neon; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_s24_to_s24; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_s24_to_s32__neon; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_s24_to_f32__neon; - } - } break; - case ma_format_s32: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_s32_to_u8__neon; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_s32_to_s16__neon; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_s32_to_s24__neon; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_s32_to_s32; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_s32_to_f32__neon; - } - } break; - case ma_format_f32: - default: - { - if (pConverter->config.formatOut == ma_format_u8) { - pConverter->onConvertPCM = ma_pcm_f32_to_u8__neon; - } else if (pConverter->config.formatOut == ma_format_s16) { - pConverter->onConvertPCM = ma_pcm_f32_to_s16__neon; - } else if (pConverter->config.formatOut == ma_format_s24) { - pConverter->onConvertPCM = ma_pcm_f32_to_s24__neon; - } else if (pConverter->config.formatOut == ma_format_s32) { - pConverter->onConvertPCM = ma_pcm_f32_to_s32__neon; - } else if (pConverter->config.formatOut == ma_format_f32) { - pConverter->onConvertPCM = ma_pcm_f32_to_f32; - } - } break; - } +/************************************************************************************************************************************************************** + +Format Conversion + +**************************************************************************************************************************************************************/ + +/* u8 */ +void ma_pcm_u8_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + (void)ditherMode; + ma_copy_memory_64(dst, src, count * sizeof(ma_uint8)); } -#endif -ma_result ma_format_converter_init(const ma_format_converter_config* pConfig, ma_format_converter* pConverter) + +static MA_INLINE void ma_pcm_u8_to_s16__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - if (pConverter == NULL) { - return MA_INVALID_ARGS; - } - ma_zero_object(pConverter); + ma_int16* dst_s16 = (ma_int16*)dst; + const ma_uint8* src_u8 = (const ma_uint8*)src; - if (pConfig == NULL) { - return MA_INVALID_ARGS; + ma_uint64 i; + for (i = 0; i < count; i += 1) { + ma_int16 x = src_u8[i]; + x = x - 128; + x = x << 8; + dst_s16[i] = x; } - pConverter->config = *pConfig; + (void)ditherMode; +} - /* SIMD */ - pConverter->useSSE2 = ma_has_sse2() && !pConfig->noSSE2; - pConverter->useAVX2 = ma_has_avx2() && !pConfig->noAVX2; - pConverter->useAVX512 = ma_has_avx512f() && !pConfig->noAVX512; - pConverter->useNEON = ma_has_neon() && !pConfig->noNEON; +static MA_INLINE void ma_pcm_u8_to_s16__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_u8_to_s16__reference(dst, src, count, ditherMode); +} -#if defined(MA_SUPPORT_AVX512) - if (pConverter->useAVX512) { - ma_format_converter_init_callbacks__avx512(pConverter); - } else +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_u8_to_s16__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_u8_to_s16__optimized(dst, src, count, ditherMode); +} #endif #if defined(MA_SUPPORT_AVX2) - if (pConverter->useAVX2) { - ma_format_converter_init_callbacks__avx2(pConverter); - } else -#endif -#if defined(MA_SUPPORT_SSE2) - if (pConverter->useSSE2) { - ma_format_converter_init_callbacks__sse2(pConverter); - } else +static MA_INLINE void ma_pcm_u8_to_s16__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_u8_to_s16__optimized(dst, src, count, ditherMode); +} #endif #if defined(MA_SUPPORT_NEON) - if (pConverter->useNEON) { - ma_format_converter_init_callbacks__neon(pConverter); - } else +static MA_INLINE void ma_pcm_u8_to_s16__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_u8_to_s16__optimized(dst, src, count, ditherMode); +} #endif - { - ma_format_converter_init_callbacks__default(pConverter); - } - switch (pConfig->formatOut) - { - case ma_format_u8: - { - pConverter->onInterleavePCM = ma_pcm_interleave_u8; - pConverter->onDeinterleavePCM = ma_pcm_deinterleave_u8; - } break; - case ma_format_s16: - { - pConverter->onInterleavePCM = ma_pcm_interleave_s16; - pConverter->onDeinterleavePCM = ma_pcm_deinterleave_s16; - } break; - case ma_format_s24: - { - pConverter->onInterleavePCM = ma_pcm_interleave_s24; - pConverter->onDeinterleavePCM = ma_pcm_deinterleave_s24; - } break; - case ma_format_s32: - { - pConverter->onInterleavePCM = ma_pcm_interleave_s32; - pConverter->onDeinterleavePCM = ma_pcm_deinterleave_s32; - } break; - case ma_format_f32: - default: +void ma_pcm_u8_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_u8_to_s16__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_u8_to_s16__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_u8_to_s16__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_u8_to_s16__neon(dst, src, count, ditherMode); + } else + #endif { - pConverter->onInterleavePCM = ma_pcm_interleave_f32; - pConverter->onDeinterleavePCM = ma_pcm_deinterleave_f32; - } break; - } - - return MA_SUCCESS; + ma_pcm_u8_to_s16__optimized(dst, src, count, ditherMode); + } +#endif } -ma_uint64 ma_format_converter_read(ma_format_converter* pConverter, ma_uint64 frameCount, void* pFramesOut, void* pUserData) + +static MA_INLINE void ma_pcm_u8_to_s24__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - ma_uint64 totalFramesRead; - ma_uint32 sampleSizeIn; - ma_uint32 sampleSizeOut; - ma_uint32 frameSizeOut; - ma_uint8* pNextFramesOut; + ma_uint8* dst_s24 = (ma_uint8*)dst; + const ma_uint8* src_u8 = (const ma_uint8*)src; - if (pConverter == NULL || pFramesOut == NULL) { - return 0; + ma_uint64 i; + for (i = 0; i < count; i += 1) { + ma_int16 x = src_u8[i]; + x = x - 128; + + dst_s24[i*3+0] = 0; + dst_s24[i*3+1] = 0; + dst_s24[i*3+2] = (ma_uint8)((ma_int8)x); } - totalFramesRead = 0; - sampleSizeIn = ma_get_bytes_per_sample(pConverter->config.formatIn); - sampleSizeOut = ma_get_bytes_per_sample(pConverter->config.formatOut); - /*frameSizeIn = sampleSizeIn * pConverter->config.channels;*/ - frameSizeOut = sampleSizeOut * pConverter->config.channels; - pNextFramesOut = (ma_uint8*)pFramesOut; - - if (pConverter->config.onRead != NULL) { - /* Input data is interleaved. */ - if (pConverter->config.formatIn == pConverter->config.formatOut) { - /* Pass through. */ - while (totalFramesRead < frameCount) { - ma_uint32 framesJustRead; - ma_uint64 framesRemaining = (frameCount - totalFramesRead); - ma_uint64 framesToReadRightNow = framesRemaining; - if (framesToReadRightNow > 0xFFFFFFFF) { - framesToReadRightNow = 0xFFFFFFFF; - } + (void)ditherMode; +} - framesJustRead = (ma_uint32)pConverter->config.onRead(pConverter, (ma_uint32)framesToReadRightNow, pNextFramesOut, pUserData); - if (framesJustRead == 0) { - break; - } +static MA_INLINE void ma_pcm_u8_to_s24__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_u8_to_s24__reference(dst, src, count, ditherMode); +} - totalFramesRead += framesJustRead; - pNextFramesOut += framesJustRead * frameSizeOut; +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_u8_to_s24__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_u8_to_s24__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE void ma_pcm_u8_to_s24__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_u8_to_s24__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_u8_to_s24__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_u8_to_s24__optimized(dst, src, count, ditherMode); +} +#endif - if (framesJustRead < framesToReadRightNow) { - break; - } - } - } else { - /* Conversion required. */ - ma_uint32 maxFramesToReadAtATime; +void ma_pcm_u8_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_u8_to_s24__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_u8_to_s24__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_u8_to_s24__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_u8_to_s24__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_u8_to_s24__optimized(dst, src, count, ditherMode); + } +#endif +} - MA_ALIGN(MA_SIMD_ALIGNMENT) ma_uint8 temp[MA_MAX_CHANNELS * MA_MAX_PCM_SAMPLE_SIZE_IN_BYTES * 128]; - ma_assert(sizeof(temp) <= 0xFFFFFFFF); - maxFramesToReadAtATime = sizeof(temp) / sampleSizeIn / pConverter->config.channels; +static MA_INLINE void ma_pcm_u8_to_s32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_int32* dst_s32 = (ma_int32*)dst; + const ma_uint8* src_u8 = (const ma_uint8*)src; - while (totalFramesRead < frameCount) { - ma_uint32 framesJustRead; - ma_uint64 framesRemaining = (frameCount - totalFramesRead); - ma_uint64 framesToReadRightNow = framesRemaining; - if (framesToReadRightNow > maxFramesToReadAtATime) { - framesToReadRightNow = maxFramesToReadAtATime; - } + ma_uint64 i; + for (i = 0; i < count; i += 1) { + ma_int32 x = src_u8[i]; + x = x - 128; + x = x << 24; + dst_s32[i] = x; + } - framesJustRead = (ma_uint32)pConverter->config.onRead(pConverter, (ma_uint32)framesToReadRightNow, temp, pUserData); - if (framesJustRead == 0) { - break; - } + (void)ditherMode; +} - pConverter->onConvertPCM(pNextFramesOut, temp, framesJustRead*pConverter->config.channels, pConverter->config.ditherMode); +static MA_INLINE void ma_pcm_u8_to_s32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_u8_to_s32__reference(dst, src, count, ditherMode); +} - totalFramesRead += framesJustRead; - pNextFramesOut += framesJustRead * frameSizeOut; +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_u8_to_s32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_u8_to_s32__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE void ma_pcm_u8_to_s32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_u8_to_s32__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_u8_to_s32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_u8_to_s32__optimized(dst, src, count, ditherMode); +} +#endif - if (framesJustRead < framesToReadRightNow) { - break; - } - } +void ma_pcm_u8_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_u8_to_s32__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_u8_to_s32__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_u8_to_s32__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_u8_to_s32__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_u8_to_s32__optimized(dst, src, count, ditherMode); } - } else { - /* Input data is deinterleaved. If a conversion is required we need to do an intermediary step. */ - void* ppTempSamplesOfOutFormat[MA_MAX_CHANNELS]; - size_t splitBufferSizeOut; - ma_uint32 maxFramesToReadAtATime; - - MA_ALIGN(MA_SIMD_ALIGNMENT) ma_uint8 tempSamplesOfOutFormat[MA_MAX_CHANNELS * MA_MAX_PCM_SAMPLE_SIZE_IN_BYTES * 128]; - ma_assert(sizeof(tempSamplesOfOutFormat) <= 0xFFFFFFFF); +#endif +} - ma_split_buffer(tempSamplesOfOutFormat, sizeof(tempSamplesOfOutFormat), pConverter->config.channels, MA_SIMD_ALIGNMENT, (void**)&ppTempSamplesOfOutFormat, &splitBufferSizeOut); - maxFramesToReadAtATime = (ma_uint32)(splitBufferSizeOut / sampleSizeIn); +static MA_INLINE void ma_pcm_u8_to_f32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + float* dst_f32 = (float*)dst; + const ma_uint8* src_u8 = (const ma_uint8*)src; - while (totalFramesRead < frameCount) { - ma_uint32 framesJustRead; - ma_uint64 framesRemaining = (frameCount - totalFramesRead); - ma_uint64 framesToReadRightNow = framesRemaining; - if (framesToReadRightNow > maxFramesToReadAtATime) { - framesToReadRightNow = maxFramesToReadAtATime; - } + ma_uint64 i; + for (i = 0; i < count; i += 1) { + float x = (float)src_u8[i]; + x = x * 0.00784313725490196078f; /* 0..255 to 0..2 */ + x = x - 1; /* 0..2 to -1..1 */ - if (pConverter->config.formatIn == pConverter->config.formatOut) { - /* Only interleaving. */ - framesJustRead = (ma_uint32)pConverter->config.onReadDeinterleaved(pConverter, (ma_uint32)framesToReadRightNow, ppTempSamplesOfOutFormat, pUserData); - if (framesJustRead == 0) { - break; - } - } else { - /* Interleaving + Conversion. Convert first, then interleave. */ - void* ppTempSamplesOfInFormat[MA_MAX_CHANNELS]; - size_t splitBufferSizeIn; - ma_uint32 iChannel; + dst_f32[i] = x; + } - MA_ALIGN(MA_SIMD_ALIGNMENT) ma_uint8 tempSamplesOfInFormat[MA_MAX_CHANNELS * MA_MAX_PCM_SAMPLE_SIZE_IN_BYTES * 128]; + (void)ditherMode; +} - ma_split_buffer(tempSamplesOfInFormat, sizeof(tempSamplesOfInFormat), pConverter->config.channels, MA_SIMD_ALIGNMENT, (void**)&ppTempSamplesOfInFormat, &splitBufferSizeIn); +static MA_INLINE void ma_pcm_u8_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_u8_to_f32__reference(dst, src, count, ditherMode); +} - if (framesToReadRightNow > (splitBufferSizeIn / sampleSizeIn)) { - framesToReadRightNow = (splitBufferSizeIn / sampleSizeIn); - } +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_u8_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE void ma_pcm_u8_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_u8_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode); +} +#endif - framesJustRead = (ma_uint32)pConverter->config.onReadDeinterleaved(pConverter, (ma_uint32)framesToReadRightNow, ppTempSamplesOfInFormat, pUserData); - if (framesJustRead == 0) { - break; - } +void ma_pcm_u8_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_u8_to_f32__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_u8_to_f32__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_u8_to_f32__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_u8_to_f32__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode); + } +#endif +} - for (iChannel = 0; iChannel < pConverter->config.channels; iChannel += 1) { - pConverter->onConvertPCM(ppTempSamplesOfOutFormat[iChannel], ppTempSamplesOfInFormat[iChannel], framesJustRead, pConverter->config.ditherMode); - } - } - pConverter->onInterleavePCM(pNextFramesOut, (const void**)ppTempSamplesOfOutFormat, framesJustRead, pConverter->config.channels); +#ifdef MA_USE_REFERENCE_CONVERSION_APIS +static MA_INLINE void ma_pcm_interleave_u8__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) +{ + ma_uint8* dst_u8 = (ma_uint8*)dst; + const ma_uint8** src_u8 = (const ma_uint8**)src; - totalFramesRead += framesJustRead; - pNextFramesOut += framesJustRead * frameSizeOut; + ma_uint64 iFrame; + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + ma_uint32 iChannel; + for (iChannel = 0; iChannel < channels; iChannel += 1) { + dst_u8[iFrame*channels + iChannel] = src_u8[iChannel][iFrame]; + } + } +} +#else +static MA_INLINE void ma_pcm_interleave_u8__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) +{ + ma_uint8* dst_u8 = (ma_uint8*)dst; + const ma_uint8** src_u8 = (const ma_uint8**)src; - if (framesJustRead < framesToReadRightNow) { - break; + if (channels == 1) { + ma_copy_memory_64(dst, src[0], frameCount * sizeof(ma_uint8)); + } else if (channels == 2) { + ma_uint64 iFrame; + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + dst_u8[iFrame*2 + 0] = src_u8[0][iFrame]; + dst_u8[iFrame*2 + 1] = src_u8[1][iFrame]; + } + } else { + ma_uint64 iFrame; + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + ma_uint32 iChannel; + for (iChannel = 0; iChannel < channels; iChannel += 1) { + dst_u8[iFrame*channels + iChannel] = src_u8[iChannel][iFrame]; } } } +} +#endif - return totalFramesRead; +void ma_pcm_interleave_u8(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_interleave_u8__reference(dst, src, frameCount, channels); +#else + ma_pcm_interleave_u8__optimized(dst, src, frameCount, channels); +#endif } -ma_uint64 ma_format_converter_read_deinterleaved(ma_format_converter* pConverter, ma_uint64 frameCount, void** ppSamplesOut, void* pUserData) + +static MA_INLINE void ma_pcm_deinterleave_u8__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) { - ma_uint64 totalFramesRead; - ma_uint32 sampleSizeIn; - ma_uint32 sampleSizeOut; - ma_uint8* ppNextSamplesOut[MA_MAX_CHANNELS]; + ma_uint8** dst_u8 = (ma_uint8**)dst; + const ma_uint8* src_u8 = (const ma_uint8*)src; - if (pConverter == NULL || ppSamplesOut == NULL) { - return 0; + ma_uint64 iFrame; + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + ma_uint32 iChannel; + for (iChannel = 0; iChannel < channels; iChannel += 1) { + dst_u8[iChannel][iFrame] = src_u8[iFrame*channels + iChannel]; + } } +} - totalFramesRead = 0; - sampleSizeIn = ma_get_bytes_per_sample(pConverter->config.formatIn); - sampleSizeOut = ma_get_bytes_per_sample(pConverter->config.formatOut); - - ma_copy_memory(ppNextSamplesOut, ppSamplesOut, sizeof(void*) * pConverter->config.channels); +static MA_INLINE void ma_pcm_deinterleave_u8__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) +{ + ma_pcm_deinterleave_u8__reference(dst, src, frameCount, channels); +} - if (pConverter->config.onRead != NULL) { - /* Input data is interleaved. */ - ma_uint32 maxFramesToReadAtATime; +void ma_pcm_deinterleave_u8(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_deinterleave_u8__reference(dst, src, frameCount, channels); +#else + ma_pcm_deinterleave_u8__optimized(dst, src, frameCount, channels); +#endif +} - MA_ALIGN(MA_SIMD_ALIGNMENT) ma_uint8 tempSamplesOfOutFormat[MA_MAX_CHANNELS * MA_MAX_PCM_SAMPLE_SIZE_IN_BYTES * 128]; - ma_assert(sizeof(tempSamplesOfOutFormat) <= 0xFFFFFFFF); - maxFramesToReadAtATime = sizeof(tempSamplesOfOutFormat) / sampleSizeIn / pConverter->config.channels; +/* s16 */ +static MA_INLINE void ma_pcm_s16_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_uint8* dst_u8 = (ma_uint8*)dst; + const ma_int16* src_s16 = (const ma_int16*)src; - while (totalFramesRead < frameCount) { - ma_uint32 iChannel; - ma_uint32 framesJustRead; - ma_uint64 framesRemaining = (frameCount - totalFramesRead); - ma_uint64 framesToReadRightNow = framesRemaining; - if (framesToReadRightNow > maxFramesToReadAtATime) { - framesToReadRightNow = maxFramesToReadAtATime; - } + if (ditherMode == ma_dither_mode_none) { + ma_uint64 i; + for (i = 0; i < count; i += 1) { + ma_int16 x = src_s16[i]; + x = x >> 8; + x = x + 128; + dst_u8[i] = (ma_uint8)x; + } + } else { + ma_uint64 i; + for (i = 0; i < count; i += 1) { + ma_int16 x = src_s16[i]; - if (pConverter->config.formatIn == pConverter->config.formatOut) { - /* Only de-interleaving. */ - framesJustRead = (ma_uint32)pConverter->config.onRead(pConverter, (ma_uint32)framesToReadRightNow, tempSamplesOfOutFormat, pUserData); - if (framesJustRead == 0) { - break; - } + /* Dither. Don't overflow. */ + ma_int32 dither = ma_dither_s32(ditherMode, -0x80, 0x7F); + if ((x + dither) <= 0x7FFF) { + x = (ma_int16)(x + dither); } else { - /* De-interleaving + Conversion. Convert first, then de-interleave. */ - MA_ALIGN(MA_SIMD_ALIGNMENT) ma_uint8 tempSamplesOfInFormat[sizeof(tempSamplesOfOutFormat)]; - - framesJustRead = (ma_uint32)pConverter->config.onRead(pConverter, (ma_uint32)framesToReadRightNow, tempSamplesOfInFormat, pUserData); - if (framesJustRead == 0) { - break; - } - - pConverter->onConvertPCM(tempSamplesOfOutFormat, tempSamplesOfInFormat, framesJustRead * pConverter->config.channels, pConverter->config.ditherMode); + x = 0x7FFF; } - pConverter->onDeinterleavePCM((void**)ppNextSamplesOut, tempSamplesOfOutFormat, framesJustRead, pConverter->config.channels); + x = x >> 8; + x = x + 128; + dst_u8[i] = (ma_uint8)x; + } + } +} - totalFramesRead += framesJustRead; - for (iChannel = 0; iChannel < pConverter->config.channels; ++iChannel) { - ppNextSamplesOut[iChannel] += framesJustRead * sampleSizeOut; - } +static MA_INLINE void ma_pcm_s16_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s16_to_u8__reference(dst, src, count, ditherMode); +} - if (framesJustRead < framesToReadRightNow) { - break; - } +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_s16_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE void ma_pcm_s16_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_s16_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode); +} +#endif + +void ma_pcm_s16_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_s16_to_u8__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_s16_to_u8__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_s16_to_u8__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_s16_to_u8__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode); } - } else { - /* Input data is deinterleaved. */ - if (pConverter->config.formatIn == pConverter->config.formatOut) { - /* Pass through. */ - while (totalFramesRead < frameCount) { - ma_uint32 iChannel; - ma_uint32 framesJustRead; - ma_uint64 framesRemaining = (frameCount - totalFramesRead); - ma_uint64 framesToReadRightNow = framesRemaining; - if (framesToReadRightNow > 0xFFFFFFFF) { - framesToReadRightNow = 0xFFFFFFFF; - } +#endif +} - framesJustRead = (ma_uint32)pConverter->config.onReadDeinterleaved(pConverter, (ma_uint32)framesToReadRightNow, (void**)ppNextSamplesOut, pUserData); - if (framesJustRead == 0) { - break; - } - totalFramesRead += framesJustRead; - for (iChannel = 0; iChannel < pConverter->config.channels; ++iChannel) { - ppNextSamplesOut[iChannel] += framesJustRead * sampleSizeOut; - } +void ma_pcm_s16_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + (void)ditherMode; + ma_copy_memory_64(dst, src, count * sizeof(ma_int16)); +} - if (framesJustRead < framesToReadRightNow) { - break; - } - } - } else { - /* Conversion required. */ - void* ppTemp[MA_MAX_CHANNELS]; - size_t splitBufferSize; - ma_uint32 maxFramesToReadAtATime; - MA_ALIGN(MA_SIMD_ALIGNMENT) ma_uint8 temp[MA_MAX_CHANNELS][MA_MAX_PCM_SAMPLE_SIZE_IN_BYTES * 128]; - ma_assert(sizeof(temp) <= 0xFFFFFFFF); +static MA_INLINE void ma_pcm_s16_to_s24__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_uint8* dst_s24 = (ma_uint8*)dst; + const ma_int16* src_s16 = (const ma_int16*)src; - ma_split_buffer(temp, sizeof(temp), pConverter->config.channels, MA_SIMD_ALIGNMENT, (void**)&ppTemp, &splitBufferSize); + ma_uint64 i; + for (i = 0; i < count; i += 1) { + dst_s24[i*3+0] = 0; + dst_s24[i*3+1] = (ma_uint8)(src_s16[i] & 0xFF); + dst_s24[i*3+2] = (ma_uint8)(src_s16[i] >> 8); + } - maxFramesToReadAtATime = (ma_uint32)(splitBufferSize / sampleSizeIn); + (void)ditherMode; +} - while (totalFramesRead < frameCount) { - ma_uint32 iChannel; - ma_uint32 framesJustRead; - ma_uint64 framesRemaining = (frameCount - totalFramesRead); - ma_uint64 framesToReadRightNow = framesRemaining; - if (framesToReadRightNow > maxFramesToReadAtATime) { - framesToReadRightNow = maxFramesToReadAtATime; - } +static MA_INLINE void ma_pcm_s16_to_s24__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s16_to_s24__reference(dst, src, count, ditherMode); +} + +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_s16_to_s24__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s16_to_s24__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE void ma_pcm_s16_to_s24__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s16_to_s24__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_s16_to_s24__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s16_to_s24__optimized(dst, src, count, ditherMode); +} +#endif - framesJustRead = (ma_uint32)pConverter->config.onReadDeinterleaved(pConverter, (ma_uint32)framesToReadRightNow, ppTemp, pUserData); - if (framesJustRead == 0) { - break; - } +void ma_pcm_s16_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_s16_to_s24__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_s16_to_s24__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_s16_to_s24__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_s16_to_s24__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_s16_to_s24__optimized(dst, src, count, ditherMode); + } +#endif +} - for (iChannel = 0; iChannel < pConverter->config.channels; iChannel += 1) { - pConverter->onConvertPCM(ppNextSamplesOut[iChannel], ppTemp[iChannel], framesJustRead, pConverter->config.ditherMode); - ppNextSamplesOut[iChannel] += framesJustRead * sampleSizeOut; - } - totalFramesRead += framesJustRead; +static MA_INLINE void ma_pcm_s16_to_s32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_int32* dst_s32 = (ma_int32*)dst; + const ma_int16* src_s16 = (const ma_int16*)src; - if (framesJustRead < framesToReadRightNow) { - break; - } - } - } + ma_uint64 i; + for (i = 0; i < count; i += 1) { + dst_s32[i] = src_s16[i] << 16; } - return totalFramesRead; + (void)ditherMode; } - -ma_format_converter_config ma_format_converter_config_init_new() +static MA_INLINE void ma_pcm_s16_to_s32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - ma_format_converter_config config; - ma_zero_object(&config); - - return config; + ma_pcm_s16_to_s32__reference(dst, src, count, ditherMode); } -ma_format_converter_config ma_format_converter_config_init(ma_format formatIn, ma_format formatOut, ma_uint32 channels, ma_format_converter_read_proc onRead, void* pUserData) +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_s16_to_s32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - ma_format_converter_config config = ma_format_converter_config_init_new(); - config.formatIn = formatIn; - config.formatOut = formatOut; - config.channels = channels; - config.onRead = onRead; - config.onReadDeinterleaved = NULL; - config.pUserData = pUserData; - - return config; + ma_pcm_s16_to_s32__optimized(dst, src, count, ditherMode); } - -ma_format_converter_config ma_format_converter_config_init_deinterleaved(ma_format formatIn, ma_format formatOut, ma_uint32 channels, ma_format_converter_read_deinterleaved_proc onReadDeinterleaved, void* pUserData) +#endif +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE void ma_pcm_s16_to_s32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s16_to_s32__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_s16_to_s32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - ma_format_converter_config config = ma_format_converter_config_init(formatIn, formatOut, channels, NULL, pUserData); - config.onReadDeinterleaved = onReadDeinterleaved; + ma_pcm_s16_to_s32__optimized(dst, src, count, ditherMode); +} +#endif - return config; +void ma_pcm_s16_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_s16_to_s32__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_s16_to_s32__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_s16_to_s32__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_s16_to_s32__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_s16_to_s32__optimized(dst, src, count, ditherMode); + } +#endif } +static MA_INLINE void ma_pcm_s16_to_f32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + float* dst_f32 = (float*)dst; + const ma_int16* src_s16 = (const ma_int16*)src; -/************************************************************************************************************************************************************** + ma_uint64 i; + for (i = 0; i < count; i += 1) { + float x = (float)src_s16[i]; -Channel Routing +#if 0 + /* The accurate way. */ + x = x + 32768.0f; /* -32768..32767 to 0..65535 */ + x = x * 0.00003051804379339284f; /* 0..65535 to 0..2 */ + x = x - 1; /* 0..2 to -1..1 */ +#else + /* The fast way. */ + x = x * 0.000030517578125f; /* -32768..32767 to -1..0.999969482421875 */ +#endif -**************************************************************************************************************************************************************/ + dst_f32[i] = x; + } -/* --X = Left, +X = Right --Y = Bottom, +Y = Top --Z = Front, +Z = Back -*/ -typedef struct -{ - float x; - float y; - float z; -} ma_vec3; + (void)ditherMode; +} -static MA_INLINE ma_vec3 ma_vec3f(float x, float y, float z) +static MA_INLINE void ma_pcm_s16_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - ma_vec3 r; - r.x = x; - r.y = y; - r.z = z; - - return r; + ma_pcm_s16_to_f32__reference(dst, src, count, ditherMode); } -static MA_INLINE ma_vec3 ma_vec3_add(ma_vec3 a, ma_vec3 b) +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_s16_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - return ma_vec3f( - a.x + b.x, - a.y + b.y, - a.z + b.z - ); + ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode); } - -static MA_INLINE ma_vec3 ma_vec3_sub(ma_vec3 a, ma_vec3 b) +#endif +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE void ma_pcm_s16_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - return ma_vec3f( - a.x - b.x, - a.y - b.y, - a.z - b.z - ); + ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode); } - -static MA_INLINE ma_vec3 ma_vec3_mul(ma_vec3 a, ma_vec3 b) +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_s16_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - return ma_vec3f( - a.x * b.x, - a.y * b.y, - a.z * b.z - ); + ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode); } +#endif -static MA_INLINE ma_vec3 ma_vec3_div(ma_vec3 a, ma_vec3 b) +void ma_pcm_s16_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - return ma_vec3f( - a.x / b.x, - a.y / b.y, - a.z / b.z - ); +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_s16_to_f32__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_s16_to_f32__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_s16_to_f32__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_s16_to_f32__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode); + } +#endif } -static MA_INLINE float ma_vec3_dot(ma_vec3 a, ma_vec3 b) + +static MA_INLINE void ma_pcm_interleave_s16__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) { - return a.x*b.x + a.y*b.y + a.z*b.z; + ma_int16* dst_s16 = (ma_int16*)dst; + const ma_int16** src_s16 = (const ma_int16**)src; + + ma_uint64 iFrame; + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + ma_uint32 iChannel; + for (iChannel = 0; iChannel < channels; iChannel += 1) { + dst_s16[iFrame*channels + iChannel] = src_s16[iChannel][iFrame]; + } + } } -static MA_INLINE float ma_vec3_length2(ma_vec3 a) +static MA_INLINE void ma_pcm_interleave_s16__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) { - return ma_vec3_dot(a, a); + ma_pcm_interleave_s16__reference(dst, src, frameCount, channels); } -static MA_INLINE float ma_vec3_length(ma_vec3 a) +void ma_pcm_interleave_s16(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) { - return (float)sqrt(ma_vec3_length2(a)); +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_interleave_s16__reference(dst, src, frameCount, channels); +#else + ma_pcm_interleave_s16__optimized(dst, src, frameCount, channels); +#endif } -static MA_INLINE ma_vec3 ma_vec3_normalize(ma_vec3 a) -{ - float len = 1 / ma_vec3_length(a); - ma_vec3 r; - r.x = a.x * len; - r.y = a.y * len; - r.z = a.z * len; +static MA_INLINE void ma_pcm_deinterleave_s16__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) +{ + ma_int16** dst_s16 = (ma_int16**)dst; + const ma_int16* src_s16 = (const ma_int16*)src; - return r; + ma_uint64 iFrame; + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + ma_uint32 iChannel; + for (iChannel = 0; iChannel < channels; iChannel += 1) { + dst_s16[iChannel][iFrame] = src_s16[iFrame*channels + iChannel]; + } + } } -static MA_INLINE float ma_vec3_distance(ma_vec3 a, ma_vec3 b) +static MA_INLINE void ma_pcm_deinterleave_s16__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) { - return ma_vec3_length(ma_vec3_sub(a, b)); + ma_pcm_deinterleave_s16__reference(dst, src, frameCount, channels); } +void ma_pcm_deinterleave_s16(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_deinterleave_s16__reference(dst, src, frameCount, channels); +#else + ma_pcm_deinterleave_s16__optimized(dst, src, frameCount, channels); +#endif +} -#define MA_PLANE_LEFT 0 -#define MA_PLANE_RIGHT 1 -#define MA_PLANE_FRONT 2 -#define MA_PLANE_BACK 3 -#define MA_PLANE_BOTTOM 4 -#define MA_PLANE_TOP 5 - -float g_maChannelPlaneRatios[MA_CHANNEL_POSITION_COUNT][6] = { - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_NONE */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_MONO */ - { 0.5f, 0.0f, 0.5f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_LEFT */ - { 0.0f, 0.5f, 0.5f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_RIGHT */ - { 0.0f, 0.0f, 1.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_CENTER */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_LFE */ - { 0.5f, 0.0f, 0.0f, 0.5f, 0.0f, 0.0f}, /* MA_CHANNEL_BACK_LEFT */ - { 0.0f, 0.5f, 0.0f, 0.5f, 0.0f, 0.0f}, /* MA_CHANNEL_BACK_RIGHT */ - { 0.25f, 0.0f, 0.75f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_LEFT_CENTER */ - { 0.0f, 0.25f, 0.75f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_RIGHT_CENTER */ - { 0.0f, 0.0f, 0.0f, 1.0f, 0.0f, 0.0f}, /* MA_CHANNEL_BACK_CENTER */ - { 1.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_SIDE_LEFT */ - { 0.0f, 1.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_SIDE_RIGHT */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 1.0f}, /* MA_CHANNEL_TOP_CENTER */ - { 0.33f, 0.0f, 0.33f, 0.0f, 0.0f, 0.34f}, /* MA_CHANNEL_TOP_FRONT_LEFT */ - { 0.0f, 0.0f, 0.5f, 0.0f, 0.0f, 0.5f}, /* MA_CHANNEL_TOP_FRONT_CENTER */ - { 0.0f, 0.33f, 0.33f, 0.0f, 0.0f, 0.34f}, /* MA_CHANNEL_TOP_FRONT_RIGHT */ - { 0.33f, 0.0f, 0.0f, 0.33f, 0.0f, 0.34f}, /* MA_CHANNEL_TOP_BACK_LEFT */ - { 0.0f, 0.0f, 0.0f, 0.5f, 0.0f, 0.5f}, /* MA_CHANNEL_TOP_BACK_CENTER */ - { 0.0f, 0.33f, 0.0f, 0.33f, 0.0f, 0.34f}, /* MA_CHANNEL_TOP_BACK_RIGHT */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_0 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_1 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_2 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_3 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_4 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_5 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_6 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_7 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_8 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_9 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_10 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_11 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_12 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_13 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_14 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_15 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_16 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_17 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_18 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_19 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_20 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_21 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_22 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_23 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_24 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_25 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_26 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_27 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_28 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_29 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_30 */ - { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_31 */ -}; -float ma_calculate_channel_position_planar_weight(ma_channel channelPositionA, ma_channel channelPositionB) +/* s24 */ +static MA_INLINE void ma_pcm_s24_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - /* - Imagine the following simplified example: You have a single input speaker which is the front/left speaker which you want to convert to - the following output configuration: - - - front/left - - side/left - - back/left - - The front/left output is easy - it the same speaker position so it receives the full contribution of the front/left input. The amount - of contribution to apply to the side/left and back/left speakers, however, is a bit more complicated. - - Imagine the front/left speaker as emitting audio from two planes - the front plane and the left plane. You can think of the front/left - speaker emitting half of it's total volume from the front, and the other half from the left. Since part of it's volume is being emitted - from the left side, and the side/left and back/left channels also emit audio from the left plane, one would expect that they would - receive some amount of contribution from front/left speaker. The amount of contribution depends on how many planes are shared between - the two speakers. Note that in the examples below I've added a top/front/left speaker as an example just to show how the math works - across 3 spatial dimensions. - - The first thing to do is figure out how each speaker's volume is spread over each of plane: - - front/left: 2 planes (front and left) = 1/2 = half it's total volume on each plane - - side/left: 1 plane (left only) = 1/1 = entire volume from left plane - - back/left: 2 planes (back and left) = 1/2 = half it's total volume on each plane - - top/front/left: 3 planes (top, front and left) = 1/3 = one third it's total volume on each plane - - The amount of volume each channel contributes to each of it's planes is what controls how much it is willing to given and take to other - channels on the same plane. The volume that is willing to the given by one channel is multiplied by the volume that is willing to be - taken by the other to produce the final contribution. - */ + ma_uint8* dst_u8 = (ma_uint8*)dst; + const ma_uint8* src_s24 = (const ma_uint8*)src; - /* Contribution = Sum(Volume to Give * Volume to Take) */ - float contribution = - g_maChannelPlaneRatios[channelPositionA][0] * g_maChannelPlaneRatios[channelPositionB][0] + - g_maChannelPlaneRatios[channelPositionA][1] * g_maChannelPlaneRatios[channelPositionB][1] + - g_maChannelPlaneRatios[channelPositionA][2] * g_maChannelPlaneRatios[channelPositionB][2] + - g_maChannelPlaneRatios[channelPositionA][3] * g_maChannelPlaneRatios[channelPositionB][3] + - g_maChannelPlaneRatios[channelPositionA][4] * g_maChannelPlaneRatios[channelPositionB][4] + - g_maChannelPlaneRatios[channelPositionA][5] * g_maChannelPlaneRatios[channelPositionB][5]; + if (ditherMode == ma_dither_mode_none) { + ma_uint64 i; + for (i = 0; i < count; i += 1) { + ma_int8 x = (ma_int8)src_s24[i*3 + 2] + 128; + dst_u8[i] = (ma_uint8)x; + } + } else { + ma_uint64 i; + for (i = 0; i < count; i += 1) { + ma_int32 x = (ma_int32)(((ma_uint32)(src_s24[i*3+0]) << 8) | ((ma_uint32)(src_s24[i*3+1]) << 16) | ((ma_uint32)(src_s24[i*3+2])) << 24); - return contribution; + /* Dither. Don't overflow. */ + ma_int32 dither = ma_dither_s32(ditherMode, -0x800000, 0x7FFFFF); + if ((ma_int64)x + dither <= 0x7FFFFFFF) { + x = x + dither; + } else { + x = 0x7FFFFFFF; + } + + x = x >> 24; + x = x + 128; + dst_u8[i] = (ma_uint8)x; + } + } } -float ma_channel_router__calculate_input_channel_planar_weight(const ma_channel_router* pRouter, ma_channel channelPositionIn, ma_channel channelPositionOut) +static MA_INLINE void ma_pcm_s24_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - ma_assert(pRouter != NULL); - (void)pRouter; + ma_pcm_s24_to_u8__reference(dst, src, count, ditherMode); +} - return ma_calculate_channel_position_planar_weight(channelPositionIn, channelPositionOut); +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_s24_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE void ma_pcm_s24_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode); } +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_s24_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode); +} +#endif -ma_bool32 ma_channel_router__is_spatial_channel_position(const ma_channel_router* pRouter, ma_channel channelPosition) +void ma_pcm_s24_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - int i; +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_s24_to_u8__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_s24_to_u8__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_s24_to_u8__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_s24_to_u8__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode); + } +#endif +} - ma_assert(pRouter != NULL); - (void)pRouter; - if (channelPosition == MA_CHANNEL_NONE || channelPosition == MA_CHANNEL_MONO || channelPosition == MA_CHANNEL_LFE) { - return MA_FALSE; - } +static MA_INLINE void ma_pcm_s24_to_s16__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_int16* dst_s16 = (ma_int16*)dst; + const ma_uint8* src_s24 = (const ma_uint8*)src; - for (i = 0; i < 6; ++i) { - if (g_maChannelPlaneRatios[channelPosition][i] != 0) { - return MA_TRUE; + if (ditherMode == ma_dither_mode_none) { + ma_uint64 i; + for (i = 0; i < count; i += 1) { + ma_uint16 dst_lo = ((ma_uint16)src_s24[i*3 + 1]); + ma_uint16 dst_hi = ((ma_uint16)src_s24[i*3 + 2]) << 8; + dst_s16[i] = (ma_int16)dst_lo | dst_hi; + } + } else { + ma_uint64 i; + for (i = 0; i < count; i += 1) { + ma_int32 x = (ma_int32)(((ma_uint32)(src_s24[i*3+0]) << 8) | ((ma_uint32)(src_s24[i*3+1]) << 16) | ((ma_uint32)(src_s24[i*3+2])) << 24); + + /* Dither. Don't overflow. */ + ma_int32 dither = ma_dither_s32(ditherMode, -0x8000, 0x7FFF); + if ((ma_int64)x + dither <= 0x7FFFFFFF) { + x = x + dither; + } else { + x = 0x7FFFFFFF; + } + + x = x >> 16; + dst_s16[i] = (ma_int16)x; } } +} - return MA_FALSE; +static MA_INLINE void ma_pcm_s24_to_s16__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s24_to_s16__reference(dst, src, count, ditherMode); } -ma_result ma_channel_router_init(const ma_channel_router_config* pConfig, ma_channel_router* pRouter) +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_s24_to_s16__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - ma_uint32 iChannelIn; - ma_uint32 iChannelOut; + ma_pcm_s24_to_s16__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE void ma_pcm_s24_to_s16__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s24_to_s16__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_s24_to_s16__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s24_to_s16__optimized(dst, src, count, ditherMode); +} +#endif - if (pRouter == NULL) { - return MA_INVALID_ARGS; - } +void ma_pcm_s24_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_s24_to_s16__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_s24_to_s16__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_s24_to_s16__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_s24_to_s16__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_s24_to_s16__optimized(dst, src, count, ditherMode); + } +#endif +} - ma_zero_object(pRouter); - if (pConfig == NULL) { - return MA_INVALID_ARGS; - } - if (pConfig->onReadDeinterleaved == NULL) { - return MA_INVALID_ARGS; - } +void ma_pcm_s24_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + (void)ditherMode; - if (!ma_channel_map_valid(pConfig->channelsIn, pConfig->channelMapIn)) { - return MA_INVALID_ARGS; /* Invalid input channel map. */ - } - if (!ma_channel_map_valid(pConfig->channelsOut, pConfig->channelMapOut)) { - return MA_INVALID_ARGS; /* Invalid output channel map. */ - } + ma_copy_memory_64(dst, src, count * 3); +} - pRouter->config = *pConfig; - /* SIMD */ - pRouter->useSSE2 = ma_has_sse2() && !pConfig->noSSE2; - pRouter->useAVX2 = ma_has_avx2() && !pConfig->noAVX2; - pRouter->useAVX512 = ma_has_avx512f() && !pConfig->noAVX512; - pRouter->useNEON = ma_has_neon() && !pConfig->noNEON; +static MA_INLINE void ma_pcm_s24_to_s32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_int32* dst_s32 = (ma_int32*)dst; + const ma_uint8* src_s24 = (const ma_uint8*)src; - /* If the input and output channels and channel maps are the same we should use a passthrough. */ - if (pRouter->config.channelsIn == pRouter->config.channelsOut) { - if (ma_channel_map_equal(pRouter->config.channelsIn, pRouter->config.channelMapIn, pRouter->config.channelMapOut)) { - pRouter->isPassthrough = MA_TRUE; - } - if (ma_channel_map_blank(pRouter->config.channelsIn, pRouter->config.channelMapIn) || ma_channel_map_blank(pRouter->config.channelsOut, pRouter->config.channelMapOut)) { - pRouter->isPassthrough = MA_TRUE; - } + ma_uint64 i; + for (i = 0; i < count; i += 1) { + dst_s32[i] = (ma_int32)(((ma_uint32)(src_s24[i*3+0]) << 8) | ((ma_uint32)(src_s24[i*3+1]) << 16) | ((ma_uint32)(src_s24[i*3+2])) << 24); } - /* - We can use a simple case for expanding the mono channel. This will when expanding a mono input into any output so long - as no LFE is present in the output. - */ - if (!pRouter->isPassthrough) { - if (pRouter->config.channelsIn == 1 && pRouter->config.channelMapIn[0] == MA_CHANNEL_MONO) { - /* Optimal case if no LFE is in the output channel map. */ - pRouter->isSimpleMonoExpansion = MA_TRUE; - if (ma_channel_map_contains_channel_position(pRouter->config.channelsOut, pRouter->config.channelMapOut, MA_CHANNEL_LFE)) { - pRouter->isSimpleMonoExpansion = MA_FALSE; - } - } - } + (void)ditherMode; +} - /* Another optimized case is stereo to mono. */ - if (!pRouter->isPassthrough) { - if (pRouter->config.channelsOut == 1 && pRouter->config.channelMapOut[0] == MA_CHANNEL_MONO && pRouter->config.channelsIn == 2) { - /* Optimal case if no LFE is in the input channel map. */ - pRouter->isStereoToMono = MA_TRUE; - if (ma_channel_map_contains_channel_position(pRouter->config.channelsIn, pRouter->config.channelMapIn, MA_CHANNEL_LFE)) { - pRouter->isStereoToMono = MA_FALSE; - } +static MA_INLINE void ma_pcm_s24_to_s32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s24_to_s32__reference(dst, src, count, ditherMode); +} + +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_s24_to_s32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s24_to_s32__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE void ma_pcm_s24_to_s32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s24_to_s32__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_s24_to_s32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s24_to_s32__optimized(dst, src, count, ditherMode); +} +#endif + +void ma_pcm_s24_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_s24_to_s32__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_s24_to_s32__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_s24_to_s32__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_s24_to_s32__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_s24_to_s32__optimized(dst, src, count, ditherMode); } - } +#endif +} - /* - Here is where we do a bit of pre-processing to know how each channel should be combined to make up the output. Rules: - - 1) If it's a passthrough, do nothing - it's just a simple memcpy(). - 2) If the channel counts are the same and every channel position in the input map is present in the output map, use a - simple shuffle. An example might be different 5.1 channel layouts. - 3) Otherwise channels are blended based on spatial locality. - */ - if (!pRouter->isPassthrough) { - if (pRouter->config.channelsIn == pRouter->config.channelsOut) { - ma_bool32 areAllChannelPositionsPresent = MA_TRUE; - for (iChannelIn = 0; iChannelIn < pRouter->config.channelsIn; ++iChannelIn) { - ma_bool32 isInputChannelPositionInOutput = MA_FALSE; - for (iChannelOut = 0; iChannelOut < pRouter->config.channelsOut; ++iChannelOut) { - if (pRouter->config.channelMapIn[iChannelIn] == pRouter->config.channelMapOut[iChannelOut]) { - isInputChannelPositionInOutput = MA_TRUE; - break; - } - } - if (!isInputChannelPositionInOutput) { - areAllChannelPositionsPresent = MA_FALSE; - break; - } - } +static MA_INLINE void ma_pcm_s24_to_f32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + float* dst_f32 = (float*)dst; + const ma_uint8* src_s24 = (const ma_uint8*)src; - if (areAllChannelPositionsPresent) { - pRouter->isSimpleShuffle = MA_TRUE; + ma_uint64 i; + for (i = 0; i < count; i += 1) { + float x = (float)(((ma_int32)(((ma_uint32)(src_s24[i*3+0]) << 8) | ((ma_uint32)(src_s24[i*3+1]) << 16) | ((ma_uint32)(src_s24[i*3+2])) << 24)) >> 8); - /* - All the router will be doing is rearranging channels which means all we need to do is use a shuffling table which is just - a mapping between the index of the input channel to the index of the output channel. - */ - for (iChannelIn = 0; iChannelIn < pRouter->config.channelsIn; ++iChannelIn) { - for (iChannelOut = 0; iChannelOut < pRouter->config.channelsOut; ++iChannelOut) { - if (pRouter->config.channelMapIn[iChannelIn] == pRouter->config.channelMapOut[iChannelOut]) { - pRouter->shuffleTable[iChannelIn] = (ma_uint8)iChannelOut; - break; - } - } - } - } - } - } +#if 0 + /* The accurate way. */ + x = x + 8388608.0f; /* -8388608..8388607 to 0..16777215 */ + x = x * 0.00000011920929665621f; /* 0..16777215 to 0..2 */ + x = x - 1; /* 0..2 to -1..1 */ +#else + /* The fast way. */ + x = x * 0.00000011920928955078125f; /* -8388608..8388607 to -1..0.999969482421875 */ +#endif + dst_f32[i] = x; + } - /* - Here is where weights are calculated. Note that we calculate the weights at all times, even when using a passthrough and simple - shuffling. We use different algorithms for calculating weights depending on our mixing mode. - - In simple mode we don't do any blending (except for converting between mono, which is done in a later step). Instead we just - map 1:1 matching channels. In this mode, if no channels in the input channel map correspond to anything in the output channel - map, nothing will be heard! - */ + (void)ditherMode; +} - /* In all cases we need to make sure all channels that are present in both channel maps have a 1:1 mapping. */ - for (iChannelIn = 0; iChannelIn < pRouter->config.channelsIn; ++iChannelIn) { - ma_channel channelPosIn = pRouter->config.channelMapIn[iChannelIn]; +static MA_INLINE void ma_pcm_s24_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s24_to_f32__reference(dst, src, count, ditherMode); +} - for (iChannelOut = 0; iChannelOut < pRouter->config.channelsOut; ++iChannelOut) { - ma_channel channelPosOut = pRouter->config.channelMapOut[iChannelOut]; +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_s24_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE void ma_pcm_s24_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_s24_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode); +} +#endif - if (channelPosIn == channelPosOut) { - pRouter->config.weights[iChannelIn][iChannelOut] = 1; - } +void ma_pcm_s24_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_s24_to_f32__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_s24_to_f32__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_s24_to_f32__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_s24_to_f32__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode); } - } +#endif +} - /* - The mono channel is accumulated on all other channels, except LFE. Make sure in this loop we exclude output mono channels since - they were handled in the pass above. - */ - for (iChannelIn = 0; iChannelIn < pRouter->config.channelsIn; ++iChannelIn) { - ma_channel channelPosIn = pRouter->config.channelMapIn[iChannelIn]; - if (channelPosIn == MA_CHANNEL_MONO) { - for (iChannelOut = 0; iChannelOut < pRouter->config.channelsOut; ++iChannelOut) { - ma_channel channelPosOut = pRouter->config.channelMapOut[iChannelOut]; +static MA_INLINE void ma_pcm_interleave_s24__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) +{ + ma_uint8* dst8 = (ma_uint8*)dst; + const ma_uint8** src8 = (const ma_uint8**)src; - if (channelPosOut != MA_CHANNEL_NONE && channelPosOut != MA_CHANNEL_MONO && channelPosOut != MA_CHANNEL_LFE) { - pRouter->config.weights[iChannelIn][iChannelOut] = 1; - } - } + ma_uint64 iFrame; + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + ma_uint32 iChannel; + for (iChannel = 0; iChannel < channels; iChannel += 1) { + dst8[iFrame*3*channels + iChannel*3 + 0] = src8[iChannel][iFrame*3 + 0]; + dst8[iFrame*3*channels + iChannel*3 + 1] = src8[iChannel][iFrame*3 + 1]; + dst8[iFrame*3*channels + iChannel*3 + 2] = src8[iChannel][iFrame*3 + 2]; } } +} - /* The output mono channel is the average of all non-none, non-mono and non-lfe input channels. */ - { - ma_uint32 len = 0; - for (iChannelIn = 0; iChannelIn < pRouter->config.channelsIn; ++iChannelIn) { - ma_channel channelPosIn = pRouter->config.channelMapIn[iChannelIn]; - - if (channelPosIn != MA_CHANNEL_NONE && channelPosIn != MA_CHANNEL_MONO && channelPosIn != MA_CHANNEL_LFE) { - len += 1; - } - } +static MA_INLINE void ma_pcm_interleave_s24__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) +{ + ma_pcm_interleave_s24__reference(dst, src, frameCount, channels); +} - if (len > 0) { - float monoWeight = 1.0f / len; +void ma_pcm_interleave_s24(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_interleave_s24__reference(dst, src, frameCount, channels); +#else + ma_pcm_interleave_s24__optimized(dst, src, frameCount, channels); +#endif +} - for (iChannelOut = 0; iChannelOut < pRouter->config.channelsOut; ++iChannelOut) { - ma_channel channelPosOut = pRouter->config.channelMapOut[iChannelOut]; - if (channelPosOut == MA_CHANNEL_MONO) { - for (iChannelIn = 0; iChannelIn < pRouter->config.channelsIn; ++iChannelIn) { - ma_channel channelPosIn = pRouter->config.channelMapIn[iChannelIn]; +static MA_INLINE void ma_pcm_deinterleave_s24__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) +{ + ma_uint8** dst8 = (ma_uint8**)dst; + const ma_uint8* src8 = (const ma_uint8*)src; - if (channelPosIn != MA_CHANNEL_NONE && channelPosIn != MA_CHANNEL_MONO && channelPosIn != MA_CHANNEL_LFE) { - pRouter->config.weights[iChannelIn][iChannelOut] += monoWeight; - } - } - } - } + ma_uint32 iFrame; + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + ma_uint32 iChannel; + for (iChannel = 0; iChannel < channels; iChannel += 1) { + dst8[iChannel][iFrame*3 + 0] = src8[iFrame*3*channels + iChannel*3 + 0]; + dst8[iChannel][iFrame*3 + 1] = src8[iFrame*3*channels + iChannel*3 + 1]; + dst8[iChannel][iFrame*3 + 2] = src8[iFrame*3*channels + iChannel*3 + 2]; } } +} +static MA_INLINE void ma_pcm_deinterleave_s24__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) +{ + ma_pcm_deinterleave_s24__reference(dst, src, frameCount, channels); +} - /* Input and output channels that are not present on the other side need to be blended in based on spatial locality. */ - switch (pRouter->config.mixingMode) - { - case ma_channel_mix_mode_rectangular: - { - /* Unmapped input channels. */ - for (iChannelIn = 0; iChannelIn < pRouter->config.channelsIn; ++iChannelIn) { - ma_channel channelPosIn = pRouter->config.channelMapIn[iChannelIn]; - - if (ma_channel_router__is_spatial_channel_position(pRouter, channelPosIn)) { - if (!ma_channel_map_contains_channel_position(pRouter->config.channelsOut, pRouter->config.channelMapOut, channelPosIn)) { - for (iChannelOut = 0; iChannelOut < pRouter->config.channelsOut; ++iChannelOut) { - ma_channel channelPosOut = pRouter->config.channelMapOut[iChannelOut]; - - if (ma_channel_router__is_spatial_channel_position(pRouter, channelPosOut)) { - float weight = 0; - if (pRouter->config.mixingMode == ma_channel_mix_mode_planar_blend) { - weight = ma_channel_router__calculate_input_channel_planar_weight(pRouter, channelPosIn, channelPosOut); - } +void ma_pcm_deinterleave_s24(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_deinterleave_s24__reference(dst, src, frameCount, channels); +#else + ma_pcm_deinterleave_s24__optimized(dst, src, frameCount, channels); +#endif +} - /* Only apply the weight if we haven't already got some contribution from the respective channels. */ - if (pRouter->config.weights[iChannelIn][iChannelOut] == 0) { - pRouter->config.weights[iChannelIn][iChannelOut] = weight; - } - } - } - } - } - } - /* Unmapped output channels. */ - for (iChannelOut = 0; iChannelOut < pRouter->config.channelsOut; ++iChannelOut) { - ma_channel channelPosOut = pRouter->config.channelMapOut[iChannelOut]; - if (ma_channel_router__is_spatial_channel_position(pRouter, channelPosOut)) { - if (!ma_channel_map_contains_channel_position(pRouter->config.channelsIn, pRouter->config.channelMapIn, channelPosOut)) { - for (iChannelIn = 0; iChannelIn < pRouter->config.channelsIn; ++iChannelIn) { - ma_channel channelPosIn = pRouter->config.channelMapIn[iChannelIn]; +/* s32 */ +static MA_INLINE void ma_pcm_s32_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_uint8* dst_u8 = (ma_uint8*)dst; + const ma_int32* src_s32 = (const ma_int32*)src; - if (ma_channel_router__is_spatial_channel_position(pRouter, channelPosIn)) { - float weight = 0; - if (pRouter->config.mixingMode == ma_channel_mix_mode_planar_blend) { - weight = ma_channel_router__calculate_input_channel_planar_weight(pRouter, channelPosIn, channelPosOut); - } + if (ditherMode == ma_dither_mode_none) { + ma_uint64 i; + for (i = 0; i < count; i += 1) { + ma_int32 x = src_s32[i]; + x = x >> 24; + x = x + 128; + dst_u8[i] = (ma_uint8)x; + } + } else { + ma_uint64 i; + for (i = 0; i < count; i += 1) { + ma_int32 x = src_s32[i]; - /* Only apply the weight if we haven't already got some contribution from the respective channels. */ - if (pRouter->config.weights[iChannelIn][iChannelOut] == 0) { - pRouter->config.weights[iChannelIn][iChannelOut] = weight; - } - } - } - } - } + /* Dither. Don't overflow. */ + ma_int32 dither = ma_dither_s32(ditherMode, -0x800000, 0x7FFFFF); + if ((ma_int64)x + dither <= 0x7FFFFFFF) { + x = x + dither; + } else { + x = 0x7FFFFFFF; } - } break; - - case ma_channel_mix_mode_custom_weights: - case ma_channel_mix_mode_simple: - default: - { - /* Fallthrough. */ - } break; + + x = x >> 24; + x = x + 128; + dst_u8[i] = (ma_uint8)x; + } } - - return MA_SUCCESS; } -static MA_INLINE ma_bool32 ma_channel_router__can_use_sse2(ma_channel_router* pRouter, const float* pSamplesOut, const float* pSamplesIn) +static MA_INLINE void ma_pcm_s32_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - return pRouter->useSSE2 && (((ma_uintptr)pSamplesOut & 15) == 0) && (((ma_uintptr)pSamplesIn & 15) == 0); + ma_pcm_s32_to_u8__reference(dst, src, count, ditherMode); } -static MA_INLINE ma_bool32 ma_channel_router__can_use_avx2(ma_channel_router* pRouter, const float* pSamplesOut, const float* pSamplesIn) +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_s32_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - return pRouter->useAVX2 && (((ma_uintptr)pSamplesOut & 31) == 0) && (((ma_uintptr)pSamplesIn & 31) == 0); + ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode); } - -static MA_INLINE ma_bool32 ma_channel_router__can_use_avx512(ma_channel_router* pRouter, const float* pSamplesOut, const float* pSamplesIn) +#endif +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE void ma_pcm_s32_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - return pRouter->useAVX512 && (((ma_uintptr)pSamplesOut & 63) == 0) && (((ma_uintptr)pSamplesIn & 63) == 0); + ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode); } - -static MA_INLINE ma_bool32 ma_channel_router__can_use_neon(ma_channel_router* pRouter, const float* pSamplesOut, const float* pSamplesIn) +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_s32_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - return pRouter->useNEON && (((ma_uintptr)pSamplesOut & 15) == 0) && (((ma_uintptr)pSamplesIn & 15) == 0); + ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode); } +#endif -void ma_channel_router__do_routing(ma_channel_router* pRouter, ma_uint64 frameCount, float** ppSamplesOut, const float** ppSamplesIn) +void ma_pcm_s32_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - ma_uint32 iChannelIn; - ma_uint32 iChannelOut; - - ma_assert(pRouter != NULL); - ma_assert(pRouter->isPassthrough == MA_FALSE); - - if (pRouter->isSimpleShuffle) { - /* A shuffle is just a re-arrangement of channels and does not require any arithmetic. */ - ma_assert(pRouter->config.channelsIn == pRouter->config.channelsOut); - for (iChannelIn = 0; iChannelIn < pRouter->config.channelsIn; ++iChannelIn) { - iChannelOut = pRouter->shuffleTable[iChannelIn]; - ma_copy_memory_64(ppSamplesOut[iChannelOut], ppSamplesIn[iChannelIn], frameCount * sizeof(float)); - } - } else if (pRouter->isSimpleMonoExpansion) { - /* Simple case for expanding from mono. */ - if (pRouter->config.channelsOut == 2) { - ma_uint64 iFrame; - for (iFrame = 0; iFrame < frameCount; ++iFrame) { - ppSamplesOut[0][iFrame] = ppSamplesIn[0][iFrame]; - ppSamplesOut[1][iFrame] = ppSamplesIn[0][iFrame]; - } - } else { - for (iChannelOut = 0; iChannelOut < pRouter->config.channelsOut; ++iChannelOut) { - ma_uint64 iFrame; - for (iFrame = 0; iFrame < frameCount; ++iFrame) { - ppSamplesOut[iChannelOut][iFrame] = ppSamplesIn[0][iFrame]; - } - } +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_s32_to_u8__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_s32_to_u8__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_s32_to_u8__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_s32_to_u8__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode); } - } else if (pRouter->isStereoToMono) { - ma_uint64 iFrame; +#endif +} - /* Simple case for going from stereo to mono. */ - ma_assert(pRouter->config.channelsIn == 2); - ma_assert(pRouter->config.channelsOut == 1); - for (iFrame = 0; iFrame < frameCount; ++iFrame) { - ppSamplesOut[0][iFrame] = (ppSamplesIn[0][iFrame] + ppSamplesIn[1][iFrame]) * 0.5f; +static MA_INLINE void ma_pcm_s32_to_s16__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_int16* dst_s16 = (ma_int16*)dst; + const ma_int32* src_s32 = (const ma_int32*)src; + + if (ditherMode == ma_dither_mode_none) { + ma_uint64 i; + for (i = 0; i < count; i += 1) { + ma_int32 x = src_s32[i]; + x = x >> 16; + dst_s16[i] = (ma_int16)x; } } else { - /* This is the more complicated case. Each of the output channels is accumulated with 0 or more input channels. */ + ma_uint64 i; + for (i = 0; i < count; i += 1) { + ma_int32 x = src_s32[i]; - /* Clear. */ - for (iChannelOut = 0; iChannelOut < pRouter->config.channelsOut; ++iChannelOut) { - ma_zero_memory_64(ppSamplesOut[iChannelOut], frameCount * sizeof(float)); + /* Dither. Don't overflow. */ + ma_int32 dither = ma_dither_s32(ditherMode, -0x8000, 0x7FFF); + if ((ma_int64)x + dither <= 0x7FFFFFFF) { + x = x + dither; + } else { + x = 0x7FFFFFFF; + } + + x = x >> 16; + dst_s16[i] = (ma_int16)x; } + } +} - /* Accumulate. */ - for (iChannelIn = 0; iChannelIn < pRouter->config.channelsIn; ++iChannelIn) { - for (iChannelOut = 0; iChannelOut < pRouter->config.channelsOut; ++iChannelOut) { - ma_uint64 iFrame = 0; -#if defined(MA_SUPPORT_NEON) - if (ma_channel_router__can_use_neon(pRouter, ppSamplesOut[iChannelOut], ppSamplesIn[iChannelIn])) { - float32x4_t weight = vmovq_n_f32(pRouter->config.weights[iChannelIn][iChannelOut]); - ma_uint64 frameCount4 = frameCount/4; - ma_uint64 iFrame4; - - for (iFrame4 = 0; iFrame4 < frameCount4; iFrame4 += 1) { - float32x4_t* pO = (float32x4_t*)ppSamplesOut[iChannelOut] + iFrame4; - float32x4_t* pI = (float32x4_t*)ppSamplesIn [iChannelIn ] + iFrame4; - *pO = vaddq_f32(*pO, vmulq_f32(*pI, weight)); - } - - iFrame += frameCount4*4; - } - else -#endif -#if defined(MA_SUPPORT_AVX512) - if (ma_channel_router__can_use_avx512(pRouter, ppSamplesOut[iChannelOut], ppSamplesIn[iChannelIn])) { - __m512 weight = _mm512_set1_ps(pRouter->config.weights[iChannelIn][iChannelOut]); - ma_uint64 frameCount16 = frameCount/16; - ma_uint64 iFrame16; - - for (iFrame16 = 0; iFrame16 < frameCount16; iFrame16 += 1) { - __m512* pO = (__m512*)ppSamplesOut[iChannelOut] + iFrame16; - __m512* pI = (__m512*)ppSamplesIn [iChannelIn ] + iFrame16; - *pO = _mm512_add_ps(*pO, _mm512_mul_ps(*pI, weight)); - } +static MA_INLINE void ma_pcm_s32_to_s16__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s32_to_s16__reference(dst, src, count, ditherMode); +} - iFrame += frameCount16*16; - } - else +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_s32_to_s16__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s32_to_s16__optimized(dst, src, count, ditherMode); +} #endif #if defined(MA_SUPPORT_AVX2) - if (ma_channel_router__can_use_avx2(pRouter, ppSamplesOut[iChannelOut], ppSamplesIn[iChannelIn])) { - __m256 weight = _mm256_set1_ps(pRouter->config.weights[iChannelIn][iChannelOut]); - ma_uint64 frameCount8 = frameCount/8; - ma_uint64 iFrame8; - - for (iFrame8 = 0; iFrame8 < frameCount8; iFrame8 += 1) { - __m256* pO = (__m256*)ppSamplesOut[iChannelOut] + iFrame8; - __m256* pI = (__m256*)ppSamplesIn [iChannelIn ] + iFrame8; - *pO = _mm256_add_ps(*pO, _mm256_mul_ps(*pI, weight)); - } +static MA_INLINE void ma_pcm_s32_to_s16__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s32_to_s16__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_s32_to_s16__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s32_to_s16__optimized(dst, src, count, ditherMode); +} +#endif - iFrame += frameCount8*8; - } - else +void ma_pcm_s32_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_s32_to_s16__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_s32_to_s16__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_s32_to_s16__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_s32_to_s16__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_s32_to_s16__optimized(dst, src, count, ditherMode); + } #endif -#if defined(MA_SUPPORT_SSE2) - if (ma_channel_router__can_use_sse2(pRouter, ppSamplesOut[iChannelOut], ppSamplesIn[iChannelIn])) { - __m128 weight = _mm_set1_ps(pRouter->config.weights[iChannelIn][iChannelOut]); - ma_uint64 frameCount4 = frameCount/4; - ma_uint64 iFrame4; - - for (iFrame4 = 0; iFrame4 < frameCount4; iFrame4 += 1) { - __m128* pO = (__m128*)ppSamplesOut[iChannelOut] + iFrame4; - __m128* pI = (__m128*)ppSamplesIn [iChannelIn ] + iFrame4; - *pO = _mm_add_ps(*pO, _mm_mul_ps(*pI, weight)); - } +} - iFrame += frameCount4*4; - } else -#endif - { /* Reference. */ - float weight0 = pRouter->config.weights[iChannelIn][iChannelOut]; - float weight1 = pRouter->config.weights[iChannelIn][iChannelOut]; - float weight2 = pRouter->config.weights[iChannelIn][iChannelOut]; - float weight3 = pRouter->config.weights[iChannelIn][iChannelOut]; - ma_uint64 frameCount4 = frameCount/4; - ma_uint64 iFrame4; - - for (iFrame4 = 0; iFrame4 < frameCount4; iFrame4 += 1) { - ppSamplesOut[iChannelOut][iFrame+0] += ppSamplesIn[iChannelIn][iFrame+0] * weight0; - ppSamplesOut[iChannelOut][iFrame+1] += ppSamplesIn[iChannelIn][iFrame+1] * weight1; - ppSamplesOut[iChannelOut][iFrame+2] += ppSamplesIn[iChannelIn][iFrame+2] * weight2; - ppSamplesOut[iChannelOut][iFrame+3] += ppSamplesIn[iChannelIn][iFrame+3] * weight3; - iFrame += 4; - } - } - /* Leftover. */ - for (; iFrame < frameCount; ++iFrame) { - ppSamplesOut[iChannelOut][iFrame] += ppSamplesIn[iChannelIn][iFrame] * pRouter->config.weights[iChannelIn][iChannelOut]; - } - } - } +static MA_INLINE void ma_pcm_s32_to_s24__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_uint8* dst_s24 = (ma_uint8*)dst; + const ma_int32* src_s32 = (const ma_int32*)src; + + ma_uint64 i; + for (i = 0; i < count; i += 1) { + ma_uint32 x = (ma_uint32)src_s32[i]; + dst_s24[i*3+0] = (ma_uint8)((x & 0x0000FF00) >> 8); + dst_s24[i*3+1] = (ma_uint8)((x & 0x00FF0000) >> 16); + dst_s24[i*3+2] = (ma_uint8)((x & 0xFF000000) >> 24); } + + (void)ditherMode; /* No dithering for s32 -> s24. */ +} + +static MA_INLINE void ma_pcm_s32_to_s24__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s32_to_s24__reference(dst, src, count, ditherMode); +} + +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_s32_to_s24__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s32_to_s24__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE void ma_pcm_s32_to_s24__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s32_to_s24__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_s32_to_s24__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s32_to_s24__optimized(dst, src, count, ditherMode); } +#endif -ma_uint64 ma_channel_router_read_deinterleaved(ma_channel_router* pRouter, ma_uint64 frameCount, void** ppSamplesOut, void* pUserData) +void ma_pcm_s32_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - if (pRouter == NULL || ppSamplesOut == NULL) { - return 0; - } +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_s32_to_s24__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_s32_to_s24__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_s32_to_s24__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_s32_to_s24__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_s32_to_s24__optimized(dst, src, count, ditherMode); + } +#endif +} - /* Fast path for a passthrough. */ - if (pRouter->isPassthrough) { - if (frameCount <= 0xFFFFFFFF) { - return (ma_uint32)pRouter->config.onReadDeinterleaved(pRouter, (ma_uint32)frameCount, ppSamplesOut, pUserData); - } else { - float* ppNextSamplesOut[MA_MAX_CHANNELS]; - ma_uint64 totalFramesRead; - ma_copy_memory(ppNextSamplesOut, ppSamplesOut, sizeof(float*) * pRouter->config.channelsOut); +void ma_pcm_s32_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + (void)ditherMode; - totalFramesRead = 0; - while (totalFramesRead < frameCount) { - ma_uint32 iChannel; - ma_uint32 framesJustRead; - ma_uint64 framesRemaining = (frameCount - totalFramesRead); - ma_uint64 framesToReadRightNow = framesRemaining; - if (framesToReadRightNow > 0xFFFFFFFF) { - framesToReadRightNow = 0xFFFFFFFF; - } + ma_copy_memory_64(dst, src, count * sizeof(ma_int32)); +} - framesJustRead = (ma_uint32)pRouter->config.onReadDeinterleaved(pRouter, (ma_uint32)framesToReadRightNow, (void**)ppNextSamplesOut, pUserData); - if (framesJustRead == 0) { - break; - } - totalFramesRead += framesJustRead; +static MA_INLINE void ma_pcm_s32_to_f32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + float* dst_f32 = (float*)dst; + const ma_int32* src_s32 = (const ma_int32*)src; - if (framesJustRead < framesToReadRightNow) { - break; - } + ma_uint64 i; + for (i = 0; i < count; i += 1) { + double x = src_s32[i]; - for (iChannel = 0; iChannel < pRouter->config.channelsOut; ++iChannel) { - ppNextSamplesOut[iChannel] += framesJustRead; - } - } +#if 0 + x = x + 2147483648.0; + x = x * 0.0000000004656612873077392578125; + x = x - 1; +#else + x = x / 2147483648.0; +#endif - return totalFramesRead; - } + dst_f32[i] = (float)x; } - /* Slower path for a non-passthrough. */ - { - float* ppNextSamplesOut[MA_MAX_CHANNELS]; - float* ppTemp[MA_MAX_CHANNELS]; - size_t maxBytesToReadPerFrameEachIteration; - size_t maxFramesToReadEachIteration; - ma_uint64 totalFramesRead; - MA_ALIGN(MA_SIMD_ALIGNMENT) float temp[MA_MAX_CHANNELS * 256]; - - ma_assert(sizeof(temp) <= 0xFFFFFFFF); - ma_copy_memory(ppNextSamplesOut, ppSamplesOut, sizeof(float*) * pRouter->config.channelsOut); - - - ma_split_buffer(temp, sizeof(temp), pRouter->config.channelsIn, MA_SIMD_ALIGNMENT, (void**)&ppTemp, &maxBytesToReadPerFrameEachIteration); + (void)ditherMode; /* No dithering for s32 -> f32. */ +} - maxFramesToReadEachIteration = maxBytesToReadPerFrameEachIteration/sizeof(float); +static MA_INLINE void ma_pcm_s32_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s32_to_f32__reference(dst, src, count, ditherMode); +} - totalFramesRead = 0; - while (totalFramesRead < frameCount) { - ma_uint32 iChannel; - ma_uint32 framesJustRead; - ma_uint64 framesRemaining = (frameCount - totalFramesRead); - ma_uint64 framesToReadRightNow = framesRemaining; - if (framesToReadRightNow > maxFramesToReadEachIteration) { - framesToReadRightNow = maxFramesToReadEachIteration; - } +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_s32_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE void ma_pcm_s32_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_s32_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode); +} +#endif - framesJustRead = pRouter->config.onReadDeinterleaved(pRouter, (ma_uint32)framesToReadRightNow, (void**)ppTemp, pUserData); - if (framesJustRead == 0) { - break; - } +void ma_pcm_s32_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_s32_to_f32__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_s32_to_f32__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_s32_to_f32__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_s32_to_f32__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode); + } +#endif +} - ma_channel_router__do_routing(pRouter, framesJustRead, (float**)ppNextSamplesOut, (const float**)ppTemp); /* <-- Real work is done here. */ - totalFramesRead += framesJustRead; - if (totalFramesRead < frameCount) { - for (iChannel = 0; iChannel < pRouter->config.channelsIn; iChannel += 1) { - ppNextSamplesOut[iChannel] += framesJustRead; - } - } +static MA_INLINE void ma_pcm_interleave_s32__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) +{ + ma_int32* dst_s32 = (ma_int32*)dst; + const ma_int32** src_s32 = (const ma_int32**)src; - if (framesJustRead < framesToReadRightNow) { - break; - } + ma_uint64 iFrame; + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + ma_uint32 iChannel; + for (iChannel = 0; iChannel < channels; iChannel += 1) { + dst_s32[iFrame*channels + iChannel] = src_s32[iChannel][iFrame]; } - - return totalFramesRead; } } -ma_channel_router_config ma_channel_router_config_init(ma_uint32 channelsIn, const ma_channel channelMapIn[MA_MAX_CHANNELS], ma_uint32 channelsOut, const ma_channel channelMapOut[MA_MAX_CHANNELS], ma_channel_mix_mode mixingMode, ma_channel_router_read_deinterleaved_proc onRead, void* pUserData) +static MA_INLINE void ma_pcm_interleave_s32__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) { - ma_channel_router_config config; - ma_uint32 iChannel; - - ma_zero_object(&config); - - config.channelsIn = channelsIn; - for (iChannel = 0; iChannel < channelsIn; ++iChannel) { - config.channelMapIn[iChannel] = channelMapIn[iChannel]; - } - - config.channelsOut = channelsOut; - for (iChannel = 0; iChannel < channelsOut; ++iChannel) { - config.channelMapOut[iChannel] = channelMapOut[iChannel]; - } - - config.mixingMode = mixingMode; - config.onReadDeinterleaved = onRead; - config.pUserData = pUserData; - - return config; + ma_pcm_interleave_s32__reference(dst, src, frameCount, channels); } +void ma_pcm_interleave_s32(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_interleave_s32__reference(dst, src, frameCount, channels); +#else + ma_pcm_interleave_s32__optimized(dst, src, frameCount, channels); +#endif +} -/************************************************************************************************************************************************************** +static MA_INLINE void ma_pcm_deinterleave_s32__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) +{ + ma_int32** dst_s32 = (ma_int32**)dst; + const ma_int32* src_s32 = (const ma_int32*)src; -SRC + ma_uint64 iFrame; + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + ma_uint32 iChannel; + for (iChannel = 0; iChannel < channels; iChannel += 1) { + dst_s32[iChannel][iFrame] = src_s32[iFrame*channels + iChannel]; + } + } +} -**************************************************************************************************************************************************************/ -#define ma_floorf(x) ((float)floor((double)(x))) -#define ma_sinf(x) ((float)sin((double)(x))) -#define ma_cosf(x) ((float)cos((double)(x))) +static MA_INLINE void ma_pcm_deinterleave_s32__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) +{ + ma_pcm_deinterleave_s32__reference(dst, src, frameCount, channels); +} -static MA_INLINE double ma_sinc(double x) +void ma_pcm_deinterleave_s32(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) { - if (x != 0) { - return sin(MA_PI_D*x) / (MA_PI_D*x); - } else { - return 1; - } +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_deinterleave_s32__reference(dst, src, frameCount, channels); +#else + ma_pcm_deinterleave_s32__optimized(dst, src, frameCount, channels); +#endif } -#define ma_sincf(x) ((float)ma_sinc((double)(x))) +/* f32 */ +static MA_INLINE void ma_pcm_f32_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_uint64 i; -ma_uint64 ma_src_read_deinterleaved__passthrough(ma_src* pSRC, ma_uint64 frameCount, void** ppSamplesOut, void* pUserData); -ma_uint64 ma_src_read_deinterleaved__linear(ma_src* pSRC, ma_uint64 frameCount, void** ppSamplesOut, void* pUserData); -ma_uint64 ma_src_read_deinterleaved__sinc(ma_src* pSRC, ma_uint64 frameCount, void** ppSamplesOut, void* pUserData); + ma_uint8* dst_u8 = (ma_uint8*)dst; + const float* src_f32 = (const float*)src; -void ma_src__build_sinc_table__sinc(ma_src* pSRC) -{ - ma_uint32 i; + float ditherMin = 0; + float ditherMax = 0; + if (ditherMode != ma_dither_mode_none) { + ditherMin = 1.0f / -128; + ditherMax = 1.0f / 127; + } - ma_assert(pSRC != NULL); + for (i = 0; i < count; i += 1) { + float x = src_f32[i]; + x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax); + x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */ + x = x + 1; /* -1..1 to 0..2 */ + x = x * 127.5f; /* 0..2 to 0..255 */ - pSRC->sinc.table[0] = 1.0f; - for (i = 1; i < ma_countof(pSRC->sinc.table); i += 1) { - double x = i*MA_PI_D / MA_SRC_SINC_LOOKUP_TABLE_RESOLUTION; - pSRC->sinc.table[i] = (float)(sin(x)/x); + dst_u8[i] = (ma_uint8)x; } } -void ma_src__build_sinc_table__rectangular(ma_src* pSRC) +static MA_INLINE void ma_pcm_f32_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - /* This is the same as the base sinc table. */ - ma_src__build_sinc_table__sinc(pSRC); + ma_pcm_f32_to_u8__reference(dst, src, count, ditherMode); } -void ma_src__build_sinc_table__hann(ma_src* pSRC) +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_f32_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - ma_uint32 i; - - ma_src__build_sinc_table__sinc(pSRC); - - for (i = 0; i < ma_countof(pSRC->sinc.table); i += 1) { - double x = pSRC->sinc.table[i]; - double N = MA_SRC_SINC_MAX_WINDOW_WIDTH*2; - double n = ((double)(i) / MA_SRC_SINC_LOOKUP_TABLE_RESOLUTION) + MA_SRC_SINC_MAX_WINDOW_WIDTH; - double w = 0.5 * (1 - cos((2*MA_PI_D*n) / (N))); + ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE void ma_pcm_f32_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_f32_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode); +} +#endif - pSRC->sinc.table[i] = (float)(x * w); - } +void ma_pcm_f32_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_f32_to_u8__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_f32_to_u8__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_f32_to_u8__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_f32_to_u8__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode); + } +#endif } -ma_result ma_src_init(const ma_src_config* pConfig, ma_src* pSRC) +#ifdef MA_USE_REFERENCE_CONVERSION_APIS +static MA_INLINE void ma_pcm_f32_to_s16__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - if (pSRC == NULL) { - return MA_INVALID_ARGS; - } + ma_uint64 i; - ma_zero_object(pSRC); + ma_int16* dst_s16 = (ma_int16*)dst; + const float* src_f32 = (const float*)src; - if (pConfig == NULL || pConfig->onReadDeinterleaved == NULL) { - return MA_INVALID_ARGS; - } - if (pConfig->channels == 0 || pConfig->channels > MA_MAX_CHANNELS) { - return MA_INVALID_ARGS; + float ditherMin = 0; + float ditherMax = 0; + if (ditherMode != ma_dither_mode_none) { + ditherMin = 1.0f / -32768; + ditherMax = 1.0f / 32767; } - pSRC->config = *pConfig; - - /* SIMD */ - pSRC->useSSE2 = ma_has_sse2() && !pConfig->noSSE2; - pSRC->useAVX2 = ma_has_avx2() && !pConfig->noAVX2; - pSRC->useAVX512 = ma_has_avx512f() && !pConfig->noAVX512; - pSRC->useNEON = ma_has_neon() && !pConfig->noNEON; + for (i = 0; i < count; i += 1) { + float x = src_f32[i]; + x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax); + x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */ - if (pSRC->config.algorithm == ma_src_algorithm_sinc) { - /* Make sure the window width within bounds. */ - if (pSRC->config.sinc.windowWidth == 0) { - pSRC->config.sinc.windowWidth = MA_SRC_SINC_DEFAULT_WINDOW_WIDTH; - } - if (pSRC->config.sinc.windowWidth < MA_SRC_SINC_MIN_WINDOW_WIDTH) { - pSRC->config.sinc.windowWidth = MA_SRC_SINC_MIN_WINDOW_WIDTH; - } - if (pSRC->config.sinc.windowWidth > MA_SRC_SINC_MAX_WINDOW_WIDTH) { - pSRC->config.sinc.windowWidth = MA_SRC_SINC_MAX_WINDOW_WIDTH; - } +#if 0 + /* The accurate way. */ + x = x + 1; /* -1..1 to 0..2 */ + x = x * 32767.5f; /* 0..2 to 0..65535 */ + x = x - 32768.0f; /* 0...65535 to -32768..32767 */ +#else + /* The fast way. */ + x = x * 32767.0f; /* -1..1 to -32767..32767 */ +#endif - /* Set up the lookup table. */ - switch (pSRC->config.sinc.windowFunction) { - case ma_src_sinc_window_function_hann: ma_src__build_sinc_table__hann(pSRC); break; - case ma_src_sinc_window_function_rectangular: ma_src__build_sinc_table__rectangular(pSRC); break; - default: return MA_INVALID_ARGS; /* <-- Hitting this means the window function is unknown to miniaudio. */ - } + dst_s16[i] = (ma_int16)x; } - - return MA_SUCCESS; } - -ma_result ma_src_set_sample_rate(ma_src* pSRC, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut) +#else +static MA_INLINE void ma_pcm_f32_to_s16__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - if (pSRC == NULL) { - return MA_INVALID_ARGS; - } + ma_uint64 i; + ma_uint64 i4; + ma_uint64 count4; - /* Must have a sample rate of > 0. */ - if (sampleRateIn == 0 || sampleRateOut == 0) { - return MA_INVALID_ARGS; + ma_int16* dst_s16 = (ma_int16*)dst; + const float* src_f32 = (const float*)src; + + float ditherMin = 0; + float ditherMax = 0; + if (ditherMode != ma_dither_mode_none) { + ditherMin = 1.0f / -32768; + ditherMax = 1.0f / 32767; } - ma_atomic_exchange_32(&pSRC->config.sampleRateIn, sampleRateIn); - ma_atomic_exchange_32(&pSRC->config.sampleRateOut, sampleRateOut); + /* Unrolled. */ + i = 0; + count4 = count >> 2; + for (i4 = 0; i4 < count4; i4 += 1) { + float d0 = ma_dither_f32(ditherMode, ditherMin, ditherMax); + float d1 = ma_dither_f32(ditherMode, ditherMin, ditherMax); + float d2 = ma_dither_f32(ditherMode, ditherMin, ditherMax); + float d3 = ma_dither_f32(ditherMode, ditherMin, ditherMax); + + float x0 = src_f32[i+0]; + float x1 = src_f32[i+1]; + float x2 = src_f32[i+2]; + float x3 = src_f32[i+3]; + + x0 = x0 + d0; + x1 = x1 + d1; + x2 = x2 + d2; + x3 = x3 + d3; + + x0 = ((x0 < -1) ? -1 : ((x0 > 1) ? 1 : x0)); + x1 = ((x1 < -1) ? -1 : ((x1 > 1) ? 1 : x1)); + x2 = ((x2 < -1) ? -1 : ((x2 > 1) ? 1 : x2)); + x3 = ((x3 < -1) ? -1 : ((x3 > 1) ? 1 : x3)); - return MA_SUCCESS; -} + x0 = x0 * 32767.0f; + x1 = x1 * 32767.0f; + x2 = x2 * 32767.0f; + x3 = x3 * 32767.0f; -ma_uint64 ma_src_read_deinterleaved(ma_src* pSRC, ma_uint64 frameCount, void** ppSamplesOut, void* pUserData) -{ - ma_src_algorithm algorithm; + dst_s16[i+0] = (ma_int16)x0; + dst_s16[i+1] = (ma_int16)x1; + dst_s16[i+2] = (ma_int16)x2; + dst_s16[i+3] = (ma_int16)x3; - if (pSRC == NULL || frameCount == 0 || ppSamplesOut == NULL) { - return 0; + i += 4; } - algorithm = pSRC->config.algorithm; + /* Leftover. */ + for (; i < count; i += 1) { + float x = src_f32[i]; + x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax); + x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */ + x = x * 32767.0f; /* -1..1 to -32767..32767 */ - /* Can use a function pointer for this. */ - switch (algorithm) { - case ma_src_algorithm_none: return ma_src_read_deinterleaved__passthrough(pSRC, frameCount, ppSamplesOut, pUserData); - case ma_src_algorithm_linear: return ma_src_read_deinterleaved__linear( pSRC, frameCount, ppSamplesOut, pUserData); - case ma_src_algorithm_sinc: return ma_src_read_deinterleaved__sinc( pSRC, frameCount, ppSamplesOut, pUserData); - default: break; + dst_s16[i] = (ma_int16)x; } - - /* Should never get here. */ - return 0; } +#endif -ma_uint64 ma_src_read_deinterleaved__passthrough(ma_src* pSRC, ma_uint64 frameCount, void** ppSamplesOut, void* pUserData) +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_f32_to_s16__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - if (frameCount <= 0xFFFFFFFF) { - return pSRC->config.onReadDeinterleaved(pSRC, (ma_uint32)frameCount, ppSamplesOut, pUserData); - } else { - ma_uint32 iChannel; - ma_uint64 totalFramesRead; - float* ppNextSamplesOut[MA_MAX_CHANNELS]; - - for (iChannel = 0; iChannel < pSRC->config.channels; ++iChannel) { - ppNextSamplesOut[iChannel] = (float*)ppSamplesOut[iChannel]; - } - - totalFramesRead = 0; - while (totalFramesRead < frameCount) { - ma_uint32 framesJustRead; - ma_uint64 framesRemaining = frameCount - totalFramesRead; - ma_uint64 framesToReadRightNow = framesRemaining; - if (framesToReadRightNow > 0xFFFFFFFF) { - framesToReadRightNow = 0xFFFFFFFF; - } - - framesJustRead = (ma_uint32)pSRC->config.onReadDeinterleaved(pSRC, (ma_uint32)framesToReadRightNow, (void**)ppNextSamplesOut, pUserData); - if (framesJustRead == 0) { - break; - } + ma_uint64 i; + ma_uint64 i8; + ma_uint64 count8; + ma_int16* dst_s16; + const float* src_f32; + float ditherMin; + float ditherMax; - totalFramesRead += framesJustRead; - for (iChannel = 0; iChannel < pSRC->config.channels; ++iChannel) { - ppNextSamplesOut[iChannel] += framesJustRead; - } + /* Both the input and output buffers need to be aligned to 16 bytes. */ + if ((((ma_uintptr)dst & 15) != 0) || (((ma_uintptr)src & 15) != 0)) { + ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode); + return; + } - if (framesJustRead < framesToReadRightNow) { - break; - } - } + dst_s16 = (ma_int16*)dst; + src_f32 = (const float*)src; - return totalFramesRead; + ditherMin = 0; + ditherMax = 0; + if (ditherMode != ma_dither_mode_none) { + ditherMin = 1.0f / -32768; + ditherMax = 1.0f / 32767; } -} - -ma_uint64 ma_src_read_deinterleaved__linear(ma_src* pSRC, ma_uint64 frameCount, void** ppSamplesOut, void* pUserData) -{ - float* ppNextSamplesOut[MA_MAX_CHANNELS]; - float factor; - ma_uint32 maxFrameCountPerChunkIn; - ma_uint64 totalFramesRead; - ma_assert(pSRC != NULL); - ma_assert(frameCount > 0); - ma_assert(ppSamplesOut != NULL); - - ma_copy_memory(ppNextSamplesOut, ppSamplesOut, sizeof(void*) * pSRC->config.channels); + i = 0; - factor = (float)pSRC->config.sampleRateIn / pSRC->config.sampleRateOut; - maxFrameCountPerChunkIn = ma_countof(pSRC->linear.input[0]); + /* SSE2. SSE allows us to output 8 s16's at a time which means our loop is unrolled 8 times. */ + count8 = count >> 3; + for (i8 = 0; i8 < count8; i8 += 1) { + __m128 d0; + __m128 d1; + __m128 x0; + __m128 x1; - totalFramesRead = 0; - while (totalFramesRead < frameCount) { - ma_uint32 iChannel; - float tBeg; - float tEnd; - float tAvailable; - float tNext; - float* ppSamplesFromClient[MA_MAX_CHANNELS]; - ma_uint32 iNextFrame; - ma_uint32 maxOutputFramesToRead; - ma_uint32 maxOutputFramesToRead4; - ma_uint32 framesToReadFromClient; - ma_uint32 framesReadFromClient; - ma_uint64 framesRemaining = frameCount - totalFramesRead; - ma_uint64 framesToRead = framesRemaining; - if (framesToRead > 16384) { - framesToRead = 16384; /* <-- Keep this small because we're using 32-bit floats for calculating sample positions and I don't want to run out of precision with huge sample counts. */ + if (ditherMode == ma_dither_mode_none) { + d0 = _mm_set1_ps(0); + d1 = _mm_set1_ps(0); + } else if (ditherMode == ma_dither_mode_rectangle) { + d0 = _mm_set_ps( + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax) + ); + d1 = _mm_set_ps( + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax) + ); + } else { + d0 = _mm_set_ps( + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax) + ); + d1 = _mm_set_ps( + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax) + ); } + x0 = *((__m128*)(src_f32 + i) + 0); + x1 = *((__m128*)(src_f32 + i) + 1); - /* Read Input Data */ - tBeg = pSRC->linear.timeIn; - tEnd = tBeg + ((ma_int64)framesToRead*factor); /* Cast to int64 required for VC6. */ + x0 = _mm_add_ps(x0, d0); + x1 = _mm_add_ps(x1, d1); - framesToReadFromClient = (ma_uint32)(tEnd) + 1 + 1; /* +1 to make tEnd 1-based and +1 because we always need to an extra sample for interpolation. */ - if (framesToReadFromClient >= maxFrameCountPerChunkIn) { - framesToReadFromClient = maxFrameCountPerChunkIn; - } + x0 = _mm_mul_ps(x0, _mm_set1_ps(32767.0f)); + x1 = _mm_mul_ps(x1, _mm_set1_ps(32767.0f)); - for (iChannel = 0; iChannel < pSRC->config.channels; ++iChannel) { - ppSamplesFromClient[iChannel] = pSRC->linear.input[iChannel] + pSRC->linear.leftoverFrames; - } + _mm_stream_si128(((__m128i*)(dst_s16 + i)), _mm_packs_epi32(_mm_cvttps_epi32(x0), _mm_cvttps_epi32(x1))); - framesReadFromClient = 0; - if (framesToReadFromClient > pSRC->linear.leftoverFrames) { - framesReadFromClient = (ma_uint32)pSRC->config.onReadDeinterleaved(pSRC, (ma_uint32)framesToReadFromClient - pSRC->linear.leftoverFrames, (void**)ppSamplesFromClient, pUserData); - } - - framesReadFromClient += pSRC->linear.leftoverFrames; /* <-- You can sort of think of it as though we've re-read the leftover samples from the client. */ - if (framesReadFromClient < 2) { - break; - } - - for (iChannel = 0; iChannel < pSRC->config.channels; ++iChannel) { - ppSamplesFromClient[iChannel] = pSRC->linear.input[iChannel]; - } - - - /* Write Output Data */ - - /* - At this point we have a bunch of frames that the client has given to us for processing. From this we can determine the maximum number of output frames - that can be processed from this input. We want to output as many samples as possible from our input data. - */ - tAvailable = framesReadFromClient - tBeg - 1; /* Subtract 1 because the last input sample is needed for interpolation and cannot be included in the output sample count calculation. */ - - maxOutputFramesToRead = (ma_uint32)(tAvailable / factor); - if (maxOutputFramesToRead == 0) { - maxOutputFramesToRead = 1; - } - if (maxOutputFramesToRead > framesToRead) { - maxOutputFramesToRead = (ma_uint32)framesToRead; - } + i += 8; + } - /* Output frames are always read in groups of 4 because I'm planning on using this as a reference for some SIMD-y stuff later. */ - maxOutputFramesToRead4 = maxOutputFramesToRead/4; - for (iChannel = 0; iChannel < pSRC->config.channels; ++iChannel) { - ma_uint32 iFrameOut; - float t0 = pSRC->linear.timeIn + factor*0; - float t1 = pSRC->linear.timeIn + factor*1; - float t2 = pSRC->linear.timeIn + factor*2; - float t3 = pSRC->linear.timeIn + factor*3; - float t; - - for (iFrameOut = 0; iFrameOut < maxOutputFramesToRead4; iFrameOut += 1) { - float iPrevSample0 = (float)floor(t0); - float iPrevSample1 = (float)floor(t1); - float iPrevSample2 = (float)floor(t2); - float iPrevSample3 = (float)floor(t3); - - float iNextSample0 = iPrevSample0 + 1; - float iNextSample1 = iPrevSample1 + 1; - float iNextSample2 = iPrevSample2 + 1; - float iNextSample3 = iPrevSample3 + 1; - - float alpha0 = t0 - iPrevSample0; - float alpha1 = t1 - iPrevSample1; - float alpha2 = t2 - iPrevSample2; - float alpha3 = t3 - iPrevSample3; - - float prevSample0 = ppSamplesFromClient[iChannel][(ma_uint32)iPrevSample0]; - float prevSample1 = ppSamplesFromClient[iChannel][(ma_uint32)iPrevSample1]; - float prevSample2 = ppSamplesFromClient[iChannel][(ma_uint32)iPrevSample2]; - float prevSample3 = ppSamplesFromClient[iChannel][(ma_uint32)iPrevSample3]; - - float nextSample0 = ppSamplesFromClient[iChannel][(ma_uint32)iNextSample0]; - float nextSample1 = ppSamplesFromClient[iChannel][(ma_uint32)iNextSample1]; - float nextSample2 = ppSamplesFromClient[iChannel][(ma_uint32)iNextSample2]; - float nextSample3 = ppSamplesFromClient[iChannel][(ma_uint32)iNextSample3]; - ppNextSamplesOut[iChannel][iFrameOut*4 + 0] = ma_mix_f32_fast(prevSample0, nextSample0, alpha0); - ppNextSamplesOut[iChannel][iFrameOut*4 + 1] = ma_mix_f32_fast(prevSample1, nextSample1, alpha1); - ppNextSamplesOut[iChannel][iFrameOut*4 + 2] = ma_mix_f32_fast(prevSample2, nextSample2, alpha2); - ppNextSamplesOut[iChannel][iFrameOut*4 + 3] = ma_mix_f32_fast(prevSample3, nextSample3, alpha3); + /* Leftover. */ + for (; i < count; i += 1) { + float x = src_f32[i]; + x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax); + x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */ + x = x * 32767.0f; /* -1..1 to -32767..32767 */ - t0 += factor*4; - t1 += factor*4; - t2 += factor*4; - t3 += factor*4; - } + dst_s16[i] = (ma_int16)x; + } +} +#endif +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE void ma_pcm_f32_to_s16__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_uint64 i; + ma_uint64 i16; + ma_uint64 count16; + ma_int16* dst_s16; + const float* src_f32; + float ditherMin; + float ditherMax; - t = pSRC->linear.timeIn + (factor*maxOutputFramesToRead4*4); - for (iFrameOut = (maxOutputFramesToRead4*4); iFrameOut < maxOutputFramesToRead; iFrameOut += 1) { - float iPrevSample = (float)floor(t); - float iNextSample = iPrevSample + 1; - float alpha = t - iPrevSample; - float prevSample; - float nextSample; + /* Both the input and output buffers need to be aligned to 32 bytes. */ + if ((((ma_uintptr)dst & 31) != 0) || (((ma_uintptr)src & 31) != 0)) { + ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode); + return; + } - ma_assert(iPrevSample < ma_countof(pSRC->linear.input[iChannel])); - ma_assert(iNextSample < ma_countof(pSRC->linear.input[iChannel])); + dst_s16 = (ma_int16*)dst; + src_f32 = (const float*)src; - prevSample = ppSamplesFromClient[iChannel][(ma_uint32)iPrevSample]; - nextSample = ppSamplesFromClient[iChannel][(ma_uint32)iNextSample]; + ditherMin = 0; + ditherMax = 0; + if (ditherMode != ma_dither_mode_none) { + ditherMin = 1.0f / -32768; + ditherMax = 1.0f / 32767; + } - ppNextSamplesOut[iChannel][iFrameOut] = ma_mix_f32_fast(prevSample, nextSample, alpha); + i = 0; - t += factor; - } + /* AVX2. AVX2 allows us to output 16 s16's at a time which means our loop is unrolled 16 times. */ + count16 = count >> 4; + for (i16 = 0; i16 < count16; i16 += 1) { + __m256 d0; + __m256 d1; + __m256 x0; + __m256 x1; + __m256i i0; + __m256i i1; + __m256i p0; + __m256i p1; + __m256i r; - ppNextSamplesOut[iChannel] += maxOutputFramesToRead; + if (ditherMode == ma_dither_mode_none) { + d0 = _mm256_set1_ps(0); + d1 = _mm256_set1_ps(0); + } else if (ditherMode == ma_dither_mode_rectangle) { + d0 = _mm256_set_ps( + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax) + ); + d1 = _mm256_set_ps( + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax), + ma_dither_f32_rectangle(ditherMin, ditherMax) + ); + } else { + d0 = _mm256_set_ps( + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax) + ); + d1 = _mm256_set_ps( + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax), + ma_dither_f32_triangle(ditherMin, ditherMax) + ); } - totalFramesRead += maxOutputFramesToRead; - + x0 = *((__m256*)(src_f32 + i) + 0); + x1 = *((__m256*)(src_f32 + i) + 1); - /* Residual */ - tNext = pSRC->linear.timeIn + (maxOutputFramesToRead*factor); + x0 = _mm256_add_ps(x0, d0); + x1 = _mm256_add_ps(x1, d1); - pSRC->linear.timeIn = tNext; - ma_assert(tNext <= framesReadFromClient+1); + x0 = _mm256_mul_ps(x0, _mm256_set1_ps(32767.0f)); + x1 = _mm256_mul_ps(x1, _mm256_set1_ps(32767.0f)); - iNextFrame = (ma_uint32)floor(tNext); - pSRC->linear.leftoverFrames = framesReadFromClient - iNextFrame; - pSRC->linear.timeIn = tNext - iNextFrame; + /* Computing the final result is a little more complicated for AVX2 than SSE2. */ + i0 = _mm256_cvttps_epi32(x0); + i1 = _mm256_cvttps_epi32(x1); + p0 = _mm256_permute2x128_si256(i0, i1, 0 | 32); + p1 = _mm256_permute2x128_si256(i0, i1, 1 | 48); + r = _mm256_packs_epi32(p0, p1); - for (iChannel = 0; iChannel < pSRC->config.channels; ++iChannel) { - ma_uint32 iFrame; - for (iFrame = 0; iFrame < pSRC->linear.leftoverFrames; ++iFrame) { - float sample = ppSamplesFromClient[iChannel][framesReadFromClient-pSRC->linear.leftoverFrames + iFrame]; - ppSamplesFromClient[iChannel][iFrame] = sample; - } - } + _mm256_stream_si256(((__m256i*)(dst_s16 + i)), r); - - /* Exit the loop if we've found everything from the client. */ - if (framesReadFromClient < framesToReadFromClient) { - break; - } + i += 16; } - return totalFramesRead; -} - - -ma_src_config ma_src_config_init_new() -{ - ma_src_config config; - ma_zero_object(&config); - - return config; -} -ma_src_config ma_src_config_init(ma_uint32 sampleRateIn, ma_uint32 sampleRateOut, ma_uint32 channels, ma_src_read_deinterleaved_proc onReadDeinterleaved, void* pUserData) -{ - ma_src_config config = ma_src_config_init_new(); - config.sampleRateIn = sampleRateIn; - config.sampleRateOut = sampleRateOut; - config.channels = channels; - config.onReadDeinterleaved = onReadDeinterleaved; - config.pUserData = pUserData; + /* Leftover. */ + for (; i < count; i += 1) { + float x = src_f32[i]; + x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax); + x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */ + x = x * 32767.0f; /* -1..1 to -32767..32767 */ - return config; + dst_s16[i] = (ma_int16)x; + } } +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_f32_to_s16__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_uint64 i; + ma_uint64 i8; + ma_uint64 count8; + ma_int16* dst_s16; + const float* src_f32; + float ditherMin; + float ditherMax; + if (!ma_has_neon()) { + return ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode); + } -/************************************************************************************************************************************************************** + /* Both the input and output buffers need to be aligned to 16 bytes. */ + if ((((ma_uintptr)dst & 15) != 0) || (((ma_uintptr)src & 15) != 0)) { + ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode); + return; + } -Sinc Sample Rate Conversion -=========================== + dst_s16 = (ma_int16*)dst; + src_f32 = (const float*)src; -The sinc SRC algorithm uses a windowed sinc to perform interpolation of samples. Currently, miniaudio's implementation supports rectangular and Hann window -methods. + ditherMin = 0; + ditherMax = 0; + if (ditherMode != ma_dither_mode_none) { + ditherMin = 1.0f / -32768; + ditherMax = 1.0f / 32767; + } -Whenever an output sample is being computed, it looks at a sub-section of the input samples. I've called this sub-section in the code below the "window", -which I realize is a bit ambigous with the mathematical "window", but it works for me when I need to conceptualize things in my head. The window is made up -of two halves. The first half contains past input samples (initialized to zero), and the second half contains future input samples. As time moves forward -and input samples are consumed, the window moves forward. The larger the window, the better the quality at the expense of slower processing. The window is -limited the range [MA_SRC_SINC_MIN_WINDOW_WIDTH, MA_SRC_SINC_MAX_WINDOW_WIDTH] and defaults to MA_SRC_SINC_DEFAULT_WINDOW_WIDTH. + i = 0; -Input samples are cached for efficiency (to prevent frequently requesting tiny numbers of samples from the client). When the window gets to the end of the -cache, it's moved back to the start, and more samples are read from the client. If the client has no more data to give, the cache is filled with zeros and -the last of the input samples will be consumed. Once the last of the input samples have been consumed, no more samples will be output. + /* NEON. NEON allows us to output 8 s16's at a time which means our loop is unrolled 8 times. */ + count8 = count >> 3; + for (i8 = 0; i8 < count8; i8 += 1) { + float32x4_t d0; + float32x4_t d1; + float32x4_t x0; + float32x4_t x1; + int32x4_t i0; + int32x4_t i1; + if (ditherMode == ma_dither_mode_none) { + d0 = vmovq_n_f32(0); + d1 = vmovq_n_f32(0); + } else if (ditherMode == ma_dither_mode_rectangle) { + float d0v[4]; + d0v[0] = ma_dither_f32_rectangle(ditherMin, ditherMax); + d0v[1] = ma_dither_f32_rectangle(ditherMin, ditherMax); + d0v[2] = ma_dither_f32_rectangle(ditherMin, ditherMax); + d0v[3] = ma_dither_f32_rectangle(ditherMin, ditherMax); + d0 = vld1q_f32(d0v); -When reading output samples, we always first read whatever is already in the input cache. Only when the cache has been fully consumed do we read more data -from the client. + float d1v[4]; + d1v[0] = ma_dither_f32_rectangle(ditherMin, ditherMax); + d1v[1] = ma_dither_f32_rectangle(ditherMin, ditherMax); + d1v[2] = ma_dither_f32_rectangle(ditherMin, ditherMax); + d1v[3] = ma_dither_f32_rectangle(ditherMin, ditherMax); + d1 = vld1q_f32(d1v); + } else { + float d0v[4]; + d0v[0] = ma_dither_f32_triangle(ditherMin, ditherMax); + d0v[1] = ma_dither_f32_triangle(ditherMin, ditherMax); + d0v[2] = ma_dither_f32_triangle(ditherMin, ditherMax); + d0v[3] = ma_dither_f32_triangle(ditherMin, ditherMax); + d0 = vld1q_f32(d0v); -To access samples in the input buffer you do so relative to the window. When the window itself is at position 0, the first item in the buffer is accessed -with "windowPos + windowWidth". Generally, to access any sample relative to the window you do "windowPos + windowWidth + sampleIndexRelativeToWindow". + float d1v[4]; + d1v[0] = ma_dither_f32_triangle(ditherMin, ditherMax); + d1v[1] = ma_dither_f32_triangle(ditherMin, ditherMax); + d1v[2] = ma_dither_f32_triangle(ditherMin, ditherMax); + d1v[3] = ma_dither_f32_triangle(ditherMin, ditherMax); + d1 = vld1q_f32(d1v); + } -**************************************************************************************************************************************************************/ + x0 = *((float32x4_t*)(src_f32 + i) + 0); + x1 = *((float32x4_t*)(src_f32 + i) + 1); -/* Comment this to disable interpolation of table lookups. Less accurate, but faster. */ -#define MA_USE_SINC_TABLE_INTERPOLATION + x0 = vaddq_f32(x0, d0); + x1 = vaddq_f32(x1, d1); -/* Retrieves a sample from the input buffer's window. Values >= 0 retrieve future samples. Negative values return past samples. */ -static MA_INLINE float ma_src_sinc__get_input_sample_from_window(const ma_src* pSRC, ma_uint32 channel, ma_uint32 windowPosInSamples, ma_int32 sampleIndex) -{ - ma_assert(pSRC != NULL); - ma_assert(channel < pSRC->config.channels); - ma_assert(sampleIndex >= -(ma_int32)pSRC->config.sinc.windowWidth); - ma_assert(sampleIndex < (ma_int32)pSRC->config.sinc.windowWidth); + x0 = vmulq_n_f32(x0, 32767.0f); + x1 = vmulq_n_f32(x1, 32767.0f); - /* The window should always be contained within the input cache. */ - ma_assert(windowPosInSamples < ma_countof(pSRC->sinc.input[0]) - pSRC->config.sinc.windowWidth); - - return pSRC->sinc.input[channel][windowPosInSamples + pSRC->config.sinc.windowWidth + sampleIndex]; -} + i0 = vcvtq_s32_f32(x0); + i1 = vcvtq_s32_f32(x1); + *((int16x8_t*)(dst_s16 + i)) = vcombine_s16(vqmovn_s32(i0), vqmovn_s32(i1)); -static MA_INLINE float ma_src_sinc__interpolation_factor(const ma_src* pSRC, float x) -{ - float xabs; - ma_int32 ixabs; + i += 8; + } - ma_assert(pSRC != NULL); - xabs = (float)fabs(x); - xabs = xabs * MA_SRC_SINC_LOOKUP_TABLE_RESOLUTION; - ixabs = (ma_int32)xabs; + /* Leftover. */ + for (; i < count; i += 1) { + float x = src_f32[i]; + x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax); + x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */ + x = x * 32767.0f; /* -1..1 to -32767..32767 */ -#if defined(MA_USE_SINC_TABLE_INTERPOLATION) - { - float a = xabs - ixabs; - return ma_mix_f32_fast(pSRC->sinc.table[ixabs], pSRC->sinc.table[ixabs+1], a); + dst_s16[i] = (ma_int16)x; } -#else - return pSRC->sinc.table[ixabs]; -#endif } +#endif -#if defined(MA_SUPPORT_SSE2) -static MA_INLINE __m128 ma_fabsf_sse2(__m128 x) +void ma_pcm_f32_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - return _mm_and_ps(_mm_castsi128_ps(_mm_set1_epi32(0x7FFFFFFF)), x); +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_f32_to_s16__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_f32_to_s16__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_f32_to_s16__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_f32_to_s16__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode); + } +#endif } -static MA_INLINE __m128 ma_truncf_sse2(__m128 x) -{ - return _mm_cvtepi32_ps(_mm_cvttps_epi32(x)); -} -static MA_INLINE __m128 ma_src_sinc__interpolation_factor__sse2(const ma_src* pSRC, __m128 x) +static MA_INLINE void ma_pcm_f32_to_s24__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - __m128 resolution128; - __m128 xabs; - __m128i ixabs; - __m128 lo; - __m128 hi; - __m128 a; - __m128 r; - int* ixabsv; - - resolution128 = _mm_set1_ps(MA_SRC_SINC_LOOKUP_TABLE_RESOLUTION); - xabs = ma_fabsf_sse2(x); - xabs = _mm_mul_ps(xabs, resolution128); - ixabs = _mm_cvttps_epi32(xabs); + ma_uint8* dst_s24 = (ma_uint8*)dst; + const float* src_f32 = (const float*)src; - ixabsv = (int*)&ixabs; - - lo = _mm_set_ps( - pSRC->sinc.table[ixabsv[3]], - pSRC->sinc.table[ixabsv[2]], - pSRC->sinc.table[ixabsv[1]], - pSRC->sinc.table[ixabsv[0]] - ); + ma_uint64 i; + for (i = 0; i < count; i += 1) { + ma_int32 r; + float x = src_f32[i]; + x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */ - hi = _mm_set_ps( - pSRC->sinc.table[ixabsv[3]+1], - pSRC->sinc.table[ixabsv[2]+1], - pSRC->sinc.table[ixabsv[1]+1], - pSRC->sinc.table[ixabsv[0]+1] - ); +#if 0 + /* The accurate way. */ + x = x + 1; /* -1..1 to 0..2 */ + x = x * 8388607.5f; /* 0..2 to 0..16777215 */ + x = x - 8388608.0f; /* 0..16777215 to -8388608..8388607 */ +#else + /* The fast way. */ + x = x * 8388607.0f; /* -1..1 to -8388607..8388607 */ +#endif - a = _mm_sub_ps(xabs, _mm_cvtepi32_ps(ixabs)); - r = ma_mix_f32_fast__sse2(lo, hi, a); + r = (ma_int32)x; + dst_s24[(i*3)+0] = (ma_uint8)((r & 0x0000FF) >> 0); + dst_s24[(i*3)+1] = (ma_uint8)((r & 0x00FF00) >> 8); + dst_s24[(i*3)+2] = (ma_uint8)((r & 0xFF0000) >> 16); + } - return r; + (void)ditherMode; /* No dithering for f32 -> s24. */ } -#endif -#if defined(MA_SUPPORT_AVX2) -static MA_INLINE __m256 ma_fabsf_avx2(__m256 x) +static MA_INLINE void ma_pcm_f32_to_s24__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - return _mm256_and_ps(_mm256_castsi256_ps(_mm256_set1_epi32(0x7FFFFFFF)), x); + ma_pcm_f32_to_s24__reference(dst, src, count, ditherMode); } -#if 0 -static MA_INLINE __m256 ma_src_sinc__interpolation_factor__avx2(const ma_src* pSRC, __m256 x) +#if defined(MA_SUPPORT_SSE2) +static MA_INLINE void ma_pcm_f32_to_s24__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - __m256 resolution256 = _mm256_set1_ps(MA_SRC_SINC_LOOKUP_TABLE_RESOLUTION); - __m256 xabs = ma_fabsf_avx2(x); - - xabs = _mm256_mul_ps(xabs, resolution256); - - __m256i ixabs = _mm256_cvttps_epi32(xabs); - __m256 a = _mm256_sub_ps(xabs, _mm256_cvtepi32_ps(ixabs)); - - - int* ixabsv = (int*)&ixabs; - - __m256 lo = _mm256_set_ps( - pSRC->sinc.table[ixabsv[7]], - pSRC->sinc.table[ixabsv[6]], - pSRC->sinc.table[ixabsv[5]], - pSRC->sinc.table[ixabsv[4]], - pSRC->sinc.table[ixabsv[3]], - pSRC->sinc.table[ixabsv[2]], - pSRC->sinc.table[ixabsv[1]], - pSRC->sinc.table[ixabsv[0]] - ); - - __m256 hi = _mm256_set_ps( - pSRC->sinc.table[ixabsv[7]+1], - pSRC->sinc.table[ixabsv[6]+1], - pSRC->sinc.table[ixabsv[5]+1], - pSRC->sinc.table[ixabsv[4]+1], - pSRC->sinc.table[ixabsv[3]+1], - pSRC->sinc.table[ixabsv[2]+1], - pSRC->sinc.table[ixabsv[1]+1], - pSRC->sinc.table[ixabsv[0]+1] - ); - - __m256 r = ma_mix_f32_fast__avx2(lo, hi, a); - - return r; + ma_pcm_f32_to_s24__optimized(dst, src, count, ditherMode); } #endif - -#endif - -#if defined(MA_SUPPORT_NEON) -static MA_INLINE float32x4_t ma_fabsf_neon(float32x4_t x) +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE void ma_pcm_f32_to_s24__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - return vabdq_f32(vmovq_n_f32(0), x); + ma_pcm_f32_to_s24__optimized(dst, src, count, ditherMode); } - -static MA_INLINE float32x4_t ma_src_sinc__interpolation_factor__neon(const ma_src* pSRC, float32x4_t x) +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_f32_to_s24__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - float32x4_t xabs; - int32x4_t ixabs; - float32x4_t a; - float32x4_t r; - int* ixabsv; - float lo[4]; - float hi[4]; - - xabs = ma_fabsf_neon(x); - xabs = vmulq_n_f32(xabs, MA_SRC_SINC_LOOKUP_TABLE_RESOLUTION); - ixabs = vcvtq_s32_f32(xabs); - - ixabsv = (int*)&ixabs; - - lo[0] = pSRC->sinc.table[ixabsv[0]]; - lo[1] = pSRC->sinc.table[ixabsv[1]]; - lo[2] = pSRC->sinc.table[ixabsv[2]]; - lo[3] = pSRC->sinc.table[ixabsv[3]]; - - hi[0] = pSRC->sinc.table[ixabsv[0]+1]; - hi[1] = pSRC->sinc.table[ixabsv[1]+1]; - hi[2] = pSRC->sinc.table[ixabsv[2]+1]; - hi[3] = pSRC->sinc.table[ixabsv[3]+1]; - - a = vsubq_f32(xabs, vcvtq_f32_s32(ixabs)); - r = ma_mix_f32_fast__neon(vld1q_f32(lo), vld1q_f32(hi), a); - - return r; + ma_pcm_f32_to_s24__optimized(dst, src, count, ditherMode); } #endif -ma_uint64 ma_src_read_deinterleaved__sinc(ma_src* pSRC, ma_uint64 frameCount, void** ppSamplesOut, void* pUserData) +void ma_pcm_f32_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) { - float factor; - float inverseFactor; - ma_int32 windowWidth; - ma_int32 windowWidth2; - ma_int32 windowWidthSIMD; - ma_int32 windowWidthSIMD2; - float* ppNextSamplesOut[MA_MAX_CHANNELS]; - float _windowSamplesUnaligned[MA_SRC_SINC_MAX_WINDOW_WIDTH*2 + MA_SIMD_ALIGNMENT]; - float* windowSamples; - float _iWindowFUnaligned[MA_SRC_SINC_MAX_WINDOW_WIDTH*2 + MA_SIMD_ALIGNMENT]; - float* iWindowF; - ma_int32 i; - ma_uint64 totalOutputFramesRead; - - ma_assert(pSRC != NULL); - ma_assert(frameCount > 0); - ma_assert(ppSamplesOut != NULL); - - factor = (float)pSRC->config.sampleRateIn / pSRC->config.sampleRateOut; - inverseFactor = 1/factor; - - windowWidth = (ma_int32)pSRC->config.sinc.windowWidth; - windowWidth2 = windowWidth*2; - - /* - There are cases where it's actually more efficient to increase the window width so that it's aligned with the respective - SIMD pipeline being used. - */ - windowWidthSIMD = windowWidth; - if (pSRC->useNEON) { - windowWidthSIMD = (windowWidthSIMD + 1) & ~(1); - } else if (pSRC->useAVX512) { - windowWidthSIMD = (windowWidthSIMD + 7) & ~(7); - } else if (pSRC->useAVX2) { - windowWidthSIMD = (windowWidthSIMD + 3) & ~(3); - } else if (pSRC->useSSE2) { - windowWidthSIMD = (windowWidthSIMD + 1) & ~(1); - } - - windowWidthSIMD2 = windowWidthSIMD*2; - (void)windowWidthSIMD2; /* <-- Silence a warning when SIMD is disabled. */ - - ma_copy_memory(ppNextSamplesOut, ppSamplesOut, sizeof(void*) * pSRC->config.channels); - - windowSamples = (float*)(((ma_uintptr)_windowSamplesUnaligned + MA_SIMD_ALIGNMENT-1) & ~(MA_SIMD_ALIGNMENT-1)); - ma_zero_memory(windowSamples, MA_SRC_SINC_MAX_WINDOW_WIDTH*2 * sizeof(float)); - - iWindowF = (float*)(((ma_uintptr)_iWindowFUnaligned + MA_SIMD_ALIGNMENT-1) & ~(MA_SIMD_ALIGNMENT-1)); - ma_zero_memory(iWindowF, MA_SRC_SINC_MAX_WINDOW_WIDTH*2 * sizeof(float)); - - for (i = 0; i < windowWidth2; ++i) { - iWindowF[i] = (float)(i - windowWidth); - } - - totalOutputFramesRead = 0; - while (totalOutputFramesRead < frameCount) { - ma_uint32 maxInputSamplesAvailableInCache; - float timeInBeg; - float timeInEnd; - ma_uint64 maxOutputFramesToRead; - ma_uint64 outputFramesRemaining; - ma_uint64 outputFramesToRead; - ma_uint32 iChannel; - ma_uint32 prevWindowPosInSamples; - ma_uint32 availableOutputFrames; - - /* - The maximum number of frames we can read this iteration depends on how many input samples we have available to us. This is the number - of input samples between the end of the window and the end of the cache. - */ - maxInputSamplesAvailableInCache = ma_countof(pSRC->sinc.input[0]) - (pSRC->config.sinc.windowWidth*2) - pSRC->sinc.windowPosInSamples; - if (maxInputSamplesAvailableInCache > pSRC->sinc.inputFrameCount) { - maxInputSamplesAvailableInCache = pSRC->sinc.inputFrameCount; - } - - /* Never consume the tail end of the input data if requested. */ - if (pSRC->config.neverConsumeEndOfInput) { - if (maxInputSamplesAvailableInCache >= pSRC->config.sinc.windowWidth) { - maxInputSamplesAvailableInCache -= pSRC->config.sinc.windowWidth; - } else { - maxInputSamplesAvailableInCache = 0; - } - } - - timeInBeg = pSRC->sinc.timeIn; - timeInEnd = (float)(pSRC->sinc.windowPosInSamples + maxInputSamplesAvailableInCache); - - ma_assert(timeInBeg >= 0); - ma_assert(timeInBeg <= timeInEnd); - - maxOutputFramesToRead = (ma_uint64)(((timeInEnd - timeInBeg) * inverseFactor)); - - outputFramesRemaining = frameCount - totalOutputFramesRead; - outputFramesToRead = outputFramesRemaining; - if (outputFramesToRead > maxOutputFramesToRead) { - outputFramesToRead = maxOutputFramesToRead; +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_f32_to_s24__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_f32_to_s24__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_f32_to_s24__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_f32_to_s24__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_f32_to_s24__optimized(dst, src, count, ditherMode); } +#endif +} - for (iChannel = 0; iChannel < pSRC->config.channels; iChannel += 1) { - /* Do SRC. */ - float timeIn = timeInBeg; - ma_uint32 iSample; - for (iSample = 0; iSample < outputFramesToRead; iSample += 1) { - float sampleOut = 0; - float iTimeInF = ma_floorf(timeIn); - ma_uint32 iTimeIn = (ma_uint32)iTimeInF; - ma_int32 iWindow = 0; - float tScalar; - - /* Pre-load the window samples into an aligned buffer to begin with. Need to put these into an aligned buffer to make SIMD easier. */ - windowSamples[0] = 0; /* <-- The first sample is always zero. */ - for (i = 1; i < windowWidth2; ++i) { - windowSamples[i] = pSRC->sinc.input[iChannel][iTimeIn + i]; - } - -#if defined(MA_SUPPORT_AVX2) || defined(MA_SUPPORT_AVX512) - if (pSRC->useAVX2 || pSRC->useAVX512) { - __m256i ixabs[MA_SRC_SINC_MAX_WINDOW_WIDTH*2/8]; - __m256 a[MA_SRC_SINC_MAX_WINDOW_WIDTH*2/8]; - __m256 resolution256; - __m256 t; - __m256 r; - ma_int32 windowWidth8; - ma_int32 iWindow8; - resolution256 = _mm256_set1_ps(MA_SRC_SINC_LOOKUP_TABLE_RESOLUTION); +static MA_INLINE void ma_pcm_f32_to_s32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_int32* dst_s32 = (ma_int32*)dst; + const float* src_f32 = (const float*)src; - t = _mm256_set1_ps((timeIn - iTimeInF)); - r = _mm256_set1_ps(0); + ma_uint32 i; + for (i = 0; i < count; i += 1) { + double x = src_f32[i]; + x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */ - windowWidth8 = windowWidthSIMD2 >> 3; - for (iWindow8 = 0; iWindow8 < windowWidth8; iWindow8 += 1) { - __m256 w = *((__m256*)iWindowF + iWindow8); +#if 0 + /* The accurate way. */ + x = x + 1; /* -1..1 to 0..2 */ + x = x * 2147483647.5; /* 0..2 to 0..4294967295 */ + x = x - 2147483648.0; /* 0...4294967295 to -2147483648..2147483647 */ +#else + /* The fast way. */ + x = x * 2147483647.0; /* -1..1 to -2147483647..2147483647 */ +#endif - __m256 xabs = _mm256_sub_ps(t, w); - xabs = ma_fabsf_avx2(xabs); - xabs = _mm256_mul_ps(xabs, resolution256); + dst_s32[i] = (ma_int32)x; + } - ixabs[iWindow8] = _mm256_cvttps_epi32(xabs); - a[iWindow8] = _mm256_sub_ps(xabs, _mm256_cvtepi32_ps(ixabs[iWindow8])); - } - - for (iWindow8 = 0; iWindow8 < windowWidth8; iWindow8 += 1) { - int* ixabsv = (int*)&ixabs[iWindow8]; - - __m256 lo = _mm256_set_ps( - pSRC->sinc.table[ixabsv[7]], - pSRC->sinc.table[ixabsv[6]], - pSRC->sinc.table[ixabsv[5]], - pSRC->sinc.table[ixabsv[4]], - pSRC->sinc.table[ixabsv[3]], - pSRC->sinc.table[ixabsv[2]], - pSRC->sinc.table[ixabsv[1]], - pSRC->sinc.table[ixabsv[0]] - ); - - __m256 hi = _mm256_set_ps( - pSRC->sinc.table[ixabsv[7]+1], - pSRC->sinc.table[ixabsv[6]+1], - pSRC->sinc.table[ixabsv[5]+1], - pSRC->sinc.table[ixabsv[4]+1], - pSRC->sinc.table[ixabsv[3]+1], - pSRC->sinc.table[ixabsv[2]+1], - pSRC->sinc.table[ixabsv[1]+1], - pSRC->sinc.table[ixabsv[0]+1] - ); - - __m256 s = *((__m256*)windowSamples + iWindow8); - r = _mm256_add_ps(r, _mm256_mul_ps(s, ma_mix_f32_fast__avx2(lo, hi, a[iWindow8]))); - } + (void)ditherMode; /* No dithering for f32 -> s32. */ +} - /* Horizontal add. */ - __m256 x = _mm256_hadd_ps(r, _mm256_permute2f128_ps(r, r, 1)); - x = _mm256_hadd_ps(x, x); - x = _mm256_hadd_ps(x, x); - sampleOut += _mm_cvtss_f32(_mm256_castps256_ps128(x)); +static MA_INLINE void ma_pcm_f32_to_s32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_f32_to_s32__reference(dst, src, count, ditherMode); +} - iWindow += windowWidth8 * 8; - } - else -#endif #if defined(MA_SUPPORT_SSE2) - if (pSRC->useSSE2) { - __m128 t = _mm_set1_ps((timeIn - iTimeInF)); - __m128 r = _mm_set1_ps(0); - - ma_int32 windowWidth4 = windowWidthSIMD2 >> 2; - ma_int32 iWindow4; - for (iWindow4 = 0; iWindow4 < windowWidth4; iWindow4 += 1) { - __m128* s = (__m128*)windowSamples + iWindow4; - __m128* w = (__m128*)iWindowF + iWindow4; - - __m128 a = ma_src_sinc__interpolation_factor__sse2(pSRC, _mm_sub_ps(t, *w)); - r = _mm_add_ps(r, _mm_mul_ps(*s, a)); - } - - sampleOut += ((float*)(&r))[0]; - sampleOut += ((float*)(&r))[1]; - sampleOut += ((float*)(&r))[2]; - sampleOut += ((float*)(&r))[3]; - - iWindow += windowWidth4 * 4; - } - else +static MA_INLINE void ma_pcm_f32_to_s32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode); +} #endif -#if defined(MA_SUPPORT_NEON) - if (pSRC->useNEON) { - float32x4_t t = vmovq_n_f32((timeIn - iTimeInF)); - float32x4_t r = vmovq_n_f32(0); - - ma_int32 windowWidth4 = windowWidthSIMD2 >> 2; - ma_int32 iWindow4; - for (iWindow4 = 0; iWindow4 < windowWidth4; iWindow4 += 1) { - float32x4_t* s = (float32x4_t*)windowSamples + iWindow4; - float32x4_t* w = (float32x4_t*)iWindowF + iWindow4; - - float32x4_t a = ma_src_sinc__interpolation_factor__neon(pSRC, vsubq_f32(t, *w)); - r = vaddq_f32(r, vmulq_f32(*s, a)); - } - - sampleOut += ((float*)(&r))[0]; - sampleOut += ((float*)(&r))[1]; - sampleOut += ((float*)(&r))[2]; - sampleOut += ((float*)(&r))[3]; - - iWindow += windowWidth4 * 4; - } - else +#if defined(MA_SUPPORT_AVX2) +static MA_INLINE void ma_pcm_f32_to_s32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode); +} +#endif +#if defined(MA_SUPPORT_NEON) +static MA_INLINE void ma_pcm_f32_to_s32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode); +} #endif - { - iWindow += 1; /* The first one is a dummy for SIMD alignment purposes. Skip it. */ - } - - /* Non-SIMD/Reference implementation. */ - tScalar = (timeIn - iTimeIn); - for (; iWindow < windowWidth2; iWindow += 1) { - float s = windowSamples[iWindow]; - float w = iWindowF[iWindow]; - - float a = ma_src_sinc__interpolation_factor(pSRC, (tScalar - w)); - float r = s * a; - - sampleOut += r; - } - - ppNextSamplesOut[iChannel][iSample] = (float)sampleOut; - - timeIn += factor; - } - ppNextSamplesOut[iChannel] += outputFramesToRead; +void ma_pcm_f32_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_f32_to_s32__reference(dst, src, count, ditherMode); +#else + # if MA_PREFERRED_SIMD == MA_SIMD_AVX2 + if (ma_has_avx2()) { + ma_pcm_f32_to_s32__avx2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2 + if (ma_has_sse2()) { + ma_pcm_f32_to_s32__sse2(dst, src, count, ditherMode); + } else + #elif MA_PREFERRED_SIMD == MA_SIMD_NEON + if (ma_has_neon()) { + ma_pcm_f32_to_s32__neon(dst, src, count, ditherMode); + } else + #endif + { + ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode); } +#endif +} - totalOutputFramesRead += outputFramesToRead; - - prevWindowPosInSamples = pSRC->sinc.windowPosInSamples; - - pSRC->sinc.timeIn += ((ma_int64)outputFramesToRead * factor); /* Cast to int64 required for VC6. */ - pSRC->sinc.windowPosInSamples = (ma_uint32)pSRC->sinc.timeIn; - pSRC->sinc.inputFrameCount -= pSRC->sinc.windowPosInSamples - prevWindowPosInSamples; - /* If the window has reached a point where we cannot read a whole output sample it needs to be moved back to the start. */ - availableOutputFrames = (ma_uint32)((timeInEnd - pSRC->sinc.timeIn) * inverseFactor); +void ma_pcm_f32_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode) +{ + (void)ditherMode; - if (availableOutputFrames == 0) { - size_t samplesToMove = ma_countof(pSRC->sinc.input[0]) - pSRC->sinc.windowPosInSamples; + ma_copy_memory_64(dst, src, count * sizeof(float)); +} - pSRC->sinc.timeIn -= ma_floorf(pSRC->sinc.timeIn); - pSRC->sinc.windowPosInSamples = 0; - /* Move everything from the end of the cache up to the front. */ - for (iChannel = 0; iChannel < pSRC->config.channels; iChannel += 1) { - memmove(pSRC->sinc.input[iChannel], pSRC->sinc.input[iChannel] + ma_countof(pSRC->sinc.input[iChannel]) - samplesToMove, samplesToMove * sizeof(*pSRC->sinc.input[iChannel])); - } - } +static void ma_pcm_interleave_f32__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) +{ + float* dst_f32 = (float*)dst; + const float** src_f32 = (const float**)src; - /* Read more data from the client if required. */ - if (pSRC->isEndOfInputLoaded) { - pSRC->isEndOfInputLoaded = MA_FALSE; - break; + ma_uint64 iFrame; + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + ma_uint32 iChannel; + for (iChannel = 0; iChannel < channels; iChannel += 1) { + dst_f32[iFrame*channels + iChannel] = src_f32[iChannel][iFrame]; } + } +} - /* - Everything beyond this point is reloading. If we're at the end of the input data we do _not_ want to try reading any more in this function call. If the - caller wants to keep trying, they can reload their internal data sources and call this function again. We should never be - */ - ma_assert(pSRC->isEndOfInputLoaded == MA_FALSE); - - if (pSRC->sinc.inputFrameCount <= pSRC->config.sinc.windowWidth || availableOutputFrames == 0) { - float* ppInputDst[MA_MAX_CHANNELS] = {0}; - ma_uint32 framesToReadFromClient; - ma_uint32 framesReadFromClient; - ma_uint32 leftoverFrames; - - for (iChannel = 0; iChannel < pSRC->config.channels; iChannel += 1) { - ppInputDst[iChannel] = pSRC->sinc.input[iChannel] + pSRC->config.sinc.windowWidth + pSRC->sinc.inputFrameCount; - } - - /* Now read data from the client. */ - framesToReadFromClient = ma_countof(pSRC->sinc.input[0]) - (pSRC->config.sinc.windowWidth + pSRC->sinc.inputFrameCount); +static void ma_pcm_interleave_f32__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) +{ + ma_pcm_interleave_f32__reference(dst, src, frameCount, channels); +} - framesReadFromClient = 0; - if (framesToReadFromClient > 0) { - framesReadFromClient = pSRC->config.onReadDeinterleaved(pSRC, framesToReadFromClient, (void**)ppInputDst, pUserData); - } +void ma_pcm_interleave_f32(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_interleave_f32__reference(dst, src, frameCount, channels); +#else + ma_pcm_interleave_f32__optimized(dst, src, frameCount, channels); +#endif +} - if (framesReadFromClient != framesToReadFromClient) { - pSRC->isEndOfInputLoaded = MA_TRUE; - } else { - pSRC->isEndOfInputLoaded = MA_FALSE; - } - if (framesReadFromClient != 0) { - pSRC->sinc.inputFrameCount += framesReadFromClient; - } else { - /* We couldn't get anything more from the client. If no more output samples can be computed from the available input samples we need to return. */ - if (pSRC->config.neverConsumeEndOfInput) { - if ((pSRC->sinc.inputFrameCount * inverseFactor) <= pSRC->config.sinc.windowWidth) { - break; - } - } else { - if ((pSRC->sinc.inputFrameCount * inverseFactor) < 1) { - break; - } - } - } +static void ma_pcm_deinterleave_f32__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) +{ + float** dst_f32 = (float**)dst; + const float* src_f32 = (const float*)src; - /* Anything left over in the cache must be set to zero. */ - leftoverFrames = ma_countof(pSRC->sinc.input[0]) - (pSRC->config.sinc.windowWidth + pSRC->sinc.inputFrameCount); - if (leftoverFrames > 0) { - for (iChannel = 0; iChannel < pSRC->config.channels; iChannel += 1) { - ma_zero_memory(pSRC->sinc.input[iChannel] + pSRC->config.sinc.windowWidth + pSRC->sinc.inputFrameCount, leftoverFrames * sizeof(float)); - } - } + ma_uint64 iFrame; + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + ma_uint32 iChannel; + for (iChannel = 0; iChannel < channels; iChannel += 1) { + dst_f32[iChannel][iFrame] = src_f32[iFrame*channels + iChannel]; } } - - return totalOutputFramesRead; } +static void ma_pcm_deinterleave_f32__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) +{ + ma_pcm_deinterleave_f32__reference(dst, src, frameCount, channels); +} +void ma_pcm_deinterleave_f32(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels) +{ +#ifdef MA_USE_REFERENCE_CONVERSION_APIS + ma_pcm_deinterleave_f32__reference(dst, src, frameCount, channels); +#else + ma_pcm_deinterleave_f32__optimized(dst, src, frameCount, channels); +#endif +} -/************************************************************************************************************************************************************** - -Format Conversion -**************************************************************************************************************************************************************/ void ma_pcm_convert(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 sampleCount, ma_dither_mode ditherMode) { if (formatOut == formatIn) { @@ -31992,6 +34515,11 @@ void ma_pcm_convert(void* pOut, ma_format formatOut, const void* pIn, ma_format } } +void ma_convert_pcm_frames_format(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 frameCount, ma_uint32 channels, ma_dither_mode ditherMode) +{ + ma_pcm_convert(pOut, formatOut, pIn, formatIn, frameCount * channels, ditherMode); +} + void ma_deinterleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void* pInterleavedPCMFrames, void** ppDeinterleavedPCMFrames) { if (pInterleavedPCMFrames == NULL || ppDeinterleavedPCMFrames == NULL) { @@ -32090,610 +34618,719 @@ void ma_interleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 fr -typedef struct +/************************************************************************************************************************************************************** + +Channel Maps + +**************************************************************************************************************************************************************/ +static void ma_get_standard_channel_map_microsoft(ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) { - ma_pcm_converter* pDSP; - void* pUserDataForClient; -} ma_pcm_converter_callback_data; + /* Based off the speaker configurations mentioned here: https://docs.microsoft.com/en-us/windows-hardware/drivers/ddi/content/ksmedia/ns-ksmedia-ksaudio_channel_config */ + switch (channels) + { + case 1: + { + channelMap[0] = MA_CHANNEL_MONO; + } break; + + case 2: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + } break; + + case 3: /* Not defined, but best guess. */ + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_FRONT_CENTER; + } break; + + case 4: + { +#ifndef MA_USE_QUAD_MICROSOFT_CHANNEL_MAP + /* Surround. Using the Surround profile has the advantage of the 3rd channel (MA_CHANNEL_FRONT_CENTER) mapping nicely with higher channel counts. */ + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_FRONT_CENTER; + channelMap[3] = MA_CHANNEL_BACK_CENTER; +#else + /* Quad. */ + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_BACK_LEFT; + channelMap[3] = MA_CHANNEL_BACK_RIGHT; +#endif + } break; + + case 5: /* Not defined, but best guess. */ + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_FRONT_CENTER; + channelMap[3] = MA_CHANNEL_BACK_LEFT; + channelMap[4] = MA_CHANNEL_BACK_RIGHT; + } break; + + case 6: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_FRONT_CENTER; + channelMap[3] = MA_CHANNEL_LFE; + channelMap[4] = MA_CHANNEL_SIDE_LEFT; + channelMap[5] = MA_CHANNEL_SIDE_RIGHT; + } break; + + case 7: /* Not defined, but best guess. */ + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_FRONT_CENTER; + channelMap[3] = MA_CHANNEL_LFE; + channelMap[4] = MA_CHANNEL_BACK_CENTER; + channelMap[5] = MA_CHANNEL_SIDE_LEFT; + channelMap[6] = MA_CHANNEL_SIDE_RIGHT; + } break; + + case 8: + default: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_FRONT_CENTER; + channelMap[3] = MA_CHANNEL_LFE; + channelMap[4] = MA_CHANNEL_BACK_LEFT; + channelMap[5] = MA_CHANNEL_BACK_RIGHT; + channelMap[6] = MA_CHANNEL_SIDE_LEFT; + channelMap[7] = MA_CHANNEL_SIDE_RIGHT; + } break; + } + + /* Remainder. */ + if (channels > 8) { + ma_uint32 iChannel; + for (iChannel = 8; iChannel < MA_MAX_CHANNELS; ++iChannel) { + channelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-8)); + } + } +} -ma_uint32 ma_pcm_converter__pre_format_converter_on_read(ma_format_converter* pConverter, ma_uint32 frameCount, void* pFramesOut, void* pUserData) +static void ma_get_standard_channel_map_alsa(ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) { - ma_pcm_converter_callback_data* pData; - ma_pcm_converter* pDSP; + switch (channels) + { + case 1: + { + channelMap[0] = MA_CHANNEL_MONO; + } break; + + case 2: + { + channelMap[0] = MA_CHANNEL_LEFT; + channelMap[1] = MA_CHANNEL_RIGHT; + } break; + + case 3: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_FRONT_CENTER; + } break; + + case 4: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_BACK_LEFT; + channelMap[3] = MA_CHANNEL_BACK_RIGHT; + } break; + + case 5: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_BACK_LEFT; + channelMap[3] = MA_CHANNEL_BACK_RIGHT; + channelMap[4] = MA_CHANNEL_FRONT_CENTER; + } break; - (void)pConverter; + case 6: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_BACK_LEFT; + channelMap[3] = MA_CHANNEL_BACK_RIGHT; + channelMap[4] = MA_CHANNEL_FRONT_CENTER; + channelMap[5] = MA_CHANNEL_LFE; + } break; - pData = (ma_pcm_converter_callback_data*)pUserData; - ma_assert(pData != NULL); + case 7: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_BACK_LEFT; + channelMap[3] = MA_CHANNEL_BACK_RIGHT; + channelMap[4] = MA_CHANNEL_FRONT_CENTER; + channelMap[5] = MA_CHANNEL_LFE; + channelMap[6] = MA_CHANNEL_BACK_CENTER; + } break; - pDSP = pData->pDSP; - ma_assert(pDSP != NULL); + case 8: + default: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_BACK_LEFT; + channelMap[3] = MA_CHANNEL_BACK_RIGHT; + channelMap[4] = MA_CHANNEL_FRONT_CENTER; + channelMap[5] = MA_CHANNEL_LFE; + channelMap[6] = MA_CHANNEL_SIDE_LEFT; + channelMap[7] = MA_CHANNEL_SIDE_RIGHT; + } break; + } - return pDSP->onRead(pDSP, pFramesOut, frameCount, pData->pUserDataForClient); + /* Remainder. */ + if (channels > 8) { + ma_uint32 iChannel; + for (iChannel = 8; iChannel < MA_MAX_CHANNELS; ++iChannel) { + channelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-8)); + } + } } -ma_uint32 ma_pcm_converter__post_format_converter_on_read(ma_format_converter* pConverter, ma_uint32 frameCount, void* pFramesOut, void* pUserData) +static void ma_get_standard_channel_map_rfc3551(ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) { - ma_pcm_converter_callback_data* pData; - ma_pcm_converter* pDSP; - - (void)pConverter; - - pData = (ma_pcm_converter_callback_data*)pUserData; - ma_assert(pData != NULL); - - pDSP = pData->pDSP; - ma_assert(pDSP != NULL); - - /* When this version of this callback is used it means we're reading directly from the client. */ - ma_assert(pDSP->isPreFormatConversionRequired == MA_FALSE); - ma_assert(pDSP->isChannelRoutingRequired == MA_FALSE); - ma_assert(pDSP->isSRCRequired == MA_FALSE); + switch (channels) + { + case 1: + { + channelMap[0] = MA_CHANNEL_MONO; + } break; - return pDSP->onRead(pDSP, pFramesOut, frameCount, pData->pUserDataForClient); -} + case 2: + { + channelMap[0] = MA_CHANNEL_LEFT; + channelMap[1] = MA_CHANNEL_RIGHT; + } break; -ma_uint32 ma_pcm_converter__post_format_converter_on_read_deinterleaved(ma_format_converter* pConverter, ma_uint32 frameCount, void** ppSamplesOut, void* pUserData) -{ - ma_pcm_converter_callback_data* pData; - ma_pcm_converter* pDSP; + case 3: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_FRONT_CENTER; + } break; - (void)pConverter; + case 4: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_CENTER; + channelMap[2] = MA_CHANNEL_FRONT_RIGHT; + channelMap[3] = MA_CHANNEL_BACK_CENTER; + } break; - pData = (ma_pcm_converter_callback_data*)pUserData; - ma_assert(pData != NULL); + case 5: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_FRONT_CENTER; + channelMap[3] = MA_CHANNEL_BACK_LEFT; + channelMap[4] = MA_CHANNEL_BACK_RIGHT; + } break; - pDSP = pData->pDSP; - ma_assert(pDSP != NULL); + case 6: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_SIDE_LEFT; + channelMap[2] = MA_CHANNEL_FRONT_CENTER; + channelMap[3] = MA_CHANNEL_FRONT_RIGHT; + channelMap[4] = MA_CHANNEL_SIDE_RIGHT; + channelMap[5] = MA_CHANNEL_BACK_CENTER; + } break; + } - if (!pDSP->isChannelRoutingAtStart) { - return (ma_uint32)ma_channel_router_read_deinterleaved(&pDSP->channelRouter, frameCount, ppSamplesOut, pUserData); - } else { - if (pDSP->isSRCRequired) { - return (ma_uint32)ma_src_read_deinterleaved(&pDSP->src, frameCount, ppSamplesOut, pUserData); - } else { - return (ma_uint32)ma_channel_router_read_deinterleaved(&pDSP->channelRouter, frameCount, ppSamplesOut, pUserData); + /* Remainder. */ + if (channels > 8) { + ma_uint32 iChannel; + for (iChannel = 6; iChannel < MA_MAX_CHANNELS; ++iChannel) { + channelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-6)); } } } -ma_uint32 ma_pcm_converter__src_on_read_deinterleaved(ma_src* pSRC, ma_uint32 frameCount, void** ppSamplesOut, void* pUserData) +static void ma_get_standard_channel_map_flac(ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) { - ma_pcm_converter_callback_data* pData; - ma_pcm_converter* pDSP; - - (void)pSRC; + switch (channels) + { + case 1: + { + channelMap[0] = MA_CHANNEL_MONO; + } break; - pData = (ma_pcm_converter_callback_data*)pUserData; - ma_assert(pData != NULL); + case 2: + { + channelMap[0] = MA_CHANNEL_LEFT; + channelMap[1] = MA_CHANNEL_RIGHT; + } break; - pDSP = pData->pDSP; - ma_assert(pDSP != NULL); + case 3: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_FRONT_CENTER; + } break; - /* If the channel routing stage is at the front we need to read from that. Otherwise we read from the pre format converter. */ - if (pDSP->isChannelRoutingAtStart) { - return (ma_uint32)ma_channel_router_read_deinterleaved(&pDSP->channelRouter, frameCount, ppSamplesOut, pUserData); - } else { - return (ma_uint32)ma_format_converter_read_deinterleaved(&pDSP->formatConverterIn, frameCount, ppSamplesOut, pUserData); - } -} + case 4: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_BACK_LEFT; + channelMap[3] = MA_CHANNEL_BACK_RIGHT; + } break; -ma_uint32 ma_pcm_converter__channel_router_on_read_deinterleaved(ma_channel_router* pRouter, ma_uint32 frameCount, void** ppSamplesOut, void* pUserData) -{ - ma_pcm_converter_callback_data* pData; - ma_pcm_converter* pDSP; + case 5: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_FRONT_CENTER; + channelMap[3] = MA_CHANNEL_BACK_LEFT; + channelMap[4] = MA_CHANNEL_BACK_RIGHT; + } break; - (void)pRouter; + case 6: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_FRONT_CENTER; + channelMap[3] = MA_CHANNEL_LFE; + channelMap[4] = MA_CHANNEL_BACK_LEFT; + channelMap[5] = MA_CHANNEL_BACK_RIGHT; + } break; - pData = (ma_pcm_converter_callback_data*)pUserData; - ma_assert(pData != NULL); + case 7: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_FRONT_CENTER; + channelMap[3] = MA_CHANNEL_LFE; + channelMap[4] = MA_CHANNEL_BACK_CENTER; + channelMap[5] = MA_CHANNEL_SIDE_LEFT; + channelMap[6] = MA_CHANNEL_SIDE_RIGHT; + } break; - pDSP = pData->pDSP; - ma_assert(pDSP != NULL); + case 8: + default: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_FRONT_CENTER; + channelMap[3] = MA_CHANNEL_LFE; + channelMap[4] = MA_CHANNEL_BACK_LEFT; + channelMap[5] = MA_CHANNEL_BACK_RIGHT; + channelMap[6] = MA_CHANNEL_SIDE_LEFT; + channelMap[7] = MA_CHANNEL_SIDE_RIGHT; + } break; + } - /* If the channel routing stage is at the front of the pipeline we read from the pre format converter. Otherwise we read from the sample rate converter. */ - if (pDSP->isChannelRoutingAtStart) { - return (ma_uint32)ma_format_converter_read_deinterleaved(&pDSP->formatConverterIn, frameCount, ppSamplesOut, pUserData); - } else { - if (pDSP->isSRCRequired) { - return (ma_uint32)ma_src_read_deinterleaved(&pDSP->src, frameCount, ppSamplesOut, pUserData); - } else { - return (ma_uint32)ma_format_converter_read_deinterleaved(&pDSP->formatConverterIn, frameCount, ppSamplesOut, pUserData); + /* Remainder. */ + if (channels > 8) { + ma_uint32 iChannel; + for (iChannel = 8; iChannel < MA_MAX_CHANNELS; ++iChannel) { + channelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-8)); } } } -ma_result ma_pcm_converter_init(const ma_pcm_converter_config* pConfig, ma_pcm_converter* pDSP) +static void ma_get_standard_channel_map_vorbis(ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) { - ma_result result; - - if (pDSP == NULL) { - return MA_INVALID_ARGS; - } - - ma_zero_object(pDSP); - pDSP->onRead = pConfig->onRead; - pDSP->pUserData = pConfig->pUserData; - pDSP->isDynamicSampleRateAllowed = pConfig->allowDynamicSampleRate; - - /* - In general, this is the pipeline used for data conversion. Note that this can actually change which is explained later. - - Pre Format Conversion -> Sample Rate Conversion -> Channel Routing -> Post Format Conversion - - Pre Format Conversion - --------------------- - This is where the sample data is converted to a format that's usable by the later stages in the pipeline. Input data - is converted to deinterleaved floating-point. - - Channel Routing - --------------- - Channel routing is where stereo is converted to 5.1, mono is converted to stereo, etc. This stage depends on the - pre format conversion stage. - - Sample Rate Conversion - ---------------------- - Sample rate conversion depends on the pre format conversion stage and as the name implies performs sample rate conversion. - - Post Format Conversion - ---------------------- - This stage is where our deinterleaved floating-point data from the previous stages are converted to the requested output - format. - - - Optimizations - ------------- - Sometimes the conversion pipeline is rearranged for efficiency. The first obvious optimization is to eliminate unnecessary - stages in the pipeline. When no channel routing nor sample rate conversion is necessary, the entire pipeline is optimized - down to just this: - - Post Format Conversion - - When sample rate conversion is not unnecessary: - - Pre Format Conversion -> Channel Routing -> Post Format Conversion - - When channel routing is unnecessary: - - Pre Format Conversion -> Sample Rate Conversion -> Post Format Conversion - - A slightly less obvious optimization is used depending on whether or not we are increasing or decreasing the number of - channels. Because everything in the pipeline works on a per-channel basis, the efficiency of the pipeline is directly - proportionate to the number of channels that need to be processed. Therefore, it's can be more efficient to move the - channel conversion stage to an earlier or later stage. When the channel count is being reduced, we move the channel - conversion stage to the start of the pipeline so that later stages can work on a smaller number of channels at a time. - Otherwise, we move the channel conversion stage to the end of the pipeline. When reducing the channel count, the pipeline - will look like this: - - Pre Format Conversion -> Channel Routing -> Sample Rate Conversion -> Post Format Conversion - - Notice how the Channel Routing and Sample Rate Conversion stages are swapped so that the SRC stage has less data to process. - */ - - /* First we need to determine what's required and what's not. */ - if (pConfig->sampleRateIn != pConfig->sampleRateOut || pConfig->allowDynamicSampleRate) { - pDSP->isSRCRequired = MA_TRUE; - } - if (pConfig->channelsIn != pConfig->channelsOut || !ma_channel_map_equal(pConfig->channelsIn, pConfig->channelMapIn, pConfig->channelMapOut)) { - pDSP->isChannelRoutingRequired = MA_TRUE; - } - - /* If neither a sample rate conversion nor channel conversion is necessary we can skip the pre format conversion. */ - if (!pDSP->isSRCRequired && !pDSP->isChannelRoutingRequired) { - /* We don't need a pre format conversion stage, but we may still need a post format conversion stage. */ - if (pConfig->formatIn != pConfig->formatOut) { - pDSP->isPostFormatConversionRequired = MA_TRUE; - } - } else { - pDSP->isPreFormatConversionRequired = MA_TRUE; - pDSP->isPostFormatConversionRequired = MA_TRUE; - } + /* In Vorbis' type 0 channel mapping, the first two channels are not always the standard left/right - it will have the center speaker where the right usually goes. Why?! */ + switch (channels) + { + case 1: + { + channelMap[0] = MA_CHANNEL_MONO; + } break; - /* Use a passthrough if none of the stages are being used. */ - if (!pDSP->isPreFormatConversionRequired && !pDSP->isPostFormatConversionRequired && !pDSP->isChannelRoutingRequired && !pDSP->isSRCRequired) { - pDSP->isPassthrough = MA_TRUE; - } + case 2: + { + channelMap[0] = MA_CHANNEL_LEFT; + channelMap[1] = MA_CHANNEL_RIGHT; + } break; - /* Move the channel conversion stage to the start of the pipeline if we are reducing the channel count. */ - if (pConfig->channelsOut < pConfig->channelsIn) { - pDSP->isChannelRoutingAtStart = MA_TRUE; - } + case 3: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_CENTER; + channelMap[2] = MA_CHANNEL_FRONT_RIGHT; + } break; + case 4: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_BACK_LEFT; + channelMap[3] = MA_CHANNEL_BACK_RIGHT; + } break; - /* - We always initialize every stage of the pipeline regardless of whether or not the stage is used because it simplifies - a few things when it comes to dynamically changing properties post-initialization. - */ - result = MA_SUCCESS; + case 5: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_CENTER; + channelMap[2] = MA_CHANNEL_FRONT_RIGHT; + channelMap[3] = MA_CHANNEL_BACK_LEFT; + channelMap[4] = MA_CHANNEL_BACK_RIGHT; + } break; - /* Pre format conversion. */ - { - ma_format_converter_config preFormatConverterConfig = ma_format_converter_config_init( - pConfig->formatIn, - ma_format_f32, - pConfig->channelsIn, - ma_pcm_converter__pre_format_converter_on_read, - pDSP - ); - preFormatConverterConfig.ditherMode = pConfig->ditherMode; - preFormatConverterConfig.noSSE2 = pConfig->noSSE2; - preFormatConverterConfig.noAVX2 = pConfig->noAVX2; - preFormatConverterConfig.noAVX512 = pConfig->noAVX512; - preFormatConverterConfig.noNEON = pConfig->noNEON; - - result = ma_format_converter_init(&preFormatConverterConfig, &pDSP->formatConverterIn); - if (result != MA_SUCCESS) { - return result; - } - } + case 6: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_CENTER; + channelMap[2] = MA_CHANNEL_FRONT_RIGHT; + channelMap[3] = MA_CHANNEL_BACK_LEFT; + channelMap[4] = MA_CHANNEL_BACK_RIGHT; + channelMap[5] = MA_CHANNEL_LFE; + } break; - /* - Post format conversion. The exact configuration for this depends on whether or not we are reading data directly from the client - or from an earlier stage in the pipeline. - */ - { - ma_format_converter_config postFormatConverterConfig = ma_format_converter_config_init_new(); - postFormatConverterConfig.formatIn = pConfig->formatIn; - postFormatConverterConfig.formatOut = pConfig->formatOut; - postFormatConverterConfig.channels = pConfig->channelsOut; - postFormatConverterConfig.ditherMode = pConfig->ditherMode; - postFormatConverterConfig.noSSE2 = pConfig->noSSE2; - postFormatConverterConfig.noAVX2 = pConfig->noAVX2; - postFormatConverterConfig.noAVX512 = pConfig->noAVX512; - postFormatConverterConfig.noNEON = pConfig->noNEON; - if (pDSP->isPreFormatConversionRequired) { - postFormatConverterConfig.onReadDeinterleaved = ma_pcm_converter__post_format_converter_on_read_deinterleaved; - postFormatConverterConfig.formatIn = ma_format_f32; - } else { - postFormatConverterConfig.onRead = ma_pcm_converter__post_format_converter_on_read; - } + case 7: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_CENTER; + channelMap[2] = MA_CHANNEL_FRONT_RIGHT; + channelMap[3] = MA_CHANNEL_SIDE_LEFT; + channelMap[4] = MA_CHANNEL_SIDE_RIGHT; + channelMap[5] = MA_CHANNEL_BACK_CENTER; + channelMap[6] = MA_CHANNEL_LFE; + } break; - result = ma_format_converter_init(&postFormatConverterConfig, &pDSP->formatConverterOut); - if (result != MA_SUCCESS) { - return result; - } + case 8: + default: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_CENTER; + channelMap[2] = MA_CHANNEL_FRONT_RIGHT; + channelMap[3] = MA_CHANNEL_SIDE_LEFT; + channelMap[4] = MA_CHANNEL_SIDE_RIGHT; + channelMap[5] = MA_CHANNEL_BACK_LEFT; + channelMap[6] = MA_CHANNEL_BACK_RIGHT; + channelMap[7] = MA_CHANNEL_LFE; + } break; } - /* SRC */ - { - ma_src_config srcConfig = ma_src_config_init( - pConfig->sampleRateIn, - pConfig->sampleRateOut, - ((pConfig->channelsIn < pConfig->channelsOut) ? pConfig->channelsIn : pConfig->channelsOut), - ma_pcm_converter__src_on_read_deinterleaved, - pDSP - ); - srcConfig.algorithm = pConfig->srcAlgorithm; - srcConfig.neverConsumeEndOfInput = pConfig->neverConsumeEndOfInput; - srcConfig.noSSE2 = pConfig->noSSE2; - srcConfig.noAVX2 = pConfig->noAVX2; - srcConfig.noAVX512 = pConfig->noAVX512; - srcConfig.noNEON = pConfig->noNEON; - ma_copy_memory(&srcConfig.sinc, &pConfig->sinc, sizeof(pConfig->sinc)); - - result = ma_src_init(&srcConfig, &pDSP->src); - if (result != MA_SUCCESS) { - return result; + /* Remainder. */ + if (channels > 8) { + ma_uint32 iChannel; + for (iChannel = 8; iChannel < MA_MAX_CHANNELS; ++iChannel) { + channelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-8)); } } +} - /* Channel conversion */ +static void ma_get_standard_channel_map_sound4(ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) +{ + switch (channels) { - ma_channel_router_config routerConfig = ma_channel_router_config_init( - pConfig->channelsIn, - pConfig->channelMapIn, - pConfig->channelsOut, - pConfig->channelMapOut, - pConfig->channelMixMode, - ma_pcm_converter__channel_router_on_read_deinterleaved, - pDSP); - routerConfig.noSSE2 = pConfig->noSSE2; - routerConfig.noAVX2 = pConfig->noAVX2; - routerConfig.noAVX512 = pConfig->noAVX512; - routerConfig.noNEON = pConfig->noNEON; - - result = ma_channel_router_init(&routerConfig, &pDSP->channelRouter); - if (result != MA_SUCCESS) { - return result; - } - } + case 1: + { + channelMap[0] = MA_CHANNEL_MONO; + } break; - return MA_SUCCESS; -} + case 2: + { + channelMap[0] = MA_CHANNEL_LEFT; + channelMap[1] = MA_CHANNEL_RIGHT; + } break; + case 3: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_BACK_CENTER; + } break; -ma_result ma_pcm_converter_refresh_sample_rate(ma_pcm_converter* pDSP) -{ - /* The SRC stage will already have been initialized so we can just set it there. */ - ma_src_set_sample_rate(&pDSP->src, pDSP->src.config.sampleRateIn, pDSP->src.config.sampleRateOut); - return MA_SUCCESS; -} + case 4: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_BACK_LEFT; + channelMap[3] = MA_CHANNEL_BACK_RIGHT; + } break; -ma_result ma_pcm_converter_set_input_sample_rate(ma_pcm_converter* pDSP, ma_uint32 sampleRateIn) -{ - if (pDSP == NULL) { - return MA_INVALID_ARGS; - } + case 5: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_BACK_LEFT; + channelMap[3] = MA_CHANNEL_BACK_RIGHT; + channelMap[4] = MA_CHANNEL_FRONT_CENTER; + } break; - /* Must have a sample rate of > 0. */ - if (sampleRateIn == 0) { - return MA_INVALID_ARGS; - } + case 6: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_BACK_LEFT; + channelMap[3] = MA_CHANNEL_BACK_RIGHT; + channelMap[4] = MA_CHANNEL_FRONT_CENTER; + channelMap[5] = MA_CHANNEL_LFE; + } break; - /* Must have been initialized with allowDynamicSampleRate. */ - if (!pDSP->isDynamicSampleRateAllowed) { - return MA_INVALID_OPERATION; + case 7: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_BACK_LEFT; + channelMap[3] = MA_CHANNEL_BACK_RIGHT; + channelMap[4] = MA_CHANNEL_FRONT_CENTER; + channelMap[5] = MA_CHANNEL_BACK_CENTER; + channelMap[6] = MA_CHANNEL_LFE; + } break; + + case 8: + default: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_BACK_LEFT; + channelMap[3] = MA_CHANNEL_BACK_RIGHT; + channelMap[4] = MA_CHANNEL_FRONT_CENTER; + channelMap[5] = MA_CHANNEL_LFE; + channelMap[6] = MA_CHANNEL_SIDE_LEFT; + channelMap[7] = MA_CHANNEL_SIDE_RIGHT; + } break; } - ma_atomic_exchange_32(&pDSP->src.config.sampleRateIn, sampleRateIn); - return ma_pcm_converter_refresh_sample_rate(pDSP); + /* Remainder. */ + if (channels > 8) { + ma_uint32 iChannel; + for (iChannel = 8; iChannel < MA_MAX_CHANNELS; ++iChannel) { + channelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-8)); + } + } } -ma_result ma_pcm_converter_set_output_sample_rate(ma_pcm_converter* pDSP, ma_uint32 sampleRateOut) +static void ma_get_standard_channel_map_sndio(ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) { - if (pDSP == NULL) { - return MA_INVALID_ARGS; - } + switch (channels) + { + case 1: + { + channelMap[0] = MA_CHANNEL_MONO; + } break; - /* Must have a sample rate of > 0. */ - if (sampleRateOut == 0) { - return MA_INVALID_ARGS; - } + case 2: + { + channelMap[0] = MA_CHANNEL_LEFT; + channelMap[1] = MA_CHANNEL_RIGHT; + } break; - /* Must have been initialized with allowDynamicSampleRate. */ - if (!pDSP->isDynamicSampleRateAllowed) { - return MA_INVALID_OPERATION; - } + case 3: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_FRONT_CENTER; + } break; - ma_atomic_exchange_32(&pDSP->src.config.sampleRateOut, sampleRateOut); - return ma_pcm_converter_refresh_sample_rate(pDSP); -} + case 4: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_BACK_LEFT; + channelMap[3] = MA_CHANNEL_BACK_RIGHT; + } break; -ma_result ma_pcm_converter_set_sample_rate(ma_pcm_converter* pDSP, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut) -{ - if (pDSP == NULL) { - return MA_INVALID_ARGS; - } + case 5: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_BACK_LEFT; + channelMap[3] = MA_CHANNEL_BACK_RIGHT; + channelMap[4] = MA_CHANNEL_FRONT_CENTER; + } break; - /* Must have a sample rate of > 0. */ - if (sampleRateIn == 0 || sampleRateOut == 0) { - return MA_INVALID_ARGS; + case 6: + default: + { + channelMap[0] = MA_CHANNEL_FRONT_LEFT; + channelMap[1] = MA_CHANNEL_FRONT_RIGHT; + channelMap[2] = MA_CHANNEL_BACK_LEFT; + channelMap[3] = MA_CHANNEL_BACK_RIGHT; + channelMap[4] = MA_CHANNEL_FRONT_CENTER; + channelMap[5] = MA_CHANNEL_LFE; + } break; } - /* Must have been initialized with allowDynamicSampleRate. */ - if (!pDSP->isDynamicSampleRateAllowed) { - return MA_INVALID_OPERATION; + /* Remainder. */ + if (channels > 6) { + ma_uint32 iChannel; + for (iChannel = 6; iChannel < MA_MAX_CHANNELS; ++iChannel) { + channelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-6)); + } } - - ma_atomic_exchange_32(&pDSP->src.config.sampleRateIn, sampleRateIn); - ma_atomic_exchange_32(&pDSP->src.config.sampleRateOut, sampleRateOut); - - return ma_pcm_converter_refresh_sample_rate(pDSP); } -ma_uint64 ma_pcm_converter_read(ma_pcm_converter* pDSP, void* pFramesOut, ma_uint64 frameCount) +void ma_get_standard_channel_map(ma_standard_channel_map standardChannelMap, ma_uint32 channels, ma_channel channelMap[MA_MAX_CHANNELS]) { - ma_pcm_converter_callback_data data; + switch (standardChannelMap) + { + case ma_standard_channel_map_alsa: + { + ma_get_standard_channel_map_alsa(channels, channelMap); + } break; - if (pDSP == NULL || pFramesOut == NULL) { - return 0; - } + case ma_standard_channel_map_rfc3551: + { + ma_get_standard_channel_map_rfc3551(channels, channelMap); + } break; - /* Fast path. */ - if (pDSP->isPassthrough) { - if (frameCount <= 0xFFFFFFFF) { - return (ma_uint32)pDSP->onRead(pDSP, pFramesOut, (ma_uint32)frameCount, pDSP->pUserData); - } else { - ma_uint8* pNextFramesOut = (ma_uint8*)pFramesOut; - - ma_uint64 totalFramesRead = 0; - while (totalFramesRead < frameCount) { - ma_uint32 framesRead; - ma_uint64 framesRemaining = (frameCount - totalFramesRead); - ma_uint64 framesToReadRightNow = framesRemaining; - if (framesToReadRightNow > 0xFFFFFFFF) { - framesToReadRightNow = 0xFFFFFFFF; - } + case ma_standard_channel_map_flac: + { + ma_get_standard_channel_map_flac(channels, channelMap); + } break; - framesRead = pDSP->onRead(pDSP, pNextFramesOut, (ma_uint32)framesToReadRightNow, pDSP->pUserData); - if (framesRead == 0) { - break; - } + case ma_standard_channel_map_vorbis: + { + ma_get_standard_channel_map_vorbis(channels, channelMap); + } break; - pNextFramesOut += framesRead * pDSP->channelRouter.config.channelsOut * ma_get_bytes_per_sample(pDSP->formatConverterOut.config.formatOut); - totalFramesRead += framesRead; - } + case ma_standard_channel_map_sound4: + { + ma_get_standard_channel_map_sound4(channels, channelMap); + } break; + + case ma_standard_channel_map_sndio: + { + ma_get_standard_channel_map_sndio(channels, channelMap); + } break; - return totalFramesRead; - } + case ma_standard_channel_map_microsoft: + default: + { + ma_get_standard_channel_map_microsoft(channels, channelMap); + } break; } - - /* Slower path. The real work is done here. To do this all we need to do is read from the last stage in the pipeline. */ - ma_assert(pDSP->isPostFormatConversionRequired == MA_TRUE); - - data.pDSP = pDSP; - data.pUserDataForClient = pDSP->pUserData; - return ma_format_converter_read(&pDSP->formatConverterOut, frameCount, pFramesOut, &data); } - -typedef struct +void ma_channel_map_copy(ma_channel* pOut, const ma_channel* pIn, ma_uint32 channels) { - const void* pDataIn; - ma_format formatIn; - ma_uint32 channelsIn; - ma_uint64 totalFrameCount; - ma_uint64 iNextFrame; - ma_bool32 isFeedingZeros; /* When set to true, feeds the DSP zero samples. */ -} ma_convert_frames__data; + if (pOut != NULL && pIn != NULL && channels > 0) { + MA_COPY_MEMORY(pOut, pIn, sizeof(*pOut) * channels); + } +} -ma_uint32 ma_convert_frames__on_read(ma_pcm_converter* pDSP, void* pFramesOut, ma_uint32 frameCount, void* pUserData) +ma_bool32 ma_channel_map_valid(ma_uint32 channels, const ma_channel channelMap[MA_MAX_CHANNELS]) { - ma_convert_frames__data* pData; - ma_uint32 framesToRead; - ma_uint64 framesRemaining; - ma_uint32 frameSizeInBytes; - - (void)pDSP; - - pData = (ma_convert_frames__data*)pUserData; - ma_assert(pData != NULL); - ma_assert(pData->totalFrameCount >= pData->iNextFrame); - - framesToRead = frameCount; - framesRemaining = (pData->totalFrameCount - pData->iNextFrame); - if (framesToRead > framesRemaining) { - framesToRead = (ma_uint32)framesRemaining; + if (channelMap == NULL) { + return MA_FALSE; } - frameSizeInBytes = ma_get_bytes_per_frame(pData->formatIn, pData->channelsIn); + /* A channel count of 0 is invalid. */ + if (channels == 0) { + return MA_FALSE; + } - if (!pData->isFeedingZeros) { - ma_copy_memory(pFramesOut, (const ma_uint8*)pData->pDataIn + (frameSizeInBytes * pData->iNextFrame), frameSizeInBytes * framesToRead); - } else { - ma_zero_memory(pFramesOut, frameSizeInBytes * framesToRead); + /* It does not make sense to have a mono channel when there is more than 1 channel. */ + if (channels > 1) { + ma_uint32 iChannel; + for (iChannel = 0; iChannel < channels; ++iChannel) { + if (channelMap[iChannel] == MA_CHANNEL_MONO) { + return MA_FALSE; + } + } } - pData->iNextFrame += framesToRead; - return framesToRead; + return MA_TRUE; } -ma_pcm_converter_config ma_pcm_converter_config_init_new() +ma_bool32 ma_channel_map_equal(ma_uint32 channels, const ma_channel channelMapA[MA_MAX_CHANNELS], const ma_channel channelMapB[MA_MAX_CHANNELS]) { - ma_pcm_converter_config config; - ma_zero_object(&config); - - return config; -} + ma_uint32 iChannel; -ma_pcm_converter_config ma_pcm_converter_config_init(ma_format formatIn, ma_uint32 channelsIn, ma_uint32 sampleRateIn, ma_format formatOut, ma_uint32 channelsOut, ma_uint32 sampleRateOut, ma_pcm_converter_read_proc onRead, void* pUserData) -{ - return ma_pcm_converter_config_init_ex(formatIn, channelsIn, sampleRateIn, NULL, formatOut, channelsOut, sampleRateOut, NULL, onRead, pUserData); -} + if (channelMapA == channelMapB) { + return MA_FALSE; + } -ma_pcm_converter_config ma_pcm_converter_config_init_ex(ma_format formatIn, ma_uint32 channelsIn, ma_uint32 sampleRateIn, ma_channel channelMapIn[MA_MAX_CHANNELS], ma_format formatOut, ma_uint32 channelsOut, ma_uint32 sampleRateOut, ma_channel channelMapOut[MA_MAX_CHANNELS], ma_pcm_converter_read_proc onRead, void* pUserData) -{ - ma_pcm_converter_config config; - ma_zero_object(&config); - config.formatIn = formatIn; - config.channelsIn = channelsIn; - config.sampleRateIn = sampleRateIn; - config.formatOut = formatOut; - config.channelsOut = channelsOut; - config.sampleRateOut = sampleRateOut; - if (channelMapIn != NULL) { - ma_copy_memory(config.channelMapIn, channelMapIn, sizeof(config.channelMapIn)); + if (channels == 0 || channels > MA_MAX_CHANNELS) { + return MA_FALSE; } - if (channelMapOut != NULL) { - ma_copy_memory(config.channelMapOut, channelMapOut, sizeof(config.channelMapOut)); + + for (iChannel = 0; iChannel < channels; ++iChannel) { + if (channelMapA[iChannel] != channelMapB[iChannel]) { + return MA_FALSE; + } } - config.onRead = onRead; - config.pUserData = pUserData; - return config; + return MA_TRUE; } - - -ma_uint64 ma_convert_frames(void* pOut, ma_format formatOut, ma_uint32 channelsOut, ma_uint32 sampleRateOut, const void* pIn, ma_format formatIn, ma_uint32 channelsIn, ma_uint32 sampleRateIn, ma_uint64 frameCount) +ma_bool32 ma_channel_map_blank(ma_uint32 channels, const ma_channel channelMap[MA_MAX_CHANNELS]) { - ma_channel channelMapOut[MA_MAX_CHANNELS]; - ma_channel channelMapIn[MA_MAX_CHANNELS]; + ma_uint32 iChannel; - ma_get_standard_channel_map(ma_standard_channel_map_default, channelsOut, channelMapOut); - ma_get_standard_channel_map(ma_standard_channel_map_default, channelsIn, channelMapIn); + for (iChannel = 0; iChannel < channels; ++iChannel) { + if (channelMap[iChannel] != MA_CHANNEL_NONE) { + return MA_FALSE; + } + } - return ma_convert_frames_ex(pOut, formatOut, channelsOut, sampleRateOut, channelMapOut, pIn, formatIn, channelsIn, sampleRateIn, channelMapIn, frameCount); + return MA_TRUE; } -ma_uint64 ma_convert_frames_ex(void* pOut, ma_format formatOut, ma_uint32 channelsOut, ma_uint32 sampleRateOut, ma_channel channelMapOut[MA_MAX_CHANNELS], const void* pIn, ma_format formatIn, ma_uint32 channelsIn, ma_uint32 sampleRateIn, ma_channel channelMapIn[MA_MAX_CHANNELS], ma_uint64 frameCount) +ma_bool32 ma_channel_map_contains_channel_position(ma_uint32 channels, const ma_channel channelMap[MA_MAX_CHANNELS], ma_channel channelPosition) { - ma_uint64 frameCountOut; - ma_convert_frames__data data; - ma_pcm_converter_config converterConfig; - ma_pcm_converter converter; - ma_uint64 totalFramesRead; - - if (frameCount == 0) { - return 0; - } - - frameCountOut = ma_calculate_frame_count_after_src(sampleRateOut, sampleRateIn, frameCount); - if (pOut == NULL) { - return frameCountOut; + ma_uint32 iChannel; + for (iChannel = 0; iChannel < channels; ++iChannel) { + if (channelMap[iChannel] == channelPosition) { + return MA_TRUE; + } } - data.pDataIn = pIn; - data.formatIn = formatIn; - data.channelsIn = channelsIn; - data.totalFrameCount = frameCount; - data.iNextFrame = 0; - data.isFeedingZeros = MA_FALSE; + return MA_FALSE; +} - ma_zero_object(&converterConfig); - converterConfig.formatIn = formatIn; - converterConfig.channelsIn = channelsIn; - converterConfig.sampleRateIn = sampleRateIn; - if (channelMapIn != NULL) { - ma_channel_map_copy(converterConfig.channelMapIn, channelMapIn, channelsIn); - } else { - ma_get_standard_channel_map(ma_standard_channel_map_default, converterConfig.channelsIn, converterConfig.channelMapIn); - } - - converterConfig.formatOut = formatOut; - converterConfig.channelsOut = channelsOut; - converterConfig.sampleRateOut = sampleRateOut; - if (channelMapOut != NULL) { - ma_channel_map_copy(converterConfig.channelMapOut, channelMapOut, channelsOut); - } else { - ma_get_standard_channel_map(ma_standard_channel_map_default, converterConfig.channelsOut, converterConfig.channelMapOut); - } - converterConfig.onRead = ma_convert_frames__on_read; - converterConfig.pUserData = &data; +/************************************************************************************************************************************************************** - if (ma_pcm_converter_init(&converterConfig, &converter) != MA_SUCCESS) { - return 0; - } +Conversion Helpers - /* - Always output our computed frame count. There is a chance the sample rate conversion routine may not output the last sample - due to precision issues with 32-bit floats, in which case we should feed the DSP zero samples so it can generate that last - frame. - */ - totalFramesRead = ma_pcm_converter_read(&converter, pOut, frameCountOut); - if (totalFramesRead < frameCountOut) { - ma_uint32 bpfOut = ma_get_bytes_per_frame(formatOut, channelsOut); +**************************************************************************************************************************************************************/ +ma_uint64 ma_convert_frames(void* pOut, ma_uint64 frameCountOut, ma_format formatOut, ma_uint32 channelsOut, ma_uint32 sampleRateOut, const void* pIn, ma_uint64 frameCountIn, ma_format formatIn, ma_uint32 channelsIn, ma_uint32 sampleRateIn) +{ + ma_data_converter_config config; - data.isFeedingZeros = MA_TRUE; - data.totalFrameCount = ((ma_uint64)0xFFFFFFFF << 32) | 0xFFFFFFFF; /* C89 does not support 64-bit constants so need to instead construct it like this. Annoying... */ /*data.totalFrameCount = 0xFFFFFFFFFFFFFFFF;*/ - data.pDataIn = NULL; + config = ma_data_converter_config_init(formatIn, formatOut, channelsIn, channelsOut, sampleRateIn, sampleRateOut); + ma_get_standard_channel_map(ma_standard_channel_map_default, channelsOut, config.channelMapOut); + ma_get_standard_channel_map(ma_standard_channel_map_default, channelsIn, config.channelMapIn); + config.resampling.linear.lpfCount = ma_min(MA_DEFAULT_RESAMPLER_LPF_FILTERS, MA_MAX_RESAMPLER_LPF_FILTERS); - while (totalFramesRead < frameCountOut) { - ma_uint64 framesToRead; - ma_uint64 framesJustRead; + return ma_convert_frames_ex(pOut, frameCountOut, pIn, frameCountIn, &config); +} - framesToRead = (frameCountOut - totalFramesRead); - ma_assert(framesToRead > 0); +ma_uint64 ma_convert_frames_ex(void* pOut, ma_uint64 frameCountOut, const void* pIn, ma_uint64 frameCountIn, const ma_data_converter_config* pConfig) +{ + ma_result result; + ma_data_converter converter; - framesJustRead = ma_pcm_converter_read(&converter, ma_offset_ptr(pOut, totalFramesRead * bpfOut), framesToRead); - totalFramesRead += framesJustRead; + if (frameCountIn == 0 || pConfig == NULL) { + return 0; + } - if (framesJustRead < framesToRead) { - break; - } - } + result = ma_data_converter_init(pConfig, &converter); + if (result != MA_SUCCESS) { + return 0; /* Failed to initialize the data converter. */ + } - /* At this point we should have output every sample, but just to be super duper sure, just fill the rest with zeros. */ - if (totalFramesRead < frameCountOut) { - ma_zero_memory_64(ma_offset_ptr(pOut, totalFramesRead * bpfOut), ((frameCountOut - totalFramesRead) * bpfOut)); - totalFramesRead = frameCountOut; + if (pOut == NULL) { + frameCountOut = ma_data_converter_get_expected_output_frame_count(&converter, frameCountIn); + } else { + result = ma_data_converter_process_pcm_frames(&converter, pIn, &frameCountIn, pOut, &frameCountOut); + if (result != MA_SUCCESS) { + frameCountOut = 0; } } - ma_assert(totalFramesRead == frameCountOut); - return totalFramesRead; + ma_data_converter_uninit(&converter); + return frameCountOut; } @@ -32714,13 +35351,13 @@ MA_INLINE ma_uint32 ma_rb__extract_offset_loop_flag(ma_uint32 encodedOffset) MA_INLINE void* ma_rb__get_read_ptr(ma_rb* pRB) { - ma_assert(pRB != NULL); + MA_ASSERT(pRB != NULL); return ma_offset_ptr(pRB->pBuffer, ma_rb__extract_offset_in_bytes(pRB->encodedReadOffset)); } MA_INLINE void* ma_rb__get_write_ptr(ma_rb* pRB) { - ma_assert(pRB != NULL); + MA_ASSERT(pRB != NULL); return ma_offset_ptr(pRB->pBuffer, ma_rb__extract_offset_in_bytes(pRB->encodedWriteOffset)); } @@ -32731,16 +35368,17 @@ MA_INLINE ma_uint32 ma_rb__construct_offset(ma_uint32 offsetInBytes, ma_uint32 o MA_INLINE void ma_rb__deconstruct_offset(ma_uint32 encodedOffset, ma_uint32* pOffsetInBytes, ma_uint32* pOffsetLoopFlag) { - ma_assert(pOffsetInBytes != NULL); - ma_assert(pOffsetLoopFlag != NULL); + MA_ASSERT(pOffsetInBytes != NULL); + MA_ASSERT(pOffsetLoopFlag != NULL); *pOffsetInBytes = ma_rb__extract_offset_in_bytes(encodedOffset); *pOffsetLoopFlag = ma_rb__extract_offset_loop_flag(encodedOffset); } -ma_result ma_rb_init_ex(size_t subbufferSizeInBytes, size_t subbufferCount, size_t subbufferStrideInBytes, void* pOptionalPreallocatedBuffer, ma_rb* pRB) +ma_result ma_rb_init_ex(size_t subbufferSizeInBytes, size_t subbufferCount, size_t subbufferStrideInBytes, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_rb* pRB) { + ma_result result; const ma_uint32 maxSubBufferSize = 0x7FFFFFFF - (MA_SIMD_ALIGNMENT-1); if (pRB == NULL) { @@ -32756,7 +35394,13 @@ ma_result ma_rb_init_ex(size_t subbufferSizeInBytes, size_t subbufferCount, size } - ma_zero_object(pRB); + MA_ZERO_OBJECT(pRB); + + result = ma_allocation_callbacks_init_copy(&pRB->allocationCallbacks, pAllocationCallbacks); + if (result != MA_SUCCESS) { + return result; + } + pRB->subbufferSizeInBytes = (ma_uint32)subbufferSizeInBytes; pRB->subbufferCount = (ma_uint32)subbufferCount; @@ -32773,21 +35417,21 @@ ma_result ma_rb_init_ex(size_t subbufferSizeInBytes, size_t subbufferCount, size pRB->subbufferStrideInBytes = (pRB->subbufferSizeInBytes + (MA_SIMD_ALIGNMENT-1)) & ~MA_SIMD_ALIGNMENT; bufferSizeInBytes = (size_t)pRB->subbufferCount*pRB->subbufferStrideInBytes; - pRB->pBuffer = ma_aligned_malloc(bufferSizeInBytes, MA_SIMD_ALIGNMENT); + pRB->pBuffer = ma_aligned_malloc(bufferSizeInBytes, MA_SIMD_ALIGNMENT, &pRB->allocationCallbacks); if (pRB->pBuffer == NULL) { return MA_OUT_OF_MEMORY; } - ma_zero_memory(pRB->pBuffer, bufferSizeInBytes); + MA_ZERO_MEMORY(pRB->pBuffer, bufferSizeInBytes); pRB->ownsBuffer = MA_TRUE; } return MA_SUCCESS; } -ma_result ma_rb_init(size_t bufferSizeInBytes, void* pOptionalPreallocatedBuffer, ma_rb* pRB) +ma_result ma_rb_init(size_t bufferSizeInBytes, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_rb* pRB) { - return ma_rb_init_ex(bufferSizeInBytes, 1, 0, pOptionalPreallocatedBuffer, pRB); + return ma_rb_init_ex(bufferSizeInBytes, 1, 0, pOptionalPreallocatedBuffer, pAllocationCallbacks, pRB); } void ma_rb_uninit(ma_rb* pRB) @@ -32797,7 +35441,7 @@ void ma_rb_uninit(ma_rb* pRB) } if (pRB->ownsBuffer) { - ma_aligned_free(pRB->pBuffer); + ma_aligned_free(pRB->pBuffer, &pRB->allocationCallbacks); } } @@ -32934,7 +35578,7 @@ ma_result ma_rb_acquire_write(ma_rb* pRB, size_t* pSizeInBytes, void** ppBufferO /* Clear the buffer if desired. */ if (pRB->clearOnWriteAcquire) { - ma_zero_memory(*ppBufferOut, *pSizeInBytes); + MA_ZERO_MEMORY(*ppBufferOut, *pSizeInBytes); } return MA_SUCCESS; @@ -33161,12 +35805,12 @@ void* ma_rb_get_subbuffer_ptr(ma_rb* pRB, size_t subbufferIndex, void* pBuffer) static MA_INLINE ma_uint32 ma_pcm_rb_get_bpf(ma_pcm_rb* pRB) { - ma_assert(pRB != NULL); + MA_ASSERT(pRB != NULL); return ma_get_bytes_per_frame(pRB->format, pRB->channels); } -ma_result ma_pcm_rb_init_ex(ma_format format, ma_uint32 channels, ma_uint32 subbufferSizeInFrames, ma_uint32 subbufferCount, ma_uint32 subbufferStrideInFrames, void* pOptionalPreallocatedBuffer, ma_pcm_rb* pRB) +ma_result ma_pcm_rb_init_ex(ma_format format, ma_uint32 channels, ma_uint32 subbufferSizeInFrames, ma_uint32 subbufferCount, ma_uint32 subbufferStrideInFrames, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_pcm_rb* pRB) { ma_uint32 bpf; ma_result result; @@ -33175,14 +35819,14 @@ ma_result ma_pcm_rb_init_ex(ma_format format, ma_uint32 channels, ma_uint32 subb return MA_INVALID_ARGS; } - ma_zero_object(pRB); + MA_ZERO_OBJECT(pRB); bpf = ma_get_bytes_per_frame(format, channels); if (bpf == 0) { return MA_INVALID_ARGS; } - result = ma_rb_init_ex(subbufferSizeInFrames*bpf, subbufferCount, subbufferStrideInFrames*bpf, pOptionalPreallocatedBuffer, &pRB->rb); + result = ma_rb_init_ex(subbufferSizeInFrames*bpf, subbufferCount, subbufferStrideInFrames*bpf, pOptionalPreallocatedBuffer, pAllocationCallbacks, &pRB->rb); if (result != MA_SUCCESS) { return result; } @@ -33193,9 +35837,9 @@ ma_result ma_pcm_rb_init_ex(ma_format format, ma_uint32 channels, ma_uint32 subb return MA_SUCCESS; } -ma_result ma_pcm_rb_init(ma_format format, ma_uint32 channels, ma_uint32 bufferSizeInFrames, void* pOptionalPreallocatedBuffer, ma_pcm_rb* pRB) +ma_result ma_pcm_rb_init(ma_format format, ma_uint32 channels, ma_uint32 bufferSizeInFrames, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_pcm_rb* pRB) { - return ma_pcm_rb_init_ex(format, channels, bufferSizeInFrames, 1, 0, pOptionalPreallocatedBuffer, pRB); + return ma_pcm_rb_init_ex(format, channels, bufferSizeInFrames, 1, 0, pOptionalPreallocatedBuffer, pAllocationCallbacks, pRB); } void ma_pcm_rb_uninit(ma_pcm_rb* pRB) @@ -33362,22 +36006,38 @@ void* ma_pcm_rb_get_subbuffer_ptr(ma_pcm_rb* pRB, ma_uint32 subbufferIndex, void Miscellaneous Helpers **************************************************************************************************************************************************************/ -void* ma_malloc(size_t sz) +void* ma_malloc(size_t sz, const ma_allocation_callbacks* pAllocationCallbacks) { - return MA_MALLOC(sz); + if (pAllocationCallbacks != NULL) { + return ma__malloc_from_callbacks(sz, pAllocationCallbacks); + } else { + return ma__malloc_default(sz, NULL); + } } -void* ma_realloc(void* p, size_t sz) +void* ma_realloc(void* p, size_t sz, const ma_allocation_callbacks* pAllocationCallbacks) { - return MA_REALLOC(p, sz); + if (pAllocationCallbacks != NULL) { + if (pAllocationCallbacks->onRealloc != NULL) { + return pAllocationCallbacks->onRealloc(p, sz, pAllocationCallbacks->pUserData); + } else { + return NULL; /* This requires a native implementation of realloc(). */ + } + } else { + return ma__realloc_default(p, sz, NULL); + } } -void ma_free(void* p) +void ma_free(void* p, const ma_allocation_callbacks* pAllocationCallbacks) { - MA_FREE(p); + if (pAllocationCallbacks != NULL) { + ma__free_from_callbacks(p, pAllocationCallbacks); + } else { + ma__free_default(p, NULL); + } } -void* ma_aligned_malloc(size_t sz, size_t alignment) +void* ma_aligned_malloc(size_t sz, size_t alignment, const ma_allocation_callbacks* pAllocationCallbacks) { size_t extraBytes; void* pUnaligned; @@ -33389,7 +36049,7 @@ void* ma_aligned_malloc(size_t sz, size_t alignment) extraBytes = alignment-1 + sizeof(void*); - pUnaligned = ma_malloc(sz + extraBytes); + pUnaligned = ma_malloc(sz + extraBytes, pAllocationCallbacks); if (pUnaligned == NULL) { return NULL; } @@ -33400,9 +36060,9 @@ void* ma_aligned_malloc(size_t sz, size_t alignment) return pAligned; } -void ma_aligned_free(void* p) +void ma_aligned_free(void* p, const ma_allocation_callbacks* pAllocationCallbacks) { - ma_free(((void**)p)[-1]); + ma_free(((void**)p)[-1], pAllocationCallbacks); } const char* ma_get_format_name(ma_format format) @@ -33449,12 +36109,12 @@ Decoding **************************************************************************************************************************************************************/ #ifndef MA_NO_DECODING -size_t ma_decoder_read_bytes(ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead) +static size_t ma_decoder_read_bytes(ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead) { size_t bytesRead; - ma_assert(pDecoder != NULL); - ma_assert(pBufferOut != NULL); + MA_ASSERT(pDecoder != NULL); + MA_ASSERT(pBufferOut != NULL); bytesRead = pDecoder->onRead(pDecoder, pBufferOut, bytesToRead); pDecoder->readPointer += bytesRead; @@ -33462,11 +36122,11 @@ size_t ma_decoder_read_bytes(ma_decoder* pDecoder, void* pBufferOut, size_t byte return bytesRead; } -ma_bool32 ma_decoder_seek_bytes(ma_decoder* pDecoder, int byteOffset, ma_seek_origin origin) +static ma_bool32 ma_decoder_seek_bytes(ma_decoder* pDecoder, int byteOffset, ma_seek_origin origin) { ma_bool32 wasSuccessful; - ma_assert(pDecoder != NULL); + MA_ASSERT(pDecoder != NULL); wasSuccessful = pDecoder->onSeek(pDecoder, byteOffset, origin); if (wasSuccessful) { @@ -33480,49 +36140,18 @@ ma_bool32 ma_decoder_seek_bytes(ma_decoder* pDecoder, int byteOffset, ma_seek_or return wasSuccessful; } -ma_bool32 ma_decoder_seek_bytes_64(ma_decoder* pDecoder, ma_uint64 byteOffset, ma_seek_origin origin) -{ - ma_assert(pDecoder != NULL); - - if (origin == ma_seek_origin_start) { - ma_uint64 bytesToSeekThisIteration = 0x7FFFFFFF; - if (bytesToSeekThisIteration > byteOffset) { - bytesToSeekThisIteration = byteOffset; - } - - if (!ma_decoder_seek_bytes(pDecoder, (int)bytesToSeekThisIteration, ma_seek_origin_start)) { - return MA_FALSE; - } - - byteOffset -= bytesToSeekThisIteration; - } - - /* Getting here means we need to seek relative to the current position. */ - while (byteOffset > 0) { - ma_uint64 bytesToSeekThisIteration = 0x7FFFFFFF; - if (bytesToSeekThisIteration > byteOffset) { - bytesToSeekThisIteration = byteOffset; - } - - if (!ma_decoder_seek_bytes(pDecoder, (int)bytesToSeekThisIteration, ma_seek_origin_current)) { - return MA_FALSE; - } - - byteOffset -= bytesToSeekThisIteration; - } - - return MA_TRUE; -} - ma_decoder_config ma_decoder_config_init(ma_format outputFormat, ma_uint32 outputChannels, ma_uint32 outputSampleRate) { ma_decoder_config config; - ma_zero_object(&config); + MA_ZERO_OBJECT(&config); config.format = outputFormat; config.channels = outputChannels; config.sampleRate = outputSampleRate; ma_get_standard_channel_map(ma_standard_channel_map_default, config.channels, config.channelMap); + config.resampling.algorithm = ma_resample_algorithm_linear; + config.resampling.linear.lpfCount = ma_min(MA_DEFAULT_RESAMPLER_LPF_FILTERS, MA_MAX_RESAMPLER_LPF_FILTERS); + config.resampling.speex.quality = 3; return config; } @@ -33533,17 +36162,17 @@ ma_decoder_config ma_decoder_config_init_copy(const ma_decoder_config* pConfig) if (pConfig != NULL) { config = *pConfig; } else { - ma_zero_object(&config); + MA_ZERO_OBJECT(&config); } return config; } -ma_result ma_decoder__init_dsp(ma_decoder* pDecoder, const ma_decoder_config* pConfig, ma_pcm_converter_read_proc onRead) +static ma_result ma_decoder__init_data_converter(ma_decoder* pDecoder, const ma_decoder_config* pConfig) { - ma_pcm_converter_config dspConfig; + ma_data_converter_config converterConfig; - ma_assert(pDecoder != NULL); + MA_ASSERT(pDecoder != NULL); /* Output format. */ if (pConfig->format == ma_format_unknown) { @@ -33567,75 +36196,76 @@ ma_result ma_decoder__init_dsp(ma_decoder* pDecoder, const ma_decoder_config* pC if (ma_channel_map_blank(pDecoder->outputChannels, pConfig->channelMap)) { ma_get_standard_channel_map(ma_standard_channel_map_default, pDecoder->outputChannels, pDecoder->outputChannelMap); } else { - ma_copy_memory(pDecoder->outputChannelMap, pConfig->channelMap, sizeof(pConfig->channelMap)); + MA_COPY_MEMORY(pDecoder->outputChannelMap, pConfig->channelMap, sizeof(pConfig->channelMap)); } + + converterConfig = ma_data_converter_config_init( + pDecoder->internalFormat, pDecoder->outputFormat, + pDecoder->internalChannels, pDecoder->outputChannels, + pDecoder->internalSampleRate, pDecoder->outputSampleRate + ); + ma_channel_map_copy(converterConfig.channelMapIn, pDecoder->internalChannelMap, pDecoder->internalChannels); + ma_channel_map_copy(converterConfig.channelMapOut, pDecoder->outputChannelMap, pDecoder->outputChannels); + converterConfig.channelMixMode = pConfig->channelMixMode; + converterConfig.ditherMode = pConfig->ditherMode; + converterConfig.resampling.allowDynamicSampleRate = MA_FALSE; /* Never allow dynamic sample rate conversion. Setting this to true will disable passthrough optimizations. */ + converterConfig.resampling.algorithm = pConfig->resampling.algorithm; + converterConfig.resampling.linear.lpfCount = pConfig->resampling.linear.lpfCount; + converterConfig.resampling.speex.quality = pConfig->resampling.speex.quality; - /* DSP. */ - dspConfig = ma_pcm_converter_config_init_ex( - pDecoder->internalFormat, pDecoder->internalChannels, pDecoder->internalSampleRate, pDecoder->internalChannelMap, - pDecoder->outputFormat, pDecoder->outputChannels, pDecoder->outputSampleRate, pDecoder->outputChannelMap, - onRead, pDecoder); - dspConfig.channelMixMode = pConfig->channelMixMode; - dspConfig.ditherMode = pConfig->ditherMode; - dspConfig.srcAlgorithm = pConfig->srcAlgorithm; - dspConfig.sinc = pConfig->src.sinc; - - return ma_pcm_converter_init(&dspConfig, &pDecoder->dsp); + return ma_data_converter_init(&converterConfig, &pDecoder->converter); } /* WAV */ #ifdef dr_wav_h #define MA_HAS_WAV -size_t ma_decoder_internal_on_read__wav(void* pUserData, void* pBufferOut, size_t bytesToRead) +static size_t ma_decoder_internal_on_read__wav(void* pUserData, void* pBufferOut, size_t bytesToRead) { ma_decoder* pDecoder = (ma_decoder*)pUserData; - ma_assert(pDecoder != NULL); + MA_ASSERT(pDecoder != NULL); return ma_decoder_read_bytes(pDecoder, pBufferOut, bytesToRead); } -drwav_bool32 ma_decoder_internal_on_seek__wav(void* pUserData, int offset, drwav_seek_origin origin) +static drwav_bool32 ma_decoder_internal_on_seek__wav(void* pUserData, int offset, drwav_seek_origin origin) { ma_decoder* pDecoder = (ma_decoder*)pUserData; - ma_assert(pDecoder != NULL); + MA_ASSERT(pDecoder != NULL); return ma_decoder_seek_bytes(pDecoder, offset, (origin == drwav_seek_origin_start) ? ma_seek_origin_start : ma_seek_origin_current); } -ma_uint32 ma_decoder_internal_on_read_pcm_frames__wav(ma_pcm_converter* pDSP, void* pFramesOut, ma_uint32 frameCount, void* pUserData) +static ma_uint64 ma_decoder_internal_on_read_pcm_frames__wav(ma_decoder* pDecoder, void* pFramesOut, ma_uint64 frameCount) { - ma_decoder* pDecoder; drwav* pWav; - (void)pDSP; - - pDecoder = (ma_decoder*)pUserData; - ma_assert(pDecoder != NULL); + MA_ASSERT(pDecoder != NULL); + MA_ASSERT(pFramesOut != NULL); pWav = (drwav*)pDecoder->pInternalDecoder; - ma_assert(pWav != NULL); + MA_ASSERT(pWav != NULL); switch (pDecoder->internalFormat) { - case ma_format_s16: return (ma_uint32)drwav_read_pcm_frames_s16(pWav, frameCount, (drwav_int16*)pFramesOut); - case ma_format_s32: return (ma_uint32)drwav_read_pcm_frames_s32(pWav, frameCount, (drwav_int32*)pFramesOut); - case ma_format_f32: return (ma_uint32)drwav_read_pcm_frames_f32(pWav, frameCount, (float*)pFramesOut); + case ma_format_s16: return drwav_read_pcm_frames_s16(pWav, frameCount, (drwav_int16*)pFramesOut); + case ma_format_s32: return drwav_read_pcm_frames_s32(pWav, frameCount, (drwav_int32*)pFramesOut); + case ma_format_f32: return drwav_read_pcm_frames_f32(pWav, frameCount, (float*)pFramesOut); default: break; } /* Should never get here. If we do, it means the internal format was not set correctly at initialization time. */ - ma_assert(MA_FALSE); + MA_ASSERT(MA_FALSE); return 0; } -ma_result ma_decoder_internal_on_seek_to_pcm_frame__wav(ma_decoder* pDecoder, ma_uint64 frameIndex) +static ma_result ma_decoder_internal_on_seek_to_pcm_frame__wav(ma_decoder* pDecoder, ma_uint64 frameIndex) { drwav* pWav; drwav_bool32 result; pWav = (drwav*)pDecoder->pInternalDecoder; - ma_assert(pWav != NULL); + MA_ASSERT(pWav != NULL); result = drwav_seek_to_pcm_frame(pWav, frameIndex); if (result) { @@ -33645,42 +36275,48 @@ ma_result ma_decoder_internal_on_seek_to_pcm_frame__wav(ma_decoder* pDecoder, ma } } -ma_result ma_decoder_internal_on_uninit__wav(ma_decoder* pDecoder) +static ma_result ma_decoder_internal_on_uninit__wav(ma_decoder* pDecoder) { drwav_uninit((drwav*)pDecoder->pInternalDecoder); - ma_free(pDecoder->pInternalDecoder); + ma__free_from_callbacks(pDecoder->pInternalDecoder, &pDecoder->allocationCallbacks); return MA_SUCCESS; } -ma_uint64 ma_decoder_internal_on_get_length_in_pcm_frames__wav(ma_decoder* pDecoder) +static ma_uint64 ma_decoder_internal_on_get_length_in_pcm_frames__wav(ma_decoder* pDecoder) { return ((drwav*)pDecoder->pInternalDecoder)->totalPCMFrameCount; } -ma_result ma_decoder_init_wav__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder) +static ma_result ma_decoder_init_wav__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder) { drwav* pWav; - ma_result result; + drwav_allocation_callbacks allocationCallbacks; - ma_assert(pConfig != NULL); - ma_assert(pDecoder != NULL); + MA_ASSERT(pConfig != NULL); + MA_ASSERT(pDecoder != NULL); - pWav = (drwav*)ma_malloc(sizeof(*pWav)); + pWav = (drwav*)ma__malloc_from_callbacks(sizeof(*pWav), &pDecoder->allocationCallbacks); if (pWav == NULL) { return MA_OUT_OF_MEMORY; } + allocationCallbacks.pUserData = pDecoder->allocationCallbacks.pUserData; + allocationCallbacks.onMalloc = pDecoder->allocationCallbacks.onMalloc; + allocationCallbacks.onRealloc = pDecoder->allocationCallbacks.onRealloc; + allocationCallbacks.onFree = pDecoder->allocationCallbacks.onFree; + /* Try opening the decoder first. */ - if (!drwav_init(pWav, ma_decoder_internal_on_read__wav, ma_decoder_internal_on_seek__wav, pDecoder, NULL)) { - ma_free(pWav); + if (!drwav_init(pWav, ma_decoder_internal_on_read__wav, ma_decoder_internal_on_seek__wav, pDecoder, &allocationCallbacks)) { + ma__free_from_callbacks(pWav, &pDecoder->allocationCallbacks); return MA_ERROR; } /* If we get here it means we successfully initialized the WAV decoder. We can now initialize the rest of the ma_decoder. */ - pDecoder->onSeekToPCMFrame = ma_decoder_internal_on_seek_to_pcm_frame__wav; - pDecoder->onUninit = ma_decoder_internal_on_uninit__wav; + pDecoder->onReadPCMFrames = ma_decoder_internal_on_read_pcm_frames__wav; + pDecoder->onSeekToPCMFrame = ma_decoder_internal_on_seek_to_pcm_frame__wav; + pDecoder->onUninit = ma_decoder_internal_on_uninit__wav; pDecoder->onGetLengthInPCMFrames = ma_decoder_internal_on_get_length_in_pcm_frames__wav; - pDecoder->pInternalDecoder = pWav; + pDecoder->pInternalDecoder = pWav; /* Try to be as optimal as possible for the internal format. If miniaudio does not support a format we will fall back to f32. */ pDecoder->internalFormat = ma_format_unknown; @@ -33720,69 +36356,59 @@ ma_result ma_decoder_init_wav__internal(const ma_decoder_config* pConfig, ma_dec pDecoder->internalSampleRate = pWav->sampleRate; ma_get_standard_channel_map(ma_standard_channel_map_microsoft, pDecoder->internalChannels, pDecoder->internalChannelMap); - result = ma_decoder__init_dsp(pDecoder, pConfig, ma_decoder_internal_on_read_pcm_frames__wav); - if (result != MA_SUCCESS) { - drwav_uninit(pWav); - ma_free(pWav); - return result; - } - return MA_SUCCESS; } -#endif +#endif /* dr_wav_h */ /* FLAC */ #ifdef dr_flac_h #define MA_HAS_FLAC -size_t ma_decoder_internal_on_read__flac(void* pUserData, void* pBufferOut, size_t bytesToRead) +static size_t ma_decoder_internal_on_read__flac(void* pUserData, void* pBufferOut, size_t bytesToRead) { ma_decoder* pDecoder = (ma_decoder*)pUserData; - ma_assert(pDecoder != NULL); + MA_ASSERT(pDecoder != NULL); return ma_decoder_read_bytes(pDecoder, pBufferOut, bytesToRead); } -drflac_bool32 ma_decoder_internal_on_seek__flac(void* pUserData, int offset, drflac_seek_origin origin) +static drflac_bool32 ma_decoder_internal_on_seek__flac(void* pUserData, int offset, drflac_seek_origin origin) { ma_decoder* pDecoder = (ma_decoder*)pUserData; - ma_assert(pDecoder != NULL); + MA_ASSERT(pDecoder != NULL); return ma_decoder_seek_bytes(pDecoder, offset, (origin == drflac_seek_origin_start) ? ma_seek_origin_start : ma_seek_origin_current); } -ma_uint32 ma_decoder_internal_on_read_pcm_frames__flac(ma_pcm_converter* pDSP, void* pFramesOut, ma_uint32 frameCount, void* pUserData) +static ma_uint64 ma_decoder_internal_on_read_pcm_frames__flac(ma_decoder* pDecoder, void* pFramesOut, ma_uint64 frameCount) { - ma_decoder* pDecoder; drflac* pFlac; - (void)pDSP; - - pDecoder = (ma_decoder*)pUserData; - ma_assert(pDecoder != NULL); + MA_ASSERT(pDecoder != NULL); + MA_ASSERT(pFramesOut != NULL); pFlac = (drflac*)pDecoder->pInternalDecoder; - ma_assert(pFlac != NULL); + MA_ASSERT(pFlac != NULL); switch (pDecoder->internalFormat) { - case ma_format_s16: return (ma_uint32)drflac_read_pcm_frames_s16(pFlac, frameCount, (drflac_int16*)pFramesOut); - case ma_format_s32: return (ma_uint32)drflac_read_pcm_frames_s32(pFlac, frameCount, (drflac_int32*)pFramesOut); - case ma_format_f32: return (ma_uint32)drflac_read_pcm_frames_f32(pFlac, frameCount, (float*)pFramesOut); + case ma_format_s16: return drflac_read_pcm_frames_s16(pFlac, frameCount, (drflac_int16*)pFramesOut); + case ma_format_s32: return drflac_read_pcm_frames_s32(pFlac, frameCount, (drflac_int32*)pFramesOut); + case ma_format_f32: return drflac_read_pcm_frames_f32(pFlac, frameCount, (float*)pFramesOut); default: break; } /* Should never get here. If we do, it means the internal format was not set correctly at initialization time. */ - ma_assert(MA_FALSE); + MA_ASSERT(MA_FALSE); return 0; } -ma_result ma_decoder_internal_on_seek_to_pcm_frame__flac(ma_decoder* pDecoder, ma_uint64 frameIndex) +static ma_result ma_decoder_internal_on_seek_to_pcm_frame__flac(ma_decoder* pDecoder, ma_uint64 frameIndex) { drflac* pFlac; drflac_bool32 result; pFlac = (drflac*)pDecoder->pInternalDecoder; - ma_assert(pFlac != NULL); + MA_ASSERT(pFlac != NULL); result = drflac_seek_to_pcm_frame(pFlac, frameIndex); if (result) { @@ -33792,36 +36418,42 @@ ma_result ma_decoder_internal_on_seek_to_pcm_frame__flac(ma_decoder* pDecoder, m } } -ma_result ma_decoder_internal_on_uninit__flac(ma_decoder* pDecoder) +static ma_result ma_decoder_internal_on_uninit__flac(ma_decoder* pDecoder) { drflac_close((drflac*)pDecoder->pInternalDecoder); return MA_SUCCESS; } -ma_uint64 ma_decoder_internal_on_get_length_in_pcm_frames__flac(ma_decoder* pDecoder) +static ma_uint64 ma_decoder_internal_on_get_length_in_pcm_frames__flac(ma_decoder* pDecoder) { return ((drflac*)pDecoder->pInternalDecoder)->totalPCMFrameCount; } -ma_result ma_decoder_init_flac__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder) +static ma_result ma_decoder_init_flac__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder) { drflac* pFlac; - ma_result result; + drflac_allocation_callbacks allocationCallbacks; - ma_assert(pConfig != NULL); - ma_assert(pDecoder != NULL); + MA_ASSERT(pConfig != NULL); + MA_ASSERT(pDecoder != NULL); + + allocationCallbacks.pUserData = pDecoder->allocationCallbacks.pUserData; + allocationCallbacks.onMalloc = pDecoder->allocationCallbacks.onMalloc; + allocationCallbacks.onRealloc = pDecoder->allocationCallbacks.onRealloc; + allocationCallbacks.onFree = pDecoder->allocationCallbacks.onFree; /* Try opening the decoder first. */ - pFlac = drflac_open(ma_decoder_internal_on_read__flac, ma_decoder_internal_on_seek__flac, pDecoder, NULL); + pFlac = drflac_open(ma_decoder_internal_on_read__flac, ma_decoder_internal_on_seek__flac, pDecoder, &allocationCallbacks); if (pFlac == NULL) { return MA_ERROR; } /* If we get here it means we successfully initialized the FLAC decoder. We can now initialize the rest of the ma_decoder. */ - pDecoder->onSeekToPCMFrame = ma_decoder_internal_on_seek_to_pcm_frame__flac; - pDecoder->onUninit = ma_decoder_internal_on_uninit__flac; + pDecoder->onReadPCMFrames = ma_decoder_internal_on_read_pcm_frames__flac; + pDecoder->onSeekToPCMFrame = ma_decoder_internal_on_seek_to_pcm_frame__flac; + pDecoder->onUninit = ma_decoder_internal_on_uninit__flac; pDecoder->onGetLengthInPCMFrames = ma_decoder_internal_on_get_length_in_pcm_frames__flac; - pDecoder->pInternalDecoder = pFlac; + pDecoder->pInternalDecoder = pFlac; /* dr_flac supports reading as s32, s16 and f32. Try to do a one-to-one mapping if possible, but fall back to s32 if not. s32 is the "native" FLAC format @@ -33838,15 +36470,9 @@ ma_result ma_decoder_init_flac__internal(const ma_decoder_config* pConfig, ma_de pDecoder->internalSampleRate = pFlac->sampleRate; ma_get_standard_channel_map(ma_standard_channel_map_flac, pDecoder->internalChannels, pDecoder->internalChannelMap); - result = ma_decoder__init_dsp(pDecoder, pConfig, ma_decoder_internal_on_read_pcm_frames__flac); - if (result != MA_SUCCESS) { - drflac_close(pFlac); - return result; - } - return MA_SUCCESS; } -#endif +#endif /* dr_flac_h */ /* Vorbis */ #ifdef STB_VORBIS_INCLUDE_STB_VORBIS_H @@ -33866,13 +36492,13 @@ typedef struct float** ppPacketData; } ma_vorbis_decoder; -ma_uint32 ma_vorbis_decoder_read_pcm_frames(ma_vorbis_decoder* pVorbis, ma_decoder* pDecoder, void* pFramesOut, ma_uint32 frameCount) +static ma_uint64 ma_vorbis_decoder_read_pcm_frames(ma_vorbis_decoder* pVorbis, ma_decoder* pDecoder, void* pFramesOut, ma_uint64 frameCount) { float* pFramesOutF; - ma_uint32 totalFramesRead; + ma_uint64 totalFramesRead; - ma_assert(pVorbis != NULL); - ma_assert(pDecoder != NULL); + MA_ASSERT(pVorbis != NULL); + MA_ASSERT(pDecoder != NULL); pFramesOutF = (float*)pFramesOut; @@ -33886,17 +36512,17 @@ ma_uint32 ma_vorbis_decoder_read_pcm_frames(ma_vorbis_decoder* pVorbis, ma_decod pFramesOutF += 1; } - pVorbis->framesConsumed += 1; + pVorbis->framesConsumed += 1; pVorbis->framesRemaining -= 1; - frameCount -= 1; - totalFramesRead += 1; + frameCount -= 1; + totalFramesRead += 1; } if (frameCount == 0) { break; } - ma_assert(pVorbis->framesRemaining == 0); + MA_ASSERT(pVorbis->framesRemaining == 0); /* We've run out of cached frames, so decode the next packet and continue iteration. */ do @@ -33926,10 +36552,11 @@ ma_uint32 ma_vorbis_decoder_read_pcm_frames(ma_vorbis_decoder* pVorbis, ma_decod size_t bytesRead; if (pVorbis->dataCapacity == pVorbis->dataSize) { /* No room. Expand. */ + size_t oldCap = pVorbis->dataCapacity; size_t newCap = pVorbis->dataCapacity + MA_VORBIS_DATA_CHUNK_SIZE; ma_uint8* pNewData; - pNewData = (ma_uint8*)ma_realloc(pVorbis->pData, newCap); + pNewData = (ma_uint8*)ma__realloc_from_callbacks(pVorbis->pData, newCap, oldCap, &pDecoder->allocationCallbacks); if (pNewData == NULL) { return totalFramesRead; /* Out of memory. */ } @@ -33952,12 +36579,12 @@ ma_uint32 ma_vorbis_decoder_read_pcm_frames(ma_vorbis_decoder* pVorbis, ma_decod return totalFramesRead; } -ma_result ma_vorbis_decoder_seek_to_pcm_frame(ma_vorbis_decoder* pVorbis, ma_decoder* pDecoder, ma_uint64 frameIndex) +static ma_result ma_vorbis_decoder_seek_to_pcm_frame(ma_vorbis_decoder* pVorbis, ma_decoder* pDecoder, ma_uint64 frameIndex) { float buffer[4096]; - ma_assert(pVorbis != NULL); - ma_assert(pDecoder != NULL); + MA_ASSERT(pVorbis != NULL); + MA_ASSERT(pDecoder != NULL); /* This is terribly inefficient because stb_vorbis does not have a good seeking solution with it's push API. Currently this just performs @@ -33969,9 +36596,9 @@ ma_result ma_vorbis_decoder_seek_to_pcm_frame(ma_vorbis_decoder* pVorbis, ma_dec } stb_vorbis_flush_pushdata(pVorbis->pInternalVorbis); - pVorbis->framesConsumed = 0; + pVorbis->framesConsumed = 0; pVorbis->framesRemaining = 0; - pVorbis->dataSize = 0; + pVorbis->dataSize = 0; while (frameIndex > 0) { ma_uint32 framesRead; @@ -33980,7 +36607,7 @@ ma_result ma_vorbis_decoder_seek_to_pcm_frame(ma_vorbis_decoder* pVorbis, ma_dec framesToRead = (ma_uint32)frameIndex; } - framesRead = ma_vorbis_decoder_read_pcm_frames(pVorbis, pDecoder, buffer, framesToRead); + framesRead = (ma_uint32)ma_vorbis_decoder_read_pcm_frames(pVorbis, pDecoder, buffer, framesToRead); if (framesRead == 0) { return MA_ERROR; } @@ -33992,53 +36619,49 @@ ma_result ma_vorbis_decoder_seek_to_pcm_frame(ma_vorbis_decoder* pVorbis, ma_dec } -ma_result ma_decoder_internal_on_seek_to_pcm_frame__vorbis(ma_decoder* pDecoder, ma_uint64 frameIndex) +static ma_result ma_decoder_internal_on_seek_to_pcm_frame__vorbis(ma_decoder* pDecoder, ma_uint64 frameIndex) { ma_vorbis_decoder* pVorbis = (ma_vorbis_decoder*)pDecoder->pInternalDecoder; - ma_assert(pVorbis != NULL); + MA_ASSERT(pVorbis != NULL); return ma_vorbis_decoder_seek_to_pcm_frame(pVorbis, pDecoder, frameIndex); } -ma_result ma_decoder_internal_on_uninit__vorbis(ma_decoder* pDecoder) +static ma_result ma_decoder_internal_on_uninit__vorbis(ma_decoder* pDecoder) { ma_vorbis_decoder* pVorbis = (ma_vorbis_decoder*)pDecoder->pInternalDecoder; - ma_assert(pVorbis != NULL); + MA_ASSERT(pVorbis != NULL); stb_vorbis_close(pVorbis->pInternalVorbis); - ma_free(pVorbis->pData); - ma_free(pVorbis); + ma__free_from_callbacks(pVorbis->pData, &pDecoder->allocationCallbacks); + ma__free_from_callbacks(pVorbis, &pDecoder->allocationCallbacks); return MA_SUCCESS; } -ma_uint32 ma_decoder_internal_on_read_pcm_frames__vorbis(ma_pcm_converter* pDSP, void* pFramesOut, ma_uint32 frameCount, void* pUserData) +static ma_uint64 ma_decoder_internal_on_read_pcm_frames__vorbis(ma_decoder* pDecoder, void* pFramesOut, ma_uint64 frameCount) { - ma_decoder* pDecoder; ma_vorbis_decoder* pVorbis; - (void)pDSP; - - pDecoder = (ma_decoder*)pUserData; - ma_assert(pDecoder != NULL); - ma_assert(pDecoder->internalFormat == ma_format_f32); + MA_ASSERT(pDecoder != NULL); + MA_ASSERT(pFramesOut != NULL); + MA_ASSERT(pDecoder->internalFormat == ma_format_f32); pVorbis = (ma_vorbis_decoder*)pDecoder->pInternalDecoder; - ma_assert(pVorbis != NULL); + MA_ASSERT(pVorbis != NULL); return ma_vorbis_decoder_read_pcm_frames(pVorbis, pDecoder, pFramesOut, frameCount); } -ma_uint64 ma_decoder_internal_on_get_length_in_pcm_frames__vorbis(ma_decoder* pDecoder) +static ma_uint64 ma_decoder_internal_on_get_length_in_pcm_frames__vorbis(ma_decoder* pDecoder) { /* No good way to do this with Vorbis. */ (void)pDecoder; return 0; } -ma_result ma_decoder_init_vorbis__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder) +static ma_result ma_decoder_init_vorbis__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder) { - ma_result result; stb_vorbis* pInternalVorbis = NULL; size_t dataSize = 0; size_t dataCapacity = 0; @@ -34047,8 +36670,8 @@ ma_result ma_decoder_init_vorbis__internal(const ma_decoder_config* pConfig, ma_ size_t vorbisDataSize; ma_vorbis_decoder* pVorbis; - ma_assert(pConfig != NULL); - ma_assert(pDecoder != NULL); + MA_ASSERT(pConfig != NULL); + MA_ASSERT(pDecoder != NULL); /* We grow the buffer in chunks. */ do @@ -34058,11 +36681,12 @@ ma_result ma_decoder_init_vorbis__internal(const ma_decoder_config* pConfig, ma_ size_t bytesRead; int vorbisError = 0; int consumedDataSize = 0; + size_t oldCapacity = dataCapacity; dataCapacity += MA_VORBIS_DATA_CHUNK_SIZE; - pNewData = (ma_uint8*)ma_realloc(pData, dataCapacity); + pNewData = (ma_uint8*)ma__realloc_from_callbacks(pData, dataCapacity, oldCapacity, &pDecoder->allocationCallbacks); if (pNewData == NULL) { - ma_free(pData); + ma__free_from_callbacks(pData, &pDecoder->allocationCallbacks); return MA_OUT_OF_MEMORY; } @@ -34109,91 +36733,81 @@ ma_result ma_decoder_init_vorbis__internal(const ma_decoder_config* pConfig, ma_ /* Don't allow more than MA_MAX_CHANNELS channels. */ if (vorbisInfo.channels > MA_MAX_CHANNELS) { stb_vorbis_close(pInternalVorbis); - ma_free(pData); + ma__free_from_callbacks(pData, &pDecoder->allocationCallbacks); return MA_ERROR; /* Too many channels. */ } vorbisDataSize = sizeof(ma_vorbis_decoder) + sizeof(float)*vorbisInfo.max_frame_size; - pVorbis = (ma_vorbis_decoder*)ma_malloc(vorbisDataSize); + pVorbis = (ma_vorbis_decoder*)ma__malloc_from_callbacks(vorbisDataSize, &pDecoder->allocationCallbacks); if (pVorbis == NULL) { stb_vorbis_close(pInternalVorbis); - ma_free(pData); + ma__free_from_callbacks(pData, &pDecoder->allocationCallbacks); return MA_OUT_OF_MEMORY; } - ma_zero_memory(pVorbis, vorbisDataSize); + MA_ZERO_MEMORY(pVorbis, vorbisDataSize); pVorbis->pInternalVorbis = pInternalVorbis; pVorbis->pData = pData; pVorbis->dataSize = dataSize; pVorbis->dataCapacity = dataCapacity; - pDecoder->onSeekToPCMFrame = ma_decoder_internal_on_seek_to_pcm_frame__vorbis; - pDecoder->onUninit = ma_decoder_internal_on_uninit__vorbis; + pDecoder->onReadPCMFrames = ma_decoder_internal_on_read_pcm_frames__vorbis; + pDecoder->onSeekToPCMFrame = ma_decoder_internal_on_seek_to_pcm_frame__vorbis; + pDecoder->onUninit = ma_decoder_internal_on_uninit__vorbis; pDecoder->onGetLengthInPCMFrames = ma_decoder_internal_on_get_length_in_pcm_frames__vorbis; - pDecoder->pInternalDecoder = pVorbis; + pDecoder->pInternalDecoder = pVorbis; /* The internal format is always f32. */ - pDecoder->internalFormat = ma_format_f32; - pDecoder->internalChannels = vorbisInfo.channels; + pDecoder->internalFormat = ma_format_f32; + pDecoder->internalChannels = vorbisInfo.channels; pDecoder->internalSampleRate = vorbisInfo.sample_rate; ma_get_standard_channel_map(ma_standard_channel_map_vorbis, pDecoder->internalChannels, pDecoder->internalChannelMap); - result = ma_decoder__init_dsp(pDecoder, pConfig, ma_decoder_internal_on_read_pcm_frames__vorbis); - if (result != MA_SUCCESS) { - stb_vorbis_close(pVorbis->pInternalVorbis); - ma_free(pVorbis->pData); - ma_free(pVorbis); - return result; - } - return MA_SUCCESS; } -#endif +#endif /* STB_VORBIS_INCLUDE_STB_VORBIS_H */ /* MP3 */ #ifdef dr_mp3_h #define MA_HAS_MP3 -size_t ma_decoder_internal_on_read__mp3(void* pUserData, void* pBufferOut, size_t bytesToRead) +static size_t ma_decoder_internal_on_read__mp3(void* pUserData, void* pBufferOut, size_t bytesToRead) { ma_decoder* pDecoder = (ma_decoder*)pUserData; - ma_assert(pDecoder != NULL); + MA_ASSERT(pDecoder != NULL); return ma_decoder_read_bytes(pDecoder, pBufferOut, bytesToRead); } -drmp3_bool32 ma_decoder_internal_on_seek__mp3(void* pUserData, int offset, drmp3_seek_origin origin) +static drmp3_bool32 ma_decoder_internal_on_seek__mp3(void* pUserData, int offset, drmp3_seek_origin origin) { ma_decoder* pDecoder = (ma_decoder*)pUserData; - ma_assert(pDecoder != NULL); + MA_ASSERT(pDecoder != NULL); return ma_decoder_seek_bytes(pDecoder, offset, (origin == drmp3_seek_origin_start) ? ma_seek_origin_start : ma_seek_origin_current); } -ma_uint32 ma_decoder_internal_on_read_pcm_frames__mp3(ma_pcm_converter* pDSP, void* pFramesOut, ma_uint32 frameCount, void* pUserData) +static ma_uint64 ma_decoder_internal_on_read_pcm_frames__mp3(ma_decoder* pDecoder, void* pFramesOut, ma_uint64 frameCount) { - ma_decoder* pDecoder; drmp3* pMP3; - (void)pDSP; - - pDecoder = (ma_decoder*)pUserData; - ma_assert(pDecoder != NULL); - ma_assert(pDecoder->internalFormat == ma_format_f32); + MA_ASSERT(pDecoder != NULL); + MA_ASSERT(pFramesOut != NULL); + MA_ASSERT(pDecoder->internalFormat == ma_format_f32); pMP3 = (drmp3*)pDecoder->pInternalDecoder; - ma_assert(pMP3 != NULL); + MA_ASSERT(pMP3 != NULL); - return (ma_uint32)drmp3_read_pcm_frames_f32(pMP3, frameCount, (float*)pFramesOut); + return drmp3_read_pcm_frames_f32(pMP3, frameCount, (float*)pFramesOut); } -ma_result ma_decoder_internal_on_seek_to_pcm_frame__mp3(ma_decoder* pDecoder, ma_uint64 frameIndex) +static ma_result ma_decoder_internal_on_seek_to_pcm_frame__mp3(ma_decoder* pDecoder, ma_uint64 frameIndex) { drmp3* pMP3; drmp3_bool32 result; pMP3 = (drmp3*)pDecoder->pInternalDecoder; - ma_assert(pMP3 != NULL); + MA_ASSERT(pMP3 != NULL); result = drmp3_seek_to_pcm_frame(pMP3, frameIndex); if (result) { @@ -34203,32 +36817,37 @@ ma_result ma_decoder_internal_on_seek_to_pcm_frame__mp3(ma_decoder* pDecoder, ma } } -ma_result ma_decoder_internal_on_uninit__mp3(ma_decoder* pDecoder) +static ma_result ma_decoder_internal_on_uninit__mp3(ma_decoder* pDecoder) { drmp3_uninit((drmp3*)pDecoder->pInternalDecoder); - ma_free(pDecoder->pInternalDecoder); + ma__free_from_callbacks(pDecoder->pInternalDecoder, &pDecoder->allocationCallbacks); return MA_SUCCESS; } -ma_uint64 ma_decoder_internal_on_get_length_in_pcm_frames__mp3(ma_decoder* pDecoder) +static ma_uint64 ma_decoder_internal_on_get_length_in_pcm_frames__mp3(ma_decoder* pDecoder) { return drmp3_get_pcm_frame_count((drmp3*)pDecoder->pInternalDecoder); } -ma_result ma_decoder_init_mp3__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder) +static ma_result ma_decoder_init_mp3__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder) { drmp3* pMP3; drmp3_config mp3Config; - ma_result result; + drmp3_allocation_callbacks allocationCallbacks; - ma_assert(pConfig != NULL); - ma_assert(pDecoder != NULL); + MA_ASSERT(pConfig != NULL); + MA_ASSERT(pDecoder != NULL); - pMP3 = (drmp3*)ma_malloc(sizeof(*pMP3)); + pMP3 = (drmp3*)ma__malloc_from_callbacks(sizeof(*pMP3), &pDecoder->allocationCallbacks); if (pMP3 == NULL) { return MA_OUT_OF_MEMORY; } + allocationCallbacks.pUserData = pDecoder->allocationCallbacks.pUserData; + allocationCallbacks.onMalloc = pDecoder->allocationCallbacks.onMalloc; + allocationCallbacks.onRealloc = pDecoder->allocationCallbacks.onRealloc; + allocationCallbacks.onFree = pDecoder->allocationCallbacks.onFree; + /* Try opening the decoder first. MP3 can have variable sample rates (it's per frame/packet). We therefore need to use some smarts to determine the most appropriate internal sample rate. These are the rules we're going @@ -34240,59 +36859,74 @@ ma_result ma_decoder_init_mp3__internal(const ma_decoder_config* pConfig, ma_dec The internal channel count is always stereo, and the internal format is always f32. */ - ma_zero_object(&mp3Config); + MA_ZERO_OBJECT(&mp3Config); mp3Config.outputChannels = 2; mp3Config.outputSampleRate = (pConfig->sampleRate != 0) ? pConfig->sampleRate : 44100; - if (!drmp3_init(pMP3, ma_decoder_internal_on_read__mp3, ma_decoder_internal_on_seek__mp3, pDecoder, &mp3Config, NULL)) { - ma_free(pMP3); + if (!drmp3_init(pMP3, ma_decoder_internal_on_read__mp3, ma_decoder_internal_on_seek__mp3, pDecoder, &mp3Config, &allocationCallbacks)) { + ma__free_from_callbacks(pMP3, &pDecoder->allocationCallbacks); return MA_ERROR; } /* If we get here it means we successfully initialized the MP3 decoder. We can now initialize the rest of the ma_decoder. */ - pDecoder->onSeekToPCMFrame = ma_decoder_internal_on_seek_to_pcm_frame__mp3; - pDecoder->onUninit = ma_decoder_internal_on_uninit__mp3; + pDecoder->onReadPCMFrames = ma_decoder_internal_on_read_pcm_frames__mp3; + pDecoder->onSeekToPCMFrame = ma_decoder_internal_on_seek_to_pcm_frame__mp3; + pDecoder->onUninit = ma_decoder_internal_on_uninit__mp3; pDecoder->onGetLengthInPCMFrames = ma_decoder_internal_on_get_length_in_pcm_frames__mp3; - pDecoder->pInternalDecoder = pMP3; + pDecoder->pInternalDecoder = pMP3; /* Internal format. */ - pDecoder->internalFormat = ma_format_f32; - pDecoder->internalChannels = pMP3->channels; + pDecoder->internalFormat = ma_format_f32; + pDecoder->internalChannels = pMP3->channels; pDecoder->internalSampleRate = pMP3->sampleRate; ma_get_standard_channel_map(ma_standard_channel_map_default, pDecoder->internalChannels, pDecoder->internalChannelMap); - result = ma_decoder__init_dsp(pDecoder, pConfig, ma_decoder_internal_on_read_pcm_frames__mp3); - if (result != MA_SUCCESS) { - drmp3_uninit(pMP3); - ma_free(pMP3); - return result; - } - return MA_SUCCESS; } -#endif +#endif /* dr_mp3_h */ /* Raw */ -ma_uint32 ma_decoder_internal_on_read_pcm_frames__raw(ma_pcm_converter* pDSP, void* pFramesOut, ma_uint32 frameCount, void* pUserData) +static ma_uint64 ma_decoder_internal_on_read_pcm_frames__raw(ma_decoder* pDecoder, void* pFramesOut, ma_uint64 frameCount) { - ma_decoder* pDecoder; ma_uint32 bpf; + ma_uint64 totalFramesRead; + void* pRunningFramesOut; - (void)pDSP; - pDecoder = (ma_decoder*)pUserData; - ma_assert(pDecoder != NULL); + MA_ASSERT(pDecoder != NULL); + MA_ASSERT(pFramesOut != NULL); /* For raw decoding we just read directly from the decoder's callbacks. */ bpf = ma_get_bytes_per_frame(pDecoder->internalFormat, pDecoder->internalChannels); - return (ma_uint32)ma_decoder_read_bytes(pDecoder, pFramesOut, frameCount * bpf) / bpf; + + totalFramesRead = 0; + pRunningFramesOut = pFramesOut; + + while (totalFramesRead < frameCount) { + ma_uint64 framesReadThisIteration; + ma_uint64 framesToReadThisIteration = (frameCount - totalFramesRead); + if (framesToReadThisIteration > MA_SIZE_MAX) { + framesToReadThisIteration = MA_SIZE_MAX; + } + + framesReadThisIteration = ma_decoder_read_bytes(pDecoder, pRunningFramesOut, (size_t)framesToReadThisIteration * bpf) / bpf; /* Safe cast to size_t. */ + + totalFramesRead += framesReadThisIteration; + pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesReadThisIteration * bpf); + + if (framesReadThisIteration < framesToReadThisIteration) { + break; /* Done. */ + } + } + + return totalFramesRead; } -ma_result ma_decoder_internal_on_seek_to_pcm_frame__raw(ma_decoder* pDecoder, ma_uint64 frameIndex) +static ma_result ma_decoder_internal_on_seek_to_pcm_frame__raw(ma_decoder* pDecoder, ma_uint64 frameIndex) { ma_bool32 result = MA_FALSE; ma_uint64 totalBytesToSeek; - ma_assert(pDecoder != NULL); + MA_ASSERT(pDecoder != NULL); if (pDecoder->onSeek == NULL) { return MA_ERROR; @@ -34332,53 +36966,61 @@ ma_result ma_decoder_internal_on_seek_to_pcm_frame__raw(ma_decoder* pDecoder, ma } } -ma_result ma_decoder_internal_on_uninit__raw(ma_decoder* pDecoder) +static ma_result ma_decoder_internal_on_uninit__raw(ma_decoder* pDecoder) { (void)pDecoder; return MA_SUCCESS; } -ma_uint64 ma_decoder_internal_on_get_length_in_pcm_frames__raw(ma_decoder* pDecoder) +static ma_uint64 ma_decoder_internal_on_get_length_in_pcm_frames__raw(ma_decoder* pDecoder) { (void)pDecoder; return 0; } -ma_result ma_decoder_init_raw__internal(const ma_decoder_config* pConfigIn, const ma_decoder_config* pConfigOut, ma_decoder* pDecoder) +static ma_result ma_decoder_init_raw__internal(const ma_decoder_config* pConfigIn, const ma_decoder_config* pConfigOut, ma_decoder* pDecoder) { - ma_result result; - - ma_assert(pConfigIn != NULL); - ma_assert(pConfigOut != NULL); - ma_assert(pDecoder != NULL); + MA_ASSERT(pConfigIn != NULL); + MA_ASSERT(pConfigOut != NULL); + MA_ASSERT(pDecoder != NULL); - pDecoder->onSeekToPCMFrame = ma_decoder_internal_on_seek_to_pcm_frame__raw; - pDecoder->onUninit = ma_decoder_internal_on_uninit__raw; + pDecoder->onReadPCMFrames = ma_decoder_internal_on_read_pcm_frames__raw; + pDecoder->onSeekToPCMFrame = ma_decoder_internal_on_seek_to_pcm_frame__raw; + pDecoder->onUninit = ma_decoder_internal_on_uninit__raw; pDecoder->onGetLengthInPCMFrames = ma_decoder_internal_on_get_length_in_pcm_frames__raw; /* Internal format. */ - pDecoder->internalFormat = pConfigIn->format; - pDecoder->internalChannels = pConfigIn->channels; + pDecoder->internalFormat = pConfigIn->format; + pDecoder->internalChannels = pConfigIn->channels; pDecoder->internalSampleRate = pConfigIn->sampleRate; ma_channel_map_copy(pDecoder->internalChannelMap, pConfigIn->channelMap, pConfigIn->channels); - result = ma_decoder__init_dsp(pDecoder, pConfigOut, ma_decoder_internal_on_read_pcm_frames__raw); - if (result != MA_SUCCESS) { - return result; - } - return MA_SUCCESS; } -ma_result ma_decoder__preinit(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder) +static ma_result ma_decoder__init_allocation_callbacks(const ma_decoder_config* pConfig, ma_decoder* pDecoder) +{ + MA_ASSERT(pDecoder != NULL); + + if (pConfig != NULL) { + return ma_allocation_callbacks_init_copy(&pDecoder->allocationCallbacks, &pConfig->allocationCallbacks); + } else { + pDecoder->allocationCallbacks = ma_allocation_callbacks_init_default(); + return MA_SUCCESS; + } +} + +static ma_result ma_decoder__preinit(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder) { - ma_assert(pConfig != NULL); + ma_result result; + + MA_ASSERT(pConfig != NULL); if (pDecoder == NULL) { return MA_INVALID_ARGS; } - ma_zero_object(pDecoder); + MA_ZERO_OBJECT(pDecoder); if (onRead == NULL || onSeek == NULL) { return MA_INVALID_ARGS; @@ -34388,10 +37030,26 @@ ma_result ma_decoder__preinit(ma_decoder_read_proc onRead, ma_decoder_seek_proc pDecoder->onSeek = onSeek; pDecoder->pUserData = pUserData; - (void)pConfig; + result = ma_decoder__init_allocation_callbacks(pConfig, pDecoder); + if (result != MA_SUCCESS) { + return result; + } + return MA_SUCCESS; } +static ma_result ma_decoder__postinit(const ma_decoder_config* pConfig, ma_decoder* pDecoder) +{ + ma_result result; + + result = ma_decoder__init_data_converter(pDecoder, pConfig); + if (result != MA_SUCCESS) { + return result; + } + + return result; +} + ma_result ma_decoder_init_wav(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder) { ma_decoder_config config; @@ -34405,10 +37063,15 @@ ma_result ma_decoder_init_wav(ma_decoder_read_proc onRead, ma_decoder_seek_proc } #ifdef MA_HAS_WAV - return ma_decoder_init_wav__internal(&config, pDecoder); + result = ma_decoder_init_wav__internal(&config, pDecoder); #else - return MA_NO_BACKEND; + result = MA_NO_BACKEND; #endif + if (result != MA_SUCCESS) { + return result; + } + + return ma_decoder__postinit(&config, pDecoder); } ma_result ma_decoder_init_flac(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder) @@ -34424,10 +37087,15 @@ ma_result ma_decoder_init_flac(ma_decoder_read_proc onRead, ma_decoder_seek_proc } #ifdef MA_HAS_FLAC - return ma_decoder_init_flac__internal(&config, pDecoder); + result = ma_decoder_init_flac__internal(&config, pDecoder); #else - return MA_NO_BACKEND; + result = MA_NO_BACKEND; #endif + if (result != MA_SUCCESS) { + return result; + } + + return ma_decoder__postinit(&config, pDecoder); } ma_result ma_decoder_init_vorbis(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder) @@ -34443,10 +37111,15 @@ ma_result ma_decoder_init_vorbis(ma_decoder_read_proc onRead, ma_decoder_seek_pr } #ifdef MA_HAS_VORBIS - return ma_decoder_init_vorbis__internal(&config, pDecoder); + result = ma_decoder_init_vorbis__internal(&config, pDecoder); #else - return MA_NO_BACKEND; + result = MA_NO_BACKEND; #endif + if (result != MA_SUCCESS) { + return result; + } + + return ma_decoder__postinit(&config, pDecoder); } ma_result ma_decoder_init_mp3(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder) @@ -34462,10 +37135,15 @@ ma_result ma_decoder_init_mp3(ma_decoder_read_proc onRead, ma_decoder_seek_proc } #ifdef MA_HAS_MP3 - return ma_decoder_init_mp3__internal(&config, pDecoder); + result = ma_decoder_init_mp3__internal(&config, pDecoder); #else - return MA_NO_BACKEND; + result = MA_NO_BACKEND; #endif + if (result != MA_SUCCESS) { + return result; + } + + return ma_decoder__postinit(&config, pDecoder); } ma_result ma_decoder_init_raw(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfigIn, const ma_decoder_config* pConfigOut, ma_decoder* pDecoder) @@ -34480,15 +37158,20 @@ ma_result ma_decoder_init_raw(ma_decoder_read_proc onRead, ma_decoder_seek_proc return result; } - return ma_decoder_init_raw__internal(pConfigIn, &config, pDecoder); + result = ma_decoder_init_raw__internal(pConfigIn, &config, pDecoder); + if (result != MA_SUCCESS) { + return result; + } + + return ma_decoder__postinit(&config, pDecoder); } -ma_result ma_decoder_init__internal(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder) +static ma_result ma_decoder_init__internal(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder) { ma_result result = MA_NO_BACKEND; - ma_assert(pConfig != NULL); - ma_assert(pDecoder != NULL); + MA_ASSERT(pConfig != NULL); + MA_ASSERT(pDecoder != NULL); /* Silence some warnings in the case that we don't have any decoder backends enabled. */ (void)onRead; @@ -34536,7 +37219,7 @@ ma_result ma_decoder_init__internal(ma_decoder_read_proc onRead, ma_decoder_seek return result; } - return result; + return ma_decoder__postinit(pConfig, pDecoder); } ma_result ma_decoder_init(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder) @@ -34555,11 +37238,11 @@ ma_result ma_decoder_init(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSe } -size_t ma_decoder__on_read_memory(ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead) +static size_t ma_decoder__on_read_memory(ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead) { size_t bytesRemaining; - ma_assert(pDecoder->memory.dataSize >= pDecoder->memory.currentReadPos); + MA_ASSERT(pDecoder->memory.dataSize >= pDecoder->memory.currentReadPos); bytesRemaining = pDecoder->memory.dataSize - pDecoder->memory.currentReadPos; if (bytesToRead > bytesRemaining) { @@ -34567,14 +37250,14 @@ size_t ma_decoder__on_read_memory(ma_decoder* pDecoder, void* pBufferOut, size_t } if (bytesToRead > 0) { - ma_copy_memory(pBufferOut, pDecoder->memory.pData + pDecoder->memory.currentReadPos, bytesToRead); + MA_COPY_MEMORY(pBufferOut, pDecoder->memory.pData + pDecoder->memory.currentReadPos, bytesToRead); pDecoder->memory.currentReadPos += bytesToRead; } return bytesToRead; } -ma_bool32 ma_decoder__on_seek_memory(ma_decoder* pDecoder, int byteOffset, ma_seek_origin origin) +static ma_bool32 ma_decoder__on_seek_memory(ma_decoder* pDecoder, int byteOffset, ma_seek_origin origin) { if (origin == ma_seek_origin_current) { if (byteOffset > 0) { @@ -34600,7 +37283,7 @@ ma_bool32 ma_decoder__on_seek_memory(ma_decoder* pDecoder, int byteOffset, ma_se return MA_TRUE; } -ma_result ma_decoder__preinit_memory(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder) +static ma_result ma_decoder__preinit_memory(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder) { ma_result result = ma_decoder__preinit(ma_decoder__on_read_memory, ma_decoder__on_seek_memory, NULL, pConfig, pDecoder); if (result != MA_SUCCESS) { @@ -34647,10 +37330,15 @@ ma_result ma_decoder_init_memory_wav(const void* pData, size_t dataSize, const m } #ifdef MA_HAS_WAV - return ma_decoder_init_wav__internal(&config, pDecoder); + result = ma_decoder_init_wav__internal(&config, pDecoder); #else - return MA_NO_BACKEND; + result = MA_NO_BACKEND; #endif + if (result != MA_SUCCESS) { + return result; + } + + return ma_decoder__postinit(&config, pDecoder); } ma_result ma_decoder_init_memory_flac(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder) @@ -34666,10 +37354,15 @@ ma_result ma_decoder_init_memory_flac(const void* pData, size_t dataSize, const } #ifdef MA_HAS_FLAC - return ma_decoder_init_flac__internal(&config, pDecoder); + result = ma_decoder_init_flac__internal(&config, pDecoder); #else - return MA_NO_BACKEND; + result = MA_NO_BACKEND; #endif + if (result != MA_SUCCESS) { + return result; + } + + return ma_decoder__postinit(&config, pDecoder); } ma_result ma_decoder_init_memory_vorbis(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder) @@ -34685,10 +37378,15 @@ ma_result ma_decoder_init_memory_vorbis(const void* pData, size_t dataSize, cons } #ifdef MA_HAS_VORBIS - return ma_decoder_init_vorbis__internal(&config, pDecoder); + result = ma_decoder_init_vorbis__internal(&config, pDecoder); #else - return MA_NO_BACKEND; + result = MA_NO_BACKEND; #endif + if (result != MA_SUCCESS) { + return result; + } + + return ma_decoder__postinit(&config, pDecoder); } ma_result ma_decoder_init_memory_mp3(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder) @@ -34704,10 +37402,15 @@ ma_result ma_decoder_init_memory_mp3(const void* pData, size_t dataSize, const m } #ifdef MA_HAS_MP3 - return ma_decoder_init_mp3__internal(&config, pDecoder); + result = ma_decoder_init_mp3__internal(&config, pDecoder); #else - return MA_NO_BACKEND; + result = MA_NO_BACKEND; #endif + if (result != MA_SUCCESS) { + return result; + } + + return ma_decoder__postinit(&config, pDecoder); } ma_result ma_decoder_init_memory_raw(const void* pData, size_t dataSize, const ma_decoder_config* pConfigIn, const ma_decoder_config* pConfigOut, ma_decoder* pDecoder) @@ -34722,17 +37425,16 @@ ma_result ma_decoder_init_memory_raw(const void* pData, size_t dataSize, const m return result; } - return ma_decoder_init_raw__internal(pConfigIn, &config, pDecoder); + result = ma_decoder_init_raw__internal(pConfigIn, &config, pDecoder); + if (result != MA_SUCCESS) { + return result; + } + + return ma_decoder__postinit(&config, pDecoder); } #ifndef MA_NO_STDIO -#include -#if !defined(_MSC_VER) && !defined(__DMC__) -#include /* For strcasecmp(). */ -#include /* For wcsrtombs() */ -#endif - -const char* ma_path_file_name(const char* path) +static const char* ma_path_file_name(const char* path) { const char* fileName; @@ -34759,7 +37461,7 @@ const char* ma_path_file_name(const char* path) return fileName; } -const wchar_t* ma_path_file_name_w(const wchar_t* path) +static const wchar_t* ma_path_file_name_w(const wchar_t* path) { const wchar_t* fileName; @@ -34787,7 +37489,7 @@ const wchar_t* ma_path_file_name_w(const wchar_t* path) } -const char* ma_path_extension(const char* path) +static const char* ma_path_extension(const char* path) { const char* extension; const char* lastOccurance; @@ -34812,7 +37514,7 @@ const char* ma_path_extension(const char* path) return (lastOccurance != NULL) ? lastOccurance : extension; } -const wchar_t* ma_path_extension_w(const wchar_t* path) +static const wchar_t* ma_path_extension_w(const wchar_t* path) { const wchar_t* extension; const wchar_t* lastOccurance; @@ -34838,7 +37540,7 @@ const wchar_t* ma_path_extension_w(const wchar_t* path) } -ma_bool32 ma_path_extension_equal(const char* path, const char* extension) +static ma_bool32 ma_path_extension_equal(const char* path, const char* extension) { const char* ext1; const char* ext2; @@ -34857,7 +37559,7 @@ ma_bool32 ma_path_extension_equal(const char* path, const char* extension) #endif } -ma_bool32 ma_path_extension_equal_w(const wchar_t* path, const wchar_t* extension) +static ma_bool32 ma_path_extension_equal_w(const wchar_t* path, const wchar_t* extension) { const wchar_t* ext1; const wchar_t* ext2; @@ -34884,8 +37586,8 @@ ma_bool32 ma_path_extension_equal_w(const wchar_t* path, const wchar_t* extensio mbstate_t mbs1; mbstate_t mbs2; - ma_zero_object(&mbs1); - ma_zero_object(&mbs2); + MA_ZERO_OBJECT(&mbs1); + MA_ZERO_OBJECT(&mbs2); if (wcsrtombs(ext1MB, &pext1, sizeof(ext1MB), &mbs1) == (size_t)-1) { return MA_FALSE; @@ -34900,30 +37602,36 @@ ma_bool32 ma_path_extension_equal_w(const wchar_t* path, const wchar_t* extensio } -size_t ma_decoder__on_read_stdio(ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead) +static size_t ma_decoder__on_read_stdio(ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead) { return fread(pBufferOut, 1, bytesToRead, (FILE*)pDecoder->pUserData); } -ma_bool32 ma_decoder__on_seek_stdio(ma_decoder* pDecoder, int byteOffset, ma_seek_origin origin) +static ma_bool32 ma_decoder__on_seek_stdio(ma_decoder* pDecoder, int byteOffset, ma_seek_origin origin) { return fseek((FILE*)pDecoder->pUserData, byteOffset, (origin == ma_seek_origin_current) ? SEEK_CUR : SEEK_SET) == 0; } -ma_result ma_decoder__preinit_file(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder) +static ma_result ma_decoder__preinit_file(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder) { + ma_result result; FILE* pFile; if (pDecoder == NULL) { return MA_INVALID_ARGS; } - ma_zero_object(pDecoder); + MA_ZERO_OBJECT(pDecoder); if (pFilePath == NULL || pFilePath[0] == '\0') { return MA_INVALID_ARGS; } + result = ma_decoder__init_allocation_callbacks(pConfig, pDecoder); + if (result != MA_SUCCESS) { + return result; + } + #if defined(_MSC_VER) && _MSC_VER >= 1400 if (fopen_s(&pFile, pFilePath, "rb") != 0) { return MA_ERROR; @@ -34938,25 +37646,47 @@ ma_result ma_decoder__preinit_file(const char* pFilePath, const ma_decoder_confi /* We need to manually set the user data so the calls to ma_decoder__on_seek_stdio() succeed. */ pDecoder->pUserData = pFile; - (void)pConfig; return MA_SUCCESS; } -ma_result ma_decoder__preinit_file_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder) +/* +_wfopen() isn't always available in all compilation environments. + + * Windows only. + * MSVC seems to support it universally as far back as VC6 from what I can tell (haven't checked further back). + * MinGW-64 (both 32- and 64-bit) seems to support it. + * MinGW wraps it in !defined(__STRICT_ANSI__). + +This can be reviewed as compatibility issues arise. The preference is to use _wfopen_s() and _wfopen() as opposed to the wcsrtombs() +fallback, so if you notice your compiler not detecting this properly I'm happy to look at adding support. +*/ +#if defined(_WIN32) + #if defined(_MSC_VER) || defined(__MINGW64__) || !defined(__STRICT_ANSI__) + #define MA_HAS_WFOPEN + #endif +#endif + +static ma_result ma_decoder__preinit_file_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder) { + ma_result result; FILE* pFile; if (pDecoder == NULL) { return MA_INVALID_ARGS; } - ma_zero_object(pDecoder); + MA_ZERO_OBJECT(pDecoder); if (pFilePath == NULL || pFilePath[0] == '\0') { return MA_INVALID_ARGS; } -#if defined(_WIN32) + result = ma_decoder__init_allocation_callbacks(pConfig, pDecoder); + if (result != MA_SUCCESS) { + return result; + } + +#if defined(MA_HAS_WFOPEN) /* Use _wfopen() on Windows. */ #if defined(_MSC_VER) && _MSC_VER >= 1400 if (_wfopen_s(&pFile, pFilePath, L"rb") != 0) { @@ -34981,24 +37711,24 @@ ma_result ma_decoder__preinit_file_w(const wchar_t* pFilePath, const ma_decoder_ char* pFilePathMB = NULL; /* Get the length first. */ - ma_zero_object(&mbs); + MA_ZERO_OBJECT(&mbs); lenMB = wcsrtombs(NULL, &pFilePathTemp, 0, &mbs); if (lenMB == (size_t)-1) { return MA_ERROR; } - pFilePathMB = (char*)MA_MALLOC(lenMB + 1); + pFilePathMB = (char*)ma__malloc_from_callbacks(lenMB + 1, &pDecoder->allocationCallbacks); if (pFilePathMB == NULL) { return MA_OUT_OF_MEMORY; } pFilePathTemp = pFilePath; - ma_zero_object(&mbs); + MA_ZERO_OBJECT(&mbs); wcsrtombs(pFilePathMB, &pFilePathTemp, lenMB + 1, &mbs); pFile = fopen(pFilePathMB, "rb"); - MA_FREE(pFilePathMB); + ma__free_from_callbacks(pFilePathMB, &pDecoder->allocationCallbacks); } if (pFile == NULL) { @@ -35175,7 +37905,7 @@ ma_result ma_decoder_init_file_mp3_w(const wchar_t* pFilePath, const ma_decoder_ return ma_decoder_init_mp3(ma_decoder__on_read_stdio, ma_decoder__on_seek_stdio, pDecoder->pUserData, pConfig, pDecoder); } -#endif +#endif /* MA_NO_STDIO */ ma_result ma_decoder_uninit(ma_decoder* pDecoder) { @@ -35194,6 +37924,8 @@ ma_result ma_decoder_uninit(ma_decoder* pDecoder) } #endif + ma_data_converter_uninit(&pDecoder->converter); + return MA_SUCCESS; } @@ -35204,7 +37936,12 @@ ma_uint64 ma_decoder_get_length_in_pcm_frames(ma_decoder* pDecoder) } if (pDecoder->onGetLengthInPCMFrames) { - return pDecoder->onGetLengthInPCMFrames(pDecoder); + ma_uint64 nativeLengthInPCMFrames = pDecoder->onGetLengthInPCMFrames(pDecoder); + if (pDecoder->internalSampleRate == pDecoder->outputSampleRate) { + return nativeLengthInPCMFrames; + } else { + return ma_calculate_frame_count_after_resampling(pDecoder->outputSampleRate, pDecoder->internalSampleRate, nativeLengthInPCMFrames); + } } return 0; @@ -35212,11 +37949,73 @@ ma_uint64 ma_decoder_get_length_in_pcm_frames(ma_decoder* pDecoder) ma_uint64 ma_decoder_read_pcm_frames(ma_decoder* pDecoder, void* pFramesOut, ma_uint64 frameCount) { + ma_result result; + ma_uint64 totalFramesReadOut; + ma_uint64 totalFramesReadIn; + void* pRunningFramesOut; + if (pDecoder == NULL) { return 0; } - return ma_pcm_converter_read(&pDecoder->dsp, pFramesOut, frameCount); + if (pDecoder->onReadPCMFrames == NULL) { + return 0; + } + + /* Fast path. */ + if (pDecoder->converter.isPassthrough) { + return pDecoder->onReadPCMFrames(pDecoder, pFramesOut, frameCount); + } + + /* Getting here means we need to do data conversion. */ + totalFramesReadOut = 0; + totalFramesReadIn = 0; + pRunningFramesOut = pFramesOut; + + while (totalFramesReadOut < frameCount) { + ma_uint8 pIntermediaryBuffer[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In internal format. */ + ma_uint64 intermediaryBufferCap = sizeof(pIntermediaryBuffer) / ma_get_bytes_per_frame(pDecoder->internalFormat, pDecoder->internalChannels); + ma_uint64 framesToReadThisIterationIn; + ma_uint64 framesReadThisIterationIn; + ma_uint64 framesToReadThisIterationOut; + ma_uint64 framesReadThisIterationOut; + ma_uint64 requiredInputFrameCount; + + framesToReadThisIterationOut = (frameCount - totalFramesReadOut); + framesToReadThisIterationIn = framesToReadThisIterationOut; + if (framesToReadThisIterationIn > intermediaryBufferCap) { + framesToReadThisIterationIn = intermediaryBufferCap; + } + + requiredInputFrameCount = ma_data_converter_get_required_input_frame_count(&pDecoder->converter, framesToReadThisIterationOut); + if (framesToReadThisIterationIn > requiredInputFrameCount) { + framesToReadThisIterationIn = requiredInputFrameCount; + } + + if (requiredInputFrameCount > 0) { + framesReadThisIterationIn = pDecoder->onReadPCMFrames(pDecoder, pIntermediaryBuffer, framesToReadThisIterationIn); + totalFramesReadIn += framesReadThisIterationIn; + } + + /* + At this point we have our decoded data in input format and now we need to convert to output format. Note that even if we didn't read any + input frames, we still want to try processing frames because there may some output frames generated from cached input data. + */ + framesReadThisIterationOut = framesToReadThisIterationOut; + result = ma_data_converter_process_pcm_frames(&pDecoder->converter, pIntermediaryBuffer, &framesReadThisIterationIn, pRunningFramesOut, &framesReadThisIterationOut); + if (result != MA_SUCCESS) { + break; + } + + totalFramesReadOut += framesReadThisIterationOut; + pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesReadThisIterationOut * ma_get_bytes_per_frame(pDecoder->outputFormat, pDecoder->outputChannels)); + + if (framesReadThisIterationIn == 0 && framesReadThisIterationOut == 0) { + break; /* We're done. */ + } + } + + return totalFramesReadOut; } ma_result ma_decoder_seek_to_pcm_frame(ma_decoder* pDecoder, ma_uint64 frameIndex) @@ -35234,14 +38033,14 @@ ma_result ma_decoder_seek_to_pcm_frame(ma_decoder* pDecoder, ma_uint64 frameInde } -ma_result ma_decoder__full_decode_and_uninit(ma_decoder* pDecoder, ma_decoder_config* pConfigOut, ma_uint64* pFrameCountOut, void** ppPCMFramesOut) +static ma_result ma_decoder__full_decode_and_uninit(ma_decoder* pDecoder, ma_decoder_config* pConfigOut, ma_uint64* pFrameCountOut, void** ppPCMFramesOut) { ma_uint64 totalFrameCount; ma_uint64 bpf; ma_uint64 dataCapInFrames; void* pPCMFramesOut; - ma_assert(pDecoder != NULL); + MA_ASSERT(pDecoder != NULL); totalFrameCount = 0; bpf = ma_get_bytes_per_frame(pDecoder->outputFormat, pDecoder->outputChannels); @@ -35256,20 +38055,21 @@ ma_result ma_decoder__full_decode_and_uninit(ma_decoder* pDecoder, ma_decoder_co /* Make room if there's not enough. */ if (totalFrameCount == dataCapInFrames) { void* pNewPCMFramesOut; + ma_uint64 oldDataCapInFrames = dataCapInFrames; ma_uint64 newDataCapInFrames = dataCapInFrames*2; if (newDataCapInFrames == 0) { newDataCapInFrames = 4096; } if ((newDataCapInFrames * bpf) > MA_SIZE_MAX) { - ma_free(pPCMFramesOut); + ma__free_from_callbacks(pPCMFramesOut, &pDecoder->allocationCallbacks); return MA_TOO_LARGE; } - pNewPCMFramesOut = (void*)ma_realloc(pPCMFramesOut, (size_t)(newDataCapInFrames * bpf)); + pNewPCMFramesOut = (void*)ma__realloc_from_callbacks(pPCMFramesOut, (size_t)(newDataCapInFrames * bpf), (size_t)(oldDataCapInFrames * bpf), &pDecoder->allocationCallbacks); if (pNewPCMFramesOut == NULL) { - ma_free(pPCMFramesOut); + ma__free_from_callbacks(pPCMFramesOut, &pDecoder->allocationCallbacks); return MA_OUT_OF_MEMORY; } @@ -35278,7 +38078,7 @@ ma_result ma_decoder__full_decode_and_uninit(ma_decoder* pDecoder, ma_decoder_co } frameCountToTryReading = dataCapInFrames - totalFrameCount; - ma_assert(frameCountToTryReading > 0); + MA_ASSERT(frameCountToTryReading > 0); framesJustRead = ma_decoder_read_pcm_frames(pDecoder, (ma_uint8*)pPCMFramesOut + (totalFrameCount * bpf), frameCountToTryReading); totalFrameCount += framesJustRead; @@ -35299,7 +38099,7 @@ ma_result ma_decoder__full_decode_and_uninit(ma_decoder* pDecoder, ma_decoder_co if (ppPCMFramesOut != NULL) { *ppPCMFramesOut = pPCMFramesOut; } else { - ma_free(pPCMFramesOut); + ma__free_from_callbacks(pPCMFramesOut, &pDecoder->allocationCallbacks); } if (pFrameCountOut != NULL) { @@ -35376,68 +38176,215 @@ ma_result ma_decode_memory(const void* pData, size_t dataSize, ma_decoder_config Generation **************************************************************************************************************************************************************/ -ma_result ma_sine_wave_init(double amplitude, double periodsPerSecond, ma_uint32 sampleRate, ma_sine_wave* pSineWave) +ma_result ma_waveform_init(ma_waveform_type type, double amplitude, double frequency, ma_uint32 sampleRate, ma_waveform* pWaveform) +{ + if (pWaveform == NULL) { + return MA_INVALID_ARGS; + } + + MA_ZERO_OBJECT(pWaveform); + + pWaveform->type = type; + pWaveform->amplitude = amplitude; + pWaveform->frequency = frequency; + pWaveform->deltaTime = 1.0 / sampleRate; + pWaveform->time = 0; + + return MA_SUCCESS; +} + +ma_result ma_waveform_set_amplitude(ma_waveform* pWaveform, double amplitude) { - if (pSineWave == NULL) { + if (pWaveform == NULL) { return MA_INVALID_ARGS; } - ma_zero_object(pSineWave); - if (amplitude == 0 || periodsPerSecond == 0) { + pWaveform->amplitude = amplitude; + return MA_SUCCESS; +} + +ma_result ma_waveform_set_frequency(ma_waveform* pWaveform, double frequency) +{ + if (pWaveform == NULL) { return MA_INVALID_ARGS; } - if (amplitude > 1) { - amplitude = 1; + pWaveform->frequency = frequency; + return MA_SUCCESS; +} + +ma_result ma_waveform_set_sample_rate(ma_waveform* pWaveform, ma_uint32 sampleRate) +{ + if (pWaveform == NULL) { + return MA_INVALID_ARGS; } - if (amplitude < -1) { - amplitude = -1; + + pWaveform->deltaTime = 1.0 / sampleRate; + return MA_SUCCESS; +} + +static float ma_waveform_sine_f32(double time, double frequency, double amplitude) +{ + return (float)(ma_sin(MA_TAU_D * time * frequency) * amplitude); +} + +static float ma_waveform_square_f32(double time, double frequency, double amplitude) +{ + double t = time * frequency; + double f = t - (ma_uint64)t; + double r; + + if (f < 0.5) { + r = amplitude; + } else { + r = -amplitude; } - pSineWave->amplitude = amplitude; - pSineWave->periodsPerSecond = periodsPerSecond; - pSineWave->delta = MA_TAU_D / sampleRate; - pSineWave->time = 0; + return (float)r; +} - return MA_SUCCESS; +static float ma_waveform_triangle_f32(double time, double frequency, double amplitude) +{ + double t = time * frequency; + double f = t - (ma_uint64)t; + double r; + + r = 2 * ma_abs(2 * (f - 0.5)) - 1; + + return (float)(r * amplitude); +} + +static float ma_waveform_sawtooth_f32(double time, double frequency, double amplitude) +{ + double t = time * frequency; + double f = t - (ma_uint64)t; + double r; + + r = 2 * (f - 0.5); + + return (float)(r * amplitude); } -ma_uint64 ma_sine_wave_read_f32(ma_sine_wave* pSineWave, ma_uint64 count, float* pSamples) +static void ma_waveform_read_pcm_frames__sine(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount, ma_format format, ma_uint32 channels) { - return ma_sine_wave_read_f32_ex(pSineWave, count, 1, ma_stream_layout_interleaved, &pSamples); + ma_uint64 iFrame; + ma_uint64 iChannel; + ma_uint32 bpf = ma_get_bytes_per_frame(format, channels); + ma_uint32 bps = ma_get_bytes_per_sample(format); + + MA_ASSERT(pWaveform != NULL); + MA_ASSERT(pFramesOut != NULL); + + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + float s = ma_waveform_sine_f32(pWaveform->time, pWaveform->frequency, pWaveform->amplitude); + pWaveform->time += pWaveform->deltaTime; + + for (iChannel = 0; iChannel < channels; iChannel += 1) { + ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), format, &s, ma_format_f32, 1, ma_dither_mode_none); + } + } +} + +static void ma_waveform_read_pcm_frames__square(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount, ma_format format, ma_uint32 channels) +{ + ma_uint64 iFrame; + ma_uint64 iChannel; + ma_uint32 bpf = ma_get_bytes_per_frame(format, channels); + ma_uint32 bps = ma_get_bytes_per_sample(format); + + MA_ASSERT(pWaveform != NULL); + MA_ASSERT(pFramesOut != NULL); + + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + float s = ma_waveform_square_f32(pWaveform->time, pWaveform->frequency, pWaveform->amplitude); + pWaveform->time += pWaveform->deltaTime; + + for (iChannel = 0; iChannel < channels; iChannel += 1) { + ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), format, &s, ma_format_f32, 1, ma_dither_mode_none); + } + } +} + +static void ma_waveform_read_pcm_frames__triangle(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount, ma_format format, ma_uint32 channels) +{ + ma_uint64 iFrame; + ma_uint64 iChannel; + ma_uint32 bpf = ma_get_bytes_per_frame(format, channels); + ma_uint32 bps = ma_get_bytes_per_sample(format); + + MA_ASSERT(pWaveform != NULL); + MA_ASSERT(pFramesOut != NULL); + + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + float s = ma_waveform_triangle_f32(pWaveform->time, pWaveform->frequency, pWaveform->amplitude); + pWaveform->time += pWaveform->deltaTime; + + for (iChannel = 0; iChannel < channels; iChannel += 1) { + ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), format, &s, ma_format_f32, 1, ma_dither_mode_none); + } + } +} + +static void ma_waveform_read_pcm_frames__sawtooth(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount, ma_format format, ma_uint32 channels) +{ + ma_uint64 iFrame; + ma_uint64 iChannel; + ma_uint32 bpf = ma_get_bytes_per_frame(format, channels); + ma_uint32 bps = ma_get_bytes_per_sample(format); + + MA_ASSERT(pWaveform != NULL); + MA_ASSERT(pFramesOut != NULL); + + for (iFrame = 0; iFrame < frameCount; iFrame += 1) { + float s = ma_waveform_sawtooth_f32(pWaveform->time, pWaveform->frequency, pWaveform->amplitude); + pWaveform->time += pWaveform->deltaTime; + + for (iChannel = 0; iChannel < channels; iChannel += 1) { + ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), format, &s, ma_format_f32, 1, ma_dither_mode_none); + } + } } -ma_uint64 ma_sine_wave_read_f32_ex(ma_sine_wave* pSineWave, ma_uint64 frameCount, ma_uint32 channels, ma_stream_layout layout, float** ppFrames) +ma_uint64 ma_waveform_read_pcm_frames(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount, ma_format format, ma_uint32 channels) { - if (pSineWave == NULL) { + if (pWaveform == NULL) { return 0; } - if (ppFrames != NULL) { - ma_uint64 iFrame; - for (iFrame = 0; iFrame < frameCount; iFrame += 1) { - ma_uint32 iChannel; + if (pFramesOut != NULL) { + switch (pWaveform->type) + { + case ma_waveform_type_sine: + { + ma_waveform_read_pcm_frames__sine(pWaveform, pFramesOut, frameCount, format, channels); + } break; + + case ma_waveform_type_square: + { + ma_waveform_read_pcm_frames__square(pWaveform, pFramesOut, frameCount, format, channels); + } break; - float s = (float)(sin(pSineWave->time * pSineWave->periodsPerSecond) * pSineWave->amplitude); - pSineWave->time += pSineWave->delta; + case ma_waveform_type_triangle: + { + ma_waveform_read_pcm_frames__triangle(pWaveform, pFramesOut, frameCount, format, channels); + } break; - if (layout == ma_stream_layout_interleaved) { - for (iChannel = 0; iChannel < channels; iChannel += 1) { - ppFrames[0][iFrame*channels + iChannel] = s; - } - } else { - for (iChannel = 0; iChannel < channels; iChannel += 1) { - ppFrames[iChannel][iFrame] = s; - } - } + case ma_waveform_type_sawtooth: + { + ma_waveform_read_pcm_frames__sawtooth(pWaveform, pFramesOut, frameCount, format, channels); + } break; + + default: return 0; } } else { - pSineWave->time += pSineWave->delta * (ma_int64)frameCount; /* Cast to int64 required for VC6. */ + pWaveform->time += pWaveform->deltaTime * (ma_int64)frameCount; /* Cast to int64 required for VC6. Won't affect anything in practice. */ } return frameCount; } + +/* End globally disabled warnings. */ #if defined(_MSC_VER) #pragma warning(pop) #endif @@ -35445,36 +38392,158 @@ ma_uint64 ma_sine_wave_read_f32_ex(ma_sine_wave* pSineWave, ma_uint64 frameCount #endif /* MINIAUDIO_IMPLEMENTATION */ /* -BACKEND IMPLEMENTATION GUIDELINES -================================= -Context -------- -- Run-time linking if possible. -- Set whether or not it's an asynchronous backend +MAJOR CHANGES IN VERSION 0.9 +============================ +Version 0.9 includes major API changes, centered mostly around full-duplex and the rebrand to "miniaudio". Before I go into +detail about the major changes I would like to apologize. I know it's annoying dealing with breaking API changes, but I think +it's best to get these changes out of the way now while the library is still relatively young and unknown. -Device ------- -- If a full-duplex device is requested and the backend does not support full duplex devices, have ma_device_init__[backend]() - return MA_DEVICE_TYPE_NOT_SUPPORTED. -- If exclusive mode is requested, but the backend does not support it, return MA_SHARE_MODE_NOT_SUPPORTED. If practical, try - not to fall back to a different share mode - give the client exactly what they asked for. Some backends, such as ALSA, may - not have a practical way to distinguish between the two. -- If pDevice->usingDefault* is set, prefer the device's native value if the backend supports it. Otherwise use the relevant - value from the config. -- If the configs buffer size in frames is 0, set it based on the buffer size in milliseconds, keeping in mind to handle the - case when the default sample rate is being used where practical. -- Backends must set the following members of pDevice before returning successfully from ma_device_init__[backend](): - - internalFormat - - internalChannels - - internalSampleRate - - internalChannelMap - - bufferSizeInFrames - - periods +There's been a lot of refactoring with this release so there's a good chance a few bugs have been introduced. I apologize in +advance for this. You may want to hold off on upgrading for the short term if you're worried. If mini_al v0.8.14 works for +you, and you don't need full-duplex support, you can avoid upgrading (though you won't be getting future bug fixes). + + +Rebranding to "miniaudio" +------------------------- +The decision was made to rename mini_al to miniaudio. Don't worry, it's the same project. The reason for this is simple: + +1) Having the word "audio" in the title makes it immediately clear that the library is related to audio; and +2) I don't like the look of the underscore. + +This rebrand has necessitated a change in namespace from "mal" to "ma". I know this is annoying, and I apologize, but it's +better to get this out of the road now rather than later. Also, since there are necessary API changes for full-duplex support +I think it's better to just get the namespace change over and done with at the same time as the full-duplex changes. I'm hoping +this will be the last of the major API changes. Fingers crossed! + +The implementation define is now "#define MINIAUDIO_IMPLEMENTATION". You can also use "#define MA_IMPLEMENTATION" if that's +your preference. + + +Full-Duplex Support +------------------- +The major feature added to version 0.9 is full-duplex. This has necessitated a few API changes. + +1) The data callback has now changed. Previously there was one type of callback for playback and another for capture. I wanted + to avoid a third callback just for full-duplex so the decision was made to break this API and unify the callbacks. Now, + there is just one callback which is the same for all three modes (playback, capture, duplex). The new callback looks like + the following: + + void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount); + + This callback allows you to move data straight out of the input buffer and into the output buffer in full-duplex mode. In + playback-only mode, pInput will be null. Likewise, pOutput will be null in capture-only mode. The sample count is no longer + returned from the callback since it's not necessary for miniaudio anymore. + +2) The device config needed to change in order to support full-duplex. Full-duplex requires the ability to allow the client + to choose a different PCM format for the playback and capture sides. The old ma_device_config object simply did not allow + this and needed to change. With these changes you now specify the device ID, format, channels, channel map and share mode + on a per-playback and per-capture basis (see example below). The sample rate must be the same for playback and capture. + + Since the device config API has changed I have also decided to take the opportunity to simplify device initialization. Now, + the device ID, device type and callback user data are set in the config. ma_device_init() is now simplified down to taking + just the context, device config and a pointer to the device object being initialized. The rationale for this change is that + it just makes more sense to me that these are set as part of the config like everything else. + + Example device initialization: + + ma_device_config config = ma_device_config_init(ma_device_type_duplex); // Or ma_device_type_playback or ma_device_type_capture. + config.playback.pDeviceID = &myPlaybackDeviceID; // Or NULL for the default playback device. + config.playback.format = ma_format_f32; + config.playback.channels = 2; + config.capture.pDeviceID = &myCaptureDeviceID; // Or NULL for the default capture device. + config.capture.format = ma_format_s16; + config.capture.channels = 1; + config.sampleRate = 44100; + config.dataCallback = data_callback; + config.pUserData = &myUserData; + + result = ma_device_init(&myContext, &config, &device); + if (result != MA_SUCCESS) { + ... handle error ... + } + + Note that the "onDataCallback" member of ma_device_config has been renamed to "dataCallback". Also, "onStopCallback" has + been renamed to "stopCallback". + +This is the first pass for full-duplex and there is a known bug. You will hear crackling on the following backends when sample +rate conversion is required for the playback device: + - Core Audio + - JACK + - AAudio + - OpenSL + - WebAudio + +In addition to the above, not all platforms have been absolutely thoroughly tested simply because I lack the hardware for such +thorough testing. If you experience a bug, an issue report on GitHub or an email would be greatly appreciated (and a sample +program that reproduces the issue if possible). + + +Other API Changes +----------------- +In addition to the above, the following API changes have been made: + +- The log callback is no longer passed to ma_context_config_init(). Instead you need to set it manually after initialization. +- The onLogCallback member of ma_context_config has been renamed to "logCallback". +- The log callback now takes a logLevel parameter. The new callback looks like: void log_callback(ma_context* pContext, ma_device* pDevice, ma_uint32 logLevel, const char* message) + - You can use ma_log_level_to_string() to convert the logLevel to human readable text if you want to log it. +- Some APIs have been renamed: + - mal_decoder_read() -> ma_decoder_read_pcm_frames() + - mal_decoder_seek_to_frame() -> ma_decoder_seek_to_pcm_frame() + - mal_sine_wave_read() -> ma_sine_wave_read_f32() + - mal_sine_wave_read_ex() -> ma_sine_wave_read_f32_ex() +- Some APIs have been removed: + - mal_device_get_buffer_size_in_bytes() + - mal_device_set_recv_callback() + - mal_device_set_send_callback() + - mal_src_set_input_sample_rate() + - mal_src_set_output_sample_rate() +- Error codes have been rearranged. If you're a binding maintainer you will need to update. +- The ma_backend enums have been rearranged to priority order. The rationale for this is to simplify automatic backend selection + and to make it easier to see the priority. If you're a binding maintainer you will need to update. +- ma_dsp has been renamed to ma_pcm_converter. The rationale for this change is that I'm expecting "ma_dsp" to conflict with + some future planned high-level APIs. +- For functions that take a pointer/count combo, such as ma_decoder_read_pcm_frames(), the parameter order has changed so that + the pointer comes before the count. The rationale for this is to keep it consistent with things like memcpy(). + + +Miscellaneous Changes +--------------------- +The following miscellaneous changes have also been made. + +- The AAudio backend has been added for Android 8 and above. This is Android's new "High-Performance Audio" API. (For the + record, this is one of the nicest audio APIs out there, just behind the BSD audio APIs). +- The WebAudio backend has been added. This is based on ScriptProcessorNode. This removes the need for SDL. +- The SDL and OpenAL backends have been removed. These were originally implemented to add support for platforms for which miniaudio + was not explicitly supported. These are no longer needed and have therefore been removed. +- Device initialization now fails if the requested share mode is not supported. If you ask for exclusive mode, you either get an + exclusive mode device, or an error. The rationale for this change is to give the client more control over how to handle cases + when the desired shared mode is unavailable. +- A lock-free ring buffer API has been added. There are two varients of this. "ma_rb" operates on bytes, whereas "ma_pcm_rb" + operates on PCM frames. +- The library is now licensed as a choice of Public Domain (Unlicense) _or_ MIT-0 (No Attribution) which is the same as MIT, but + removes the attribution requirement. The rationale for this is to support countries that don't recognize public domain. */ /* REVISION HISTORY ================ +v0.xx.xx - 2020-xx-xx + - Fix potential crash when ma_device object's are not aligned to MA_SIMD_ALIGNMENT. + +v0.9.10 - 2020-01-15 + - Fix compilation errors due to #if/#endif mismatches. + - WASAPI: Fix a bug where automatic stream routing is being performed for devices that are initialized with an explicit device ID. + - iOS: Fix a crash on device uninitialization. + +v0.9.9 - 2020-01-09 + - Fix compilation errors with MinGW. + - Fix compilation errors when compiling on Apple platforms. + - WASAPI: Add support for disabling hardware offloading. + - WASAPI: Add support for disabling automatic stream routing. + - Core Audio: Fix bugs in the case where the internal device uses deinterleaved buffers. + - Core Audio: Add support for controlling the session category (AVAudioSessionCategory) and options (AVAudioSessionCategoryOptions). + - JACK: Fix bug where incorrect ports are connected. + v0.9.8 - 2019-10-07 - WASAPI: Fix a potential deadlock when starting a full-duplex device. - WASAPI: Enable automatic resampling by default. Disable with config.wasapi.noAutoConvertSRC. @@ -35864,7 +38933,7 @@ For more information, please refer to =============================================================================== ALTERNATIVE 2 - MIT No Attribution =============================================================================== -Copyright 2019 David Reid +Copyright 2020 David Reid Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files (the "Software"), to deal in -- cgit v1.2.3