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author | Luca Sas <sas.luca.alex@gmail.com> | 2020-03-06 17:48:44 +0000 |
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committer | Luca Sas <sas.luca.alex@gmail.com> | 2020-03-06 17:48:44 +0000 |
commit | 581538a8b371c0a9003dc0f1bf081222b8c4fdd9 (patch) | |
tree | f5759a699424211d4a66e24365a596072818ab33 /libs/raylib/src/external/dr_wav.h | |
download | gamejam-slgj-2024-581538a8b371c0a9003dc0f1bf081222b8c4fdd9.tar.gz gamejam-slgj-2024-581538a8b371c0a9003dc0f1bf081222b8c4fdd9.tar.bz2 gamejam-slgj-2024-581538a8b371c0a9003dc0f1bf081222b8c4fdd9.zip |
Setup the project
Diffstat (limited to 'libs/raylib/src/external/dr_wav.h')
-rw-r--r-- | libs/raylib/src/external/dr_wav.h | 4874 |
1 files changed, 4874 insertions, 0 deletions
diff --git a/libs/raylib/src/external/dr_wav.h b/libs/raylib/src/external/dr_wav.h new file mode 100644 index 0000000..7c48ac4 --- /dev/null +++ b/libs/raylib/src/external/dr_wav.h @@ -0,0 +1,4874 @@ +/* +WAV audio loader and writer. Choice of public domain or MIT-0. See license statements at the end of this file. +dr_wav - v0.9.2 - 2019-05-21 + +David Reid - mackron@gmail.com +*/ + +/* +DEPRECATED APIS +=============== +Version 0.9.0 deprecated the per-sample reading and seeking APIs and replaced them with versions that work on the resolution +of a PCM frame instead. For example, given a stereo WAV file, previously you would pass 2 to drwav_read_f32() to read one +PCM frame, whereas now you would pass in 1 to drwav_read_pcm_frames_f32(). The old APIs would return the number of samples +read, whereas now it will return the number of PCM frames. Below is a list of APIs that have been deprecated and their +replacements. + + drwav_read() -> drwav_read_pcm_frames() + drwav_read_s16() -> drwav_read_pcm_frames_s16() + drwav_read_f32() -> drwav_read_pcm_frames_f32() + drwav_read_s32() -> drwav_read_pcm_frames_s32() + drwav_seek_to_sample() -> drwav_seek_to_pcm_frame() + drwav_write() -> drwav_write_pcm_frames() + drwav_open_and_read_s16() -> drwav_open_and_read_pcm_frames_s16() + drwav_open_and_read_f32() -> drwav_open_and_read_pcm_frames_f32() + drwav_open_and_read_s32() -> drwav_open_and_read_pcm_frames_s32() + drwav_open_file_and_read_s16() -> drwav_open_file_and_read_pcm_frames_s16() + drwav_open_file_and_read_f32() -> drwav_open_file_and_read_pcm_frames_f32() + drwav_open_file_and_read_s32() -> drwav_open_file_and_read_pcm_frames_s32() + drwav_open_memory_and_read_s16() -> drwav_open_memory_and_read_pcm_frames_s16() + drwav_open_memory_and_read_f32() -> drwav_open_memory_and_read_pcm_frames_f32() + drwav_open_memory_and_read_s32() -> drwav_open_memory_and_read_pcm_frames_s32() + drwav::totalSampleCount -> drwav::totalPCMFrameCount + +Rationale: + 1) Most programs will want to read in multiples of the channel count which demands a per-frame reading API. Per-sample + reading just adds complexity and maintenance costs for no practical benefit. + 2) This is consistent with my other decoders - dr_flac and dr_mp3. + +These APIs will be removed completely in version 0.10.0. You can continue to use drwav_read_raw() if you need per-sample +reading. +*/ + +/* +USAGE +===== +This is a single-file library. To use it, do something like the following in one .c file. + #define DR_WAV_IMPLEMENTATION + #include "dr_wav.h" + +You can then #include this file in other parts of the program as you would with any other header file. Do something +like the following to read audio data: + + drwav wav; + if (!drwav_init_file(&wav, "my_song.wav")) { + // Error opening WAV file. + } + + drwav_int32* pDecodedInterleavedSamples = malloc(wav.totalPCMFrameCount * wav.channels * sizeof(drwav_int32)); + size_t numberOfSamplesActuallyDecoded = drwav_read_pcm_frames_s32(&wav, wav.totalPCMFrameCount, pDecodedInterleavedSamples); + + ... + + drwav_uninit(&wav); + +You can also use drwav_open() to allocate and initialize the loader for you: + + drwav* pWav = drwav_open_file("my_song.wav"); + if (pWav == NULL) { + // Error opening WAV file. + } + + ... + + drwav_close(pWav); + +If you just want to quickly open and read the audio data in a single operation you can do something like this: + + unsigned int channels; + unsigned int sampleRate; + drwav_uint64 totalPCMFrameCount; + float* pSampleData = drwav_open_file_and_read_pcm_frames_f32("my_song.wav", &channels, &sampleRate, &totalPCMFrameCount); + if (pSampleData == NULL) { + // Error opening and reading WAV file. + } + + ... + + drwav_free(pSampleData); + +The examples above use versions of the API that convert the audio data to a consistent format (32-bit signed PCM, in +this case), but you can still output the audio data in its internal format (see notes below for supported formats): + + size_t samplesRead = drwav_read_pcm_frames(&wav, wav.totalPCMFrameCount, pDecodedInterleavedSamples); + +You can also read the raw bytes of audio data, which could be useful if dr_wav does not have native support for +a particular data format: + + size_t bytesRead = drwav_read_raw(&wav, bytesToRead, pRawDataBuffer); + + +dr_wav can also be used to output WAV files. This does not currently support compressed formats. To use this, look at +drwav_open_write(), drwav_open_file_write(), etc. Use drwav_write_pcm_frames() to write samples, or drwav_write_raw() +to write raw data in the "data" chunk. + + drwav_data_format format; + format.container = drwav_container_riff; // <-- drwav_container_riff = normal WAV files, drwav_container_w64 = Sony Wave64. + format.format = DR_WAVE_FORMAT_PCM; // <-- Any of the DR_WAVE_FORMAT_* codes. + format.channels = 2; + format.sampleRate = 44100; + format.bitsPerSample = 16; + drwav* pWav = drwav_open_file_write("data/recording.wav", &format); + + ... + + drwav_uint64 samplesWritten = drwav_write_pcm_frames(pWav, frameCount, pSamples); + + +dr_wav has seamless support the Sony Wave64 format. The decoder will automatically detect it and it should Just Work +without any manual intervention. + + +OPTIONS +======= +#define these options before including this file. + +#define DR_WAV_NO_CONVERSION_API + Disables conversion APIs such as drwav_read_pcm_frames_f32() and drwav_s16_to_f32(). + +#define DR_WAV_NO_STDIO + Disables drwav_open_file(), drwav_open_file_write(), etc. + + + +QUICK NOTES +=========== +- Samples are always interleaved. +- The default read function does not do any data conversion. Use drwav_read_pcm_frames_f32(), drwav_read_pcm_frames_s32() + and drwav_read_pcm_frames_s16() to read and convert audio data to 32-bit floating point, signed 32-bit integer and + signed 16-bit integer samples respectively. Tested and supported internal formats include the following: + - Unsigned 8-bit PCM + - Signed 12-bit PCM + - Signed 16-bit PCM + - Signed 24-bit PCM + - Signed 32-bit PCM + - IEEE 32-bit floating point + - IEEE 64-bit floating point + - A-law and u-law + - Microsoft ADPCM + - IMA ADPCM (DVI, format code 0x11) +- dr_wav will try to read the WAV file as best it can, even if it's not strictly conformant to the WAV format. +*/ + +#ifndef dr_wav_h +#define dr_wav_h + +#include <stddef.h> + +#if defined(_MSC_VER) && _MSC_VER < 1600 +typedef signed char drwav_int8; +typedef unsigned char drwav_uint8; +typedef signed short drwav_int16; +typedef unsigned short drwav_uint16; +typedef signed int drwav_int32; +typedef unsigned int drwav_uint32; +typedef signed __int64 drwav_int64; +typedef unsigned __int64 drwav_uint64; +#else +#include <stdint.h> +typedef int8_t drwav_int8; +typedef uint8_t drwav_uint8; +typedef int16_t drwav_int16; +typedef uint16_t drwav_uint16; +typedef int32_t drwav_int32; +typedef uint32_t drwav_uint32; +typedef int64_t drwav_int64; +typedef uint64_t drwav_uint64; +#endif +typedef drwav_uint8 drwav_bool8; +typedef drwav_uint32 drwav_bool32; +#define DRWAV_TRUE 1 +#define DRWAV_FALSE 0 + +#ifdef __cplusplus +extern "C" { +#endif + +/* Common data formats. */ +#define DR_WAVE_FORMAT_PCM 0x1 +#define DR_WAVE_FORMAT_ADPCM 0x2 +#define DR_WAVE_FORMAT_IEEE_FLOAT 0x3 +#define DR_WAVE_FORMAT_ALAW 0x6 +#define DR_WAVE_FORMAT_MULAW 0x7 +#define DR_WAVE_FORMAT_DVI_ADPCM 0x11 +#define DR_WAVE_FORMAT_EXTENSIBLE 0xFFFE + +/* Constants. */ +#ifndef DRWAV_MAX_SMPL_LOOPS +#define DRWAV_MAX_SMPL_LOOPS 1 +#endif + +/* Flags to pass into drwav_init_ex(), etc. */ +#define DRWAV_SEQUENTIAL 0x00000001 + +typedef enum +{ + drwav_seek_origin_start, + drwav_seek_origin_current +} drwav_seek_origin; + +typedef enum +{ + drwav_container_riff, + drwav_container_w64 +} drwav_container; + +typedef struct +{ + union + { + drwav_uint8 fourcc[4]; + drwav_uint8 guid[16]; + } id; + + /* The size in bytes of the chunk. */ + drwav_uint64 sizeInBytes; + + /* + RIFF = 2 byte alignment. + W64 = 8 byte alignment. + */ + unsigned int paddingSize; +} drwav_chunk_header; + +/* +Callback for when data is read. Return value is the number of bytes actually read. + +pUserData [in] The user data that was passed to drwav_init(), drwav_open() and family. +pBufferOut [out] The output buffer. +bytesToRead [in] The number of bytes to read. + +Returns the number of bytes actually read. + +A return value of less than bytesToRead indicates the end of the stream. Do _not_ return from this callback until +either the entire bytesToRead is filled or you have reached the end of the stream. +*/ +typedef size_t (* drwav_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead); + +/* +Callback for when data is written. Returns value is the number of bytes actually written. + +pUserData [in] The user data that was passed to drwav_init_write(), drwav_open_write() and family. +pData [out] A pointer to the data to write. +bytesToWrite [in] The number of bytes to write. + +Returns the number of bytes actually written. + +If the return value differs from bytesToWrite, it indicates an error. +*/ +typedef size_t (* drwav_write_proc)(void* pUserData, const void* pData, size_t bytesToWrite); + +/* +Callback for when data needs to be seeked. + +pUserData [in] The user data that was passed to drwav_init(), drwav_open() and family. +offset [in] The number of bytes to move, relative to the origin. Will never be negative. +origin [in] The origin of the seek - the current position or the start of the stream. + +Returns whether or not the seek was successful. + +Whether or not it is relative to the beginning or current position is determined by the "origin" parameter which +will be either drwav_seek_origin_start or drwav_seek_origin_current. +*/ +typedef drwav_bool32 (* drwav_seek_proc)(void* pUserData, int offset, drwav_seek_origin origin); + +/* +Callback for when drwav_init_ex/drwav_open_ex finds a chunk. + +pChunkUserData [in] The user data that was passed to the pChunkUserData parameter of drwav_init_ex(), drwav_open_ex() and family. +onRead [in] A pointer to the function to call when reading. +onSeek [in] A pointer to the function to call when seeking. +pReadSeekUserData [in] The user data that was passed to the pReadSeekUserData parameter of drwav_init_ex(), drwav_open_ex() and family. +pChunkHeader [in] A pointer to an object containing basic header information about the chunk. Use this to identify the chunk. + +Returns the number of bytes read + seeked. + +To read data from the chunk, call onRead(), passing in pReadSeekUserData as the first parameter. Do the same +for seeking with onSeek(). The return value must be the total number of bytes you have read _plus_ seeked. + +You must not attempt to read beyond the boundary of the chunk. +*/ +typedef drwav_uint64 (* drwav_chunk_proc)(void* pChunkUserData, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pReadSeekUserData, const drwav_chunk_header* pChunkHeader); + +/* Structure for internal use. Only used for loaders opened with drwav_open_memory(). */ +typedef struct +{ + const drwav_uint8* data; + size_t dataSize; + size_t currentReadPos; +} drwav__memory_stream; + +/* Structure for internal use. Only used for writers opened with drwav_open_memory_write(). */ +typedef struct +{ + void** ppData; + size_t* pDataSize; + size_t dataSize; + size_t dataCapacity; + size_t currentWritePos; +} drwav__memory_stream_write; + +typedef struct +{ + drwav_container container; /* RIFF, W64. */ + drwav_uint32 format; /* DR_WAVE_FORMAT_* */ + drwav_uint32 channels; + drwav_uint32 sampleRate; + drwav_uint32 bitsPerSample; +} drwav_data_format; + +typedef struct +{ + /* + The format tag exactly as specified in the wave file's "fmt" chunk. This can be used by applications + that require support for data formats not natively supported by dr_wav. + */ + drwav_uint16 formatTag; + + /* The number of channels making up the audio data. When this is set to 1 it is mono, 2 is stereo, etc. */ + drwav_uint16 channels; + + /* The sample rate. Usually set to something like 44100. */ + drwav_uint32 sampleRate; + + /* Average bytes per second. You probably don't need this, but it's left here for informational purposes. */ + drwav_uint32 avgBytesPerSec; + + /* Block align. This is equal to the number of channels * bytes per sample. */ + drwav_uint16 blockAlign; + + /* Bits per sample. */ + drwav_uint16 bitsPerSample; + + /* The size of the extended data. Only used internally for validation, but left here for informational purposes. */ + drwav_uint16 extendedSize; + + /* + The number of valid bits per sample. When <formatTag> is equal to WAVE_FORMAT_EXTENSIBLE, <bitsPerSample> + is always rounded up to the nearest multiple of 8. This variable contains information about exactly how + many bits a valid per sample. Mainly used for informational purposes. + */ + drwav_uint16 validBitsPerSample; + + /* The channel mask. Not used at the moment. */ + drwav_uint32 channelMask; + + /* The sub-format, exactly as specified by the wave file. */ + drwav_uint8 subFormat[16]; +} drwav_fmt; + +typedef struct +{ + drwav_uint32 cuePointId; + drwav_uint32 type; + drwav_uint32 start; + drwav_uint32 end; + drwav_uint32 fraction; + drwav_uint32 playCount; +} drwav_smpl_loop; + + typedef struct +{ + drwav_uint32 manufacturer; + drwav_uint32 product; + drwav_uint32 samplePeriod; + drwav_uint32 midiUnityNotes; + drwav_uint32 midiPitchFraction; + drwav_uint32 smpteFormat; + drwav_uint32 smpteOffset; + drwav_uint32 numSampleLoops; + drwav_uint32 samplerData; + drwav_smpl_loop loops[DRWAV_MAX_SMPL_LOOPS]; +} drwav_smpl; + +typedef struct +{ + /* A pointer to the function to call when more data is needed. */ + drwav_read_proc onRead; + + /* A pointer to the function to call when data needs to be written. Only used when the drwav object is opened in write mode. */ + drwav_write_proc onWrite; + + /* A pointer to the function to call when the wav file needs to be seeked. */ + drwav_seek_proc onSeek; + + /* The user data to pass to callbacks. */ + void* pUserData; + + + /* Whether or not the WAV file is formatted as a standard RIFF file or W64. */ + drwav_container container; + + + /* Structure containing format information exactly as specified by the wav file. */ + drwav_fmt fmt; + + /* The sample rate. Will be set to something like 44100. */ + drwav_uint32 sampleRate; + + /* The number of channels. This will be set to 1 for monaural streams, 2 for stereo, etc. */ + drwav_uint16 channels; + + /* The bits per sample. Will be set to something like 16, 24, etc. */ + drwav_uint16 bitsPerSample; + + /* Equal to fmt.formatTag, or the value specified by fmt.subFormat if fmt.formatTag is equal to 65534 (WAVE_FORMAT_EXTENSIBLE). */ + drwav_uint16 translatedFormatTag; + + /* The total number of PCM frames making up the audio data. */ + drwav_uint64 totalPCMFrameCount; + + + /* The size in bytes of the data chunk. */ + drwav_uint64 dataChunkDataSize; + + /* The position in the stream of the first byte of the data chunk. This is used for seeking. */ + drwav_uint64 dataChunkDataPos; + + /* The number of bytes remaining in the data chunk. */ + drwav_uint64 bytesRemaining; + + + /* + Only used in sequential write mode. Keeps track of the desired size of the "data" chunk at the point of initialization time. Always + set to 0 for non-sequential writes and when the drwav object is opened in read mode. Used for validation. + */ + drwav_uint64 dataChunkDataSizeTargetWrite; + + /* Keeps track of whether or not the wav writer was initialized in sequential mode. */ + drwav_bool32 isSequentialWrite; + + + /* smpl chunk. */ + drwav_smpl smpl; + + + /* A hack to avoid a DRWAV_MALLOC() when opening a decoder with drwav_open_memory(). */ + drwav__memory_stream memoryStream; + drwav__memory_stream_write memoryStreamWrite; + + /* Generic data for compressed formats. This data is shared across all block-compressed formats. */ + struct + { + drwav_uint64 iCurrentSample; /* The index of the next sample that will be read by drwav_read_*(). This is used with "totalSampleCount" to ensure we don't read excess samples at the end of the last block. */ + } compressed; + + /* Microsoft ADPCM specific data. */ + struct + { + drwav_uint32 bytesRemainingInBlock; + drwav_uint16 predictor[2]; + drwav_int32 delta[2]; + drwav_int32 cachedSamples[4]; /* Samples are stored in this cache during decoding. */ + drwav_uint32 cachedSampleCount; + drwav_int32 prevSamples[2][2]; /* The previous 2 samples for each channel (2 channels at most). */ + } msadpcm; + + /* IMA ADPCM specific data. */ + struct + { + drwav_uint32 bytesRemainingInBlock; + drwav_int32 predictor[2]; + drwav_int32 stepIndex[2]; + drwav_int32 cachedSamples[16]; /* Samples are stored in this cache during decoding. */ + drwav_uint32 cachedSampleCount; + } ima; + + + drwav_uint64 totalSampleCount; /* <-- DEPRECATED. Will be removed in a future version. */ +} drwav; + + +/* +Initializes a pre-allocated drwav object. + +pWav [out] A pointer to the drwav object being initialized. +onRead [in] The function to call when data needs to be read from the client. +onSeek [in] The function to call when the read position of the client data needs to move. +onChunk [in, optional] The function to call when a chunk is enumerated at initialized time. +pUserData, pReadSeekUserData [in, optional] A pointer to application defined data that will be passed to onRead and onSeek. +pChunkUserData [in, optional] A pointer to application defined data that will be passed to onChunk. +flags [in, optional] A set of flags for controlling how things are loaded. + +Returns true if successful; false otherwise. + +Close the loader with drwav_uninit(). + +This is the lowest level function for initializing a WAV file. You can also use drwav_init_file() and drwav_init_memory() +to open the stream from a file or from a block of memory respectively. + +If you want dr_wav to manage the memory allocation for you, consider using drwav_open() instead. This will allocate +a drwav object on the heap and return a pointer to it. + +Possible values for flags: + DRWAV_SEQUENTIAL: Never perform a backwards seek while loading. This disables the chunk callback and will cause this function + to return as soon as the data chunk is found. Any chunks after the data chunk will be ignored. + +drwav_init() is equivalent to "drwav_init_ex(pWav, onRead, onSeek, NULL, pUserData, NULL, 0);". + +The onChunk callback is not called for the WAVE or FMT chunks. The contents of the FMT chunk can be read from pWav->fmt +after the function returns. + +See also: drwav_init_file(), drwav_init_memory(), drwav_uninit() +*/ +drwav_bool32 drwav_init(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData); +drwav_bool32 drwav_init_ex(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_chunk_proc onChunk, void* pReadSeekUserData, void* pChunkUserData, drwav_uint32 flags); + +/* +Initializes a pre-allocated drwav object for writing. + +onWrite [in] The function to call when data needs to be written. +onSeek [in] The function to call when the write position needs to move. +pUserData [in, optional] A pointer to application defined data that will be passed to onWrite and onSeek. + +Returns true if successful; false otherwise. + +Close the writer with drwav_uninit(). + +This is the lowest level function for initializing a WAV file. You can also use drwav_init_file() and drwav_init_memory() +to open the stream from a file or from a block of memory respectively. + +If the total sample count is known, you can use drwav_init_write_sequential(). This avoids the need for dr_wav to perform +a post-processing step for storing the total sample count and the size of the data chunk which requires a backwards seek. + +If you want dr_wav to manage the memory allocation for you, consider using drwav_open() instead. This will allocate +a drwav object on the heap and return a pointer to it. + +See also: drwav_init_file_write(), drwav_init_memory_write(), drwav_uninit() +*/ +drwav_bool32 drwav_init_write(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData); +drwav_bool32 drwav_init_write_sequential(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData); + +/* +Uninitializes the given drwav object. + +Use this only for objects initialized with drwav_init(). +*/ +void drwav_uninit(drwav* pWav); + + +/* +Opens a wav file using the given callbacks. + +onRead [in] The function to call when data needs to be read from the client. +onSeek [in] The function to call when the read position of the client data needs to move. +pUserData [in, optional] A pointer to application defined data that will be passed to onRead and onSeek. + +Returns null on error. + +Close the loader with drwav_close(). + +You can also use drwav_open_file() and drwav_open_memory() to open the stream from a file or from a block of +memory respectively. + +This is different from drwav_init() in that it will allocate the drwav object for you via DRWAV_MALLOC() before +initializing it. + +See also: drwav_init(), drwav_open_file(), drwav_open_memory(), drwav_close() +*/ +drwav* drwav_open(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData); +drwav* drwav_open_ex(drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_chunk_proc onChunk, void* pReadSeekUserData, void* pChunkUserData, drwav_uint32 flags); + +/* +Opens a wav file for writing using the given callbacks. + +onWrite [in] The function to call when data needs to be written. +onSeek [in] The function to call when the write position needs to move. +pUserData [in, optional] A pointer to application defined data that will be passed to onWrite and onSeek. + +Returns null on error. + +Close the loader with drwav_close(). + +You can also use drwav_open_file_write() and drwav_open_memory_write() to open the stream from a file or from a block +of memory respectively. + +This is different from drwav_init_write() in that it will allocate the drwav object for you via DRWAV_MALLOC() before +initializing it. + +See also: drwav_open_file_write(), drwav_open_memory_write(), drwav_close() +*/ +drwav* drwav_open_write(const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData); +drwav* drwav_open_write_sequential(const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData); + +/* +Uninitializes and deletes the the given drwav object. + +Use this only for objects created with drwav_open(). +*/ +void drwav_close(drwav* pWav); + + +/* +Reads raw audio data. + +This is the lowest level function for reading audio data. It simply reads the given number of +bytes of the raw internal sample data. + +Consider using drwav_read_pcm_frames_s16(), drwav_read_pcm_frames_s32() or drwav_read_pcm_frames_f32() for +reading sample data in a consistent format. + +Returns the number of bytes actually read. +*/ +size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOut); + +/* +Reads a chunk of audio data in the native internal format. + +This is typically the most efficient way to retrieve audio data, but it does not do any format +conversions which means you'll need to convert the data manually if required. + +If the return value is less than <framesToRead> it means the end of the file has been reached or +you have requested more samples than can possibly fit in the output buffer. + +This function will only work when sample data is of a fixed size and uncompressed. If you are +using a compressed format consider using drwav_read_raw() or drwav_read_pcm_frames_s16/s32/f32/etc(). +*/ +drwav_uint64 drwav_read_pcm_frames(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut); + +/* +Seeks to the given PCM frame. + +Returns true if successful; false otherwise. +*/ +drwav_bool32 drwav_seek_to_pcm_frame(drwav* pWav, drwav_uint64 targetFrameIndex); + + +/* +Writes raw audio data. + +Returns the number of bytes actually written. If this differs from bytesToWrite, it indicates an error. +*/ +size_t drwav_write_raw(drwav* pWav, size_t bytesToWrite, const void* pData); + +/* +Writes PCM frames. + +Returns the number of PCM frames written. +*/ +drwav_uint64 drwav_write_pcm_frames(drwav* pWav, drwav_uint64 framesToWrite, const void* pData); + + +/* Conversion Utilities */ +#ifndef DR_WAV_NO_CONVERSION_API + +/* +Reads a chunk of audio data and converts it to signed 16-bit PCM samples. + +Returns the number of PCM frames actually read. + +If the return value is less than <framesToRead> it means the end of the file has been reached. +*/ +drwav_uint64 drwav_read_pcm_frames_s16(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut); + +/* Low-level function for converting unsigned 8-bit PCM samples to signed 16-bit PCM samples. */ +void drwav_u8_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 24-bit PCM samples to signed 16-bit PCM samples. */ +void drwav_s24_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 32-bit PCM samples to signed 16-bit PCM samples. */ +void drwav_s32_to_s16(drwav_int16* pOut, const drwav_int32* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 32-bit floating point samples to signed 16-bit PCM samples. */ +void drwav_f32_to_s16(drwav_int16* pOut, const float* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 64-bit floating point samples to signed 16-bit PCM samples. */ +void drwav_f64_to_s16(drwav_int16* pOut, const double* pIn, size_t sampleCount); + +/* Low-level function for converting A-law samples to signed 16-bit PCM samples. */ +void drwav_alaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting u-law samples to signed 16-bit PCM samples. */ +void drwav_mulaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + + +/* +Reads a chunk of audio data and converts it to IEEE 32-bit floating point samples. + +Returns the number of PCM frames actually read. + +If the return value is less than <framesToRead> it means the end of the file has been reached. +*/ +drwav_uint64 drwav_read_pcm_frames_f32(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut); + +/* Low-level function for converting unsigned 8-bit PCM samples to IEEE 32-bit floating point samples. */ +void drwav_u8_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 16-bit PCM samples to IEEE 32-bit floating point samples. */ +void drwav_s16_to_f32(float* pOut, const drwav_int16* pIn, size_t sampleCount); + +/* Low-level function for converting signed 24-bit PCM samples to IEEE 32-bit floating point samples. */ +void drwav_s24_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 32-bit PCM samples to IEEE 32-bit floating point samples. */ +void drwav_s32_to_f32(float* pOut, const drwav_int32* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 64-bit floating point samples to IEEE 32-bit floating point samples. */ +void drwav_f64_to_f32(float* pOut, const double* pIn, size_t sampleCount); + +/* Low-level function for converting A-law samples to IEEE 32-bit floating point samples. */ +void drwav_alaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting u-law samples to IEEE 32-bit floating point samples. */ +void drwav_mulaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + + +/* +Reads a chunk of audio data and converts it to signed 32-bit PCM samples. + +Returns the number of PCM frames actually read. + +If the return value is less than <framesToRead> it means the end of the file has been reached. +*/ +drwav_uint64 drwav_read_pcm_frames_s32(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut); + +/* Low-level function for converting unsigned 8-bit PCM samples to signed 32-bit PCM samples. */ +void drwav_u8_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 16-bit PCM samples to signed 32-bit PCM samples. */ +void drwav_s16_to_s32(drwav_int32* pOut, const drwav_int16* pIn, size_t sampleCount); + +/* Low-level function for converting signed 24-bit PCM samples to signed 32-bit PCM samples. */ +void drwav_s24_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 32-bit floating point samples to signed 32-bit PCM samples. */ +void drwav_f32_to_s32(drwav_int32* pOut, const float* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 64-bit floating point samples to signed 32-bit PCM samples. */ +void drwav_f64_to_s32(drwav_int32* pOut, const double* pIn, size_t sampleCount); + +/* Low-level function for converting A-law samples to signed 32-bit PCM samples. */ +void drwav_alaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting u-law samples to signed 32-bit PCM samples. */ +void drwav_mulaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +#endif /* DR_WAV_NO_CONVERSION_API */ + + +/* High-Level Convenience Helpers */ + +#ifndef DR_WAV_NO_STDIO +/* +Helper for initializing a wave file using stdio. + +This holds the internal FILE object until drwav_uninit() is called. Keep this in mind if you're caching drwav +objects because the operating system may restrict the number of file handles an application can have open at +any given time. +*/ +drwav_bool32 drwav_init_file(drwav* pWav, const char* filename); +drwav_bool32 drwav_init_file_ex(drwav* pWav, const char* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags); + +/* +Helper for initializing a wave file for writing using stdio. + +This holds the internal FILE object until drwav_uninit() is called. Keep this in mind if you're caching drwav +objects because the operating system may restrict the number of file handles an application can have open at +any given time. +*/ +drwav_bool32 drwav_init_file_write(drwav* pWav, const char* filename, const drwav_data_format* pFormat); +drwav_bool32 drwav_init_file_write_sequential(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount); + +/* +Helper for opening a wave file using stdio. + +This holds the internal FILE object until drwav_close() is called. Keep this in mind if you're caching drwav +objects because the operating system may restrict the number of file handles an application can have open at +any given time. +*/ +drwav* drwav_open_file(const char* filename); +drwav* drwav_open_file_ex(const char* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags); + +/* +Helper for opening a wave file for writing using stdio. + +This holds the internal FILE object until drwav_close() is called. Keep this in mind if you're caching drwav +objects because the operating system may restrict the number of file handles an application can have open at +any given time. +*/ +drwav* drwav_open_file_write(const char* filename, const drwav_data_format* pFormat); +drwav* drwav_open_file_write_sequential(const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount); + +#endif /* DR_WAV_NO_STDIO */ + +/* +Helper for initializing a loader from a pre-allocated memory buffer. + +This does not create a copy of the data. It is up to the application to ensure the buffer remains valid for +the lifetime of the drwav object. + +The buffer should contain the contents of the entire wave file, not just the sample data. +*/ +drwav_bool32 drwav_init_memory(drwav* pWav, const void* data, size_t dataSize); +drwav_bool32 drwav_init_memory_ex(drwav* pWav, const void* data, size_t dataSize, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags); + +/* +Helper for initializing a writer which outputs data to a memory buffer. + +dr_wav will manage the memory allocations, however it is up to the caller to free the data with drwav_free(). + +The buffer will remain allocated even after drwav_uninit() is called. Indeed, the buffer should not be +considered valid until after drwav_uninit() has been called anyway. +*/ +drwav_bool32 drwav_init_memory_write(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat); +drwav_bool32 drwav_init_memory_write_sequential(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount); + +/* +Helper for opening a loader from a pre-allocated memory buffer. + +This does not create a copy of the data. It is up to the application to ensure the buffer remains valid for +the lifetime of the drwav object. + +The buffer should contain the contents of the entire wave file, not just the sample data. +*/ +drwav* drwav_open_memory(const void* data, size_t dataSize); +drwav* drwav_open_memory_ex(const void* data, size_t dataSize, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags); + +/* +Helper for opening a writer which outputs data to a memory buffer. + +dr_wav will manage the memory allocations, however it is up to the caller to free the data with drwav_free(). + +The buffer will remain allocated even after drwav_close() is called. Indeed, the buffer should not be +considered valid until after drwav_close() has been called anyway. +*/ +drwav* drwav_open_memory_write(void** ppData, size_t* pDataSize, const drwav_data_format* pFormat); +drwav* drwav_open_memory_write_sequential(void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount); + + +#ifndef DR_WAV_NO_CONVERSION_API +/* Opens and reads a wav file in a single operation. */ +drwav_int16* drwav_open_and_read_pcm_frames_s16(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount); +float* drwav_open_and_read_pcm_frames_f32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount); +drwav_int32* drwav_open_and_read_pcm_frames_s32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount); +#ifndef DR_WAV_NO_STDIO +/* Opens and decodes a wav file in a single operation. */ +drwav_int16* drwav_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount); +float* drwav_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount); +drwav_int32* drwav_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount); +#endif + +/* Opens and decodes a wav file from a block of memory in a single operation. */ +drwav_int16* drwav_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount); +float* drwav_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount); +drwav_int32* drwav_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount); +#endif + +/* Frees data that was allocated internally by dr_wav. */ +void drwav_free(void* pDataReturnedByOpenAndRead); + + +/* DEPRECATED APIS */ +drwav_uint64 drwav_read(drwav* pWav, drwav_uint64 samplesToRead, void* pBufferOut); +drwav_uint64 drwav_read_s16(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut); +drwav_uint64 drwav_read_f32(drwav* pWav, drwav_uint64 samplesToRead, float* pBufferOut); +drwav_uint64 drwav_read_s32(drwav* pWav, drwav_uint64 samplesToRead, drwav_int32* pBufferOut); +drwav_bool32 drwav_seek_to_sample(drwav* pWav, drwav_uint64 sample); +drwav_uint64 drwav_write(drwav* pWav, drwav_uint64 samplesToWrite, const void* pData); +#ifndef DR_WAV_NO_CONVERSION_API +drwav_int16* drwav_open_and_read_s16(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount); +float* drwav_open_and_read_f32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount); +drwav_int32* drwav_open_and_read_s32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount); +#ifndef DR_WAV_NO_STDIO +drwav_int16* drwav_open_memory_and_read_s16(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount); +float* drwav_open_file_and_read_f32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount); +drwav_int32* drwav_open_file_and_read_s32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount); +#endif +drwav_int16* drwav_open_memory_and_read_s16(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount); +float* drwav_open_memory_and_read_f32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount); +drwav_int32* drwav_open_memory_and_read_s32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount); +#endif + + +#ifdef __cplusplus +} +#endif +#endif /* dr_wav_h */ + + +/************************************************************************************************************************************************************ + ************************************************************************************************************************************************************ + + IMPLEMENTATION + + ************************************************************************************************************************************************************ + ************************************************************************************************************************************************************/ +#ifdef DR_WAV_IMPLEMENTATION +#include <stdlib.h> +#include <string.h> /* For memcpy(), memset() */ +#include <limits.h> /* For INT_MAX */ + +#ifndef DR_WAV_NO_STDIO +#include <stdio.h> +#endif + +/* Standard library stuff. */ +#ifndef DRWAV_ASSERT +#include <assert.h> +#define DRWAV_ASSERT(expression) assert(expression) +#endif +#ifndef DRWAV_MALLOC +#define DRWAV_MALLOC(sz) malloc((sz)) +#endif +#ifndef DRWAV_REALLOC +#define DRWAV_REALLOC(p, sz) realloc((p), (sz)) +#endif +#ifndef DRWAV_FREE +#define DRWAV_FREE(p) free((p)) +#endif +#ifndef DRWAV_COPY_MEMORY +#define DRWAV_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz)) +#endif +#ifndef DRWAV_ZERO_MEMORY +#define DRWAV_ZERO_MEMORY(p, sz) memset((p), 0, (sz)) +#endif + +#define drwav_countof(x) (sizeof(x) / sizeof(x[0])) +#define drwav_align(x, a) ((((x) + (a) - 1) / (a)) * (a)) +#define drwav_min(a, b) (((a) < (b)) ? (a) : (b)) +#define drwav_max(a, b) (((a) > (b)) ? (a) : (b)) +#define drwav_clamp(x, lo, hi) (drwav_max((lo), drwav_min((hi), (x)))) + +#define drwav_assert DRWAV_ASSERT +#define drwav_copy_memory DRWAV_COPY_MEMORY +#define drwav_zero_memory DRWAV_ZERO_MEMORY + +typedef drwav_int32 drwav_result; +#define DRWAV_SUCCESS 0 +#define DRWAV_ERROR -1 +#define DRWAV_INVALID_ARGS -2 +#define DRWAV_INVALID_OPERATION -3 +#define DRWAV_INVALID_FILE -100 +#define DRWAV_EOF -101 + +#define DRWAV_MAX_SIMD_VECTOR_SIZE 64 /* 64 for AVX-512 in the future. */ + +#ifdef _MSC_VER +#define DRWAV_INLINE __forceinline +#else +#ifdef __GNUC__ +#define DRWAV_INLINE __inline__ __attribute__((always_inline)) +#else +#define DRWAV_INLINE +#endif +#endif + +#if defined(SIZE_MAX) + #define DRWAV_SIZE_MAX SIZE_MAX +#else + #if defined(_WIN64) || defined(_LP64) || defined(__LP64__) + #define DRWAV_SIZE_MAX ((drwav_uint64)0xFFFFFFFFFFFFFFFF) + #else + #define DRWAV_SIZE_MAX 0xFFFFFFFF + #endif +#endif + +static const drwav_uint8 drwavGUID_W64_RIFF[16] = {0x72,0x69,0x66,0x66, 0x2E,0x91, 0xCF,0x11, 0xA5,0xD6, 0x28,0xDB,0x04,0xC1,0x00,0x00}; /* 66666972-912E-11CF-A5D6-28DB04C10000 */ +static const drwav_uint8 drwavGUID_W64_WAVE[16] = {0x77,0x61,0x76,0x65, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 65766177-ACF3-11D3-8CD1-00C04F8EDB8A */ +static const drwav_uint8 drwavGUID_W64_JUNK[16] = {0x6A,0x75,0x6E,0x6B, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 6B6E756A-ACF3-11D3-8CD1-00C04F8EDB8A */ +static const drwav_uint8 drwavGUID_W64_FMT [16] = {0x66,0x6D,0x74,0x20, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 20746D66-ACF3-11D3-8CD1-00C04F8EDB8A */ +static const drwav_uint8 drwavGUID_W64_FACT[16] = {0x66,0x61,0x63,0x74, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 74636166-ACF3-11D3-8CD1-00C04F8EDB8A */ +static const drwav_uint8 drwavGUID_W64_DATA[16] = {0x64,0x61,0x74,0x61, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 61746164-ACF3-11D3-8CD1-00C04F8EDB8A */ +static const drwav_uint8 drwavGUID_W64_SMPL[16] = {0x73,0x6D,0x70,0x6C, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 6C706D73-ACF3-11D3-8CD1-00C04F8EDB8A */ + +static DRWAV_INLINE drwav_bool32 drwav__guid_equal(const drwav_uint8 a[16], const drwav_uint8 b[16]) +{ + const drwav_uint32* a32 = (const drwav_uint32*)a; + const drwav_uint32* b32 = (const drwav_uint32*)b; + + return + a32[0] == b32[0] && + a32[1] == b32[1] && + a32[2] == b32[2] && + a32[3] == b32[3]; +} + +static DRWAV_INLINE drwav_bool32 drwav__fourcc_equal(const unsigned char* a, const char* b) +{ + return + a[0] == b[0] && + a[1] == b[1] && + a[2] == b[2] && + a[3] == b[3]; +} + + + +static DRWAV_INLINE int drwav__is_little_endian() +{ + int n = 1; + return (*(char*)&n) == 1; +} + +static DRWAV_INLINE unsigned short drwav__bytes_to_u16(const unsigned char* data) +{ + return (data[0] << 0) | (data[1] << 8); +} + +static DRWAV_INLINE short drwav__bytes_to_s16(const unsigned char* data) +{ + return (short)drwav__bytes_to_u16(data); +} + +static DRWAV_INLINE unsigned int drwav__bytes_to_u32(const unsigned char* data) +{ + return (data[0] << 0) | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); +} + +static DRWAV_INLINE drwav_uint64 drwav__bytes_to_u64(const unsigned char* data) +{ + return + ((drwav_uint64)data[0] << 0) | ((drwav_uint64)data[1] << 8) | ((drwav_uint64)data[2] << 16) | ((drwav_uint64)data[3] << 24) | + ((drwav_uint64)data[4] << 32) | ((drwav_uint64)data[5] << 40) | ((drwav_uint64)data[6] << 48) | ((drwav_uint64)data[7] << 56); +} + +static DRWAV_INLINE void drwav__bytes_to_guid(const unsigned char* data, drwav_uint8* guid) +{ + int i; + for (i = 0; i < 16; ++i) { + guid[i] = data[i]; + } +} + + +static DRWAV_INLINE drwav_bool32 drwav__is_compressed_format_tag(drwav_uint16 formatTag) +{ + return + formatTag == DR_WAVE_FORMAT_ADPCM || + formatTag == DR_WAVE_FORMAT_DVI_ADPCM; +} + +drwav_uint64 drwav_read_s16__msadpcm(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut); +drwav_uint64 drwav_read_s16__ima(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut); +drwav_bool32 drwav_init_write__internal(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData); +drwav* drwav_open_write__internal(const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData); + +static drwav_result drwav__read_chunk_header(drwav_read_proc onRead, void* pUserData, drwav_container container, drwav_uint64* pRunningBytesReadOut, drwav_chunk_header* pHeaderOut) +{ + if (container == drwav_container_riff) { + unsigned char sizeInBytes[4]; + + if (onRead(pUserData, pHeaderOut->id.fourcc, 4) != 4) { + return DRWAV_EOF; + } + + if (onRead(pUserData, sizeInBytes, 4) != 4) { + return DRWAV_INVALID_FILE; + } + + pHeaderOut->sizeInBytes = drwav__bytes_to_u32(sizeInBytes); + pHeaderOut->paddingSize = (unsigned int)(pHeaderOut->sizeInBytes % 2); + *pRunningBytesReadOut += 8; + } else { + unsigned char sizeInBytes[8]; + + if (onRead(pUserData, pHeaderOut->id.guid, 16) != 16) { + return DRWAV_EOF; + } + + if (onRead(pUserData, sizeInBytes, 8) != 8) { + return DRWAV_INVALID_FILE; + } + + pHeaderOut->sizeInBytes = drwav__bytes_to_u64(sizeInBytes) - 24; /* <-- Subtract 24 because w64 includes the size of the header. */ + pHeaderOut->paddingSize = (unsigned int)(pHeaderOut->sizeInBytes % 8); + *pRunningBytesReadOut += 24; + } + + return DRWAV_SUCCESS; +} + +static drwav_bool32 drwav__seek_forward(drwav_seek_proc onSeek, drwav_uint64 offset, void* pUserData) +{ + drwav_uint64 bytesRemainingToSeek = offset; + while (bytesRemainingToSeek > 0) { + if (bytesRemainingToSeek > 0x7FFFFFFF) { + if (!onSeek(pUserData, 0x7FFFFFFF, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + bytesRemainingToSeek -= 0x7FFFFFFF; + } else { + if (!onSeek(pUserData, (int)bytesRemainingToSeek, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + bytesRemainingToSeek = 0; + } + } + + return DRWAV_TRUE; +} + +static drwav_bool32 drwav__seek_from_start(drwav_seek_proc onSeek, drwav_uint64 offset, void* pUserData) +{ + if (offset <= 0x7FFFFFFF) { + return onSeek(pUserData, (int)offset, drwav_seek_origin_start); + } + + /* Larger than 32-bit seek. */ + if (!onSeek(pUserData, 0x7FFFFFFF, drwav_seek_origin_start)) { + return DRWAV_FALSE; + } + offset -= 0x7FFFFFFF; + + for (;;) { + if (offset <= 0x7FFFFFFF) { + return onSeek(pUserData, (int)offset, drwav_seek_origin_current); + } + + if (!onSeek(pUserData, 0x7FFFFFFF, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + offset -= 0x7FFFFFFF; + } + + /* Should never get here. */ + /*return DRWAV_TRUE; */ +} + + +static drwav_bool32 drwav__read_fmt(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, drwav_container container, drwav_uint64* pRunningBytesReadOut, drwav_fmt* fmtOut) +{ + drwav_chunk_header header; + unsigned char fmt[16]; + + if (drwav__read_chunk_header(onRead, pUserData, container, pRunningBytesReadOut, &header) != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + + /* Skip non-fmt chunks. */ + while ((container == drwav_container_riff && !drwav__fourcc_equal(header.id.fourcc, "fmt ")) || (container == drwav_container_w64 && !drwav__guid_equal(header.id.guid, drwavGUID_W64_FMT))) { + if (!drwav__seek_forward(onSeek, header.sizeInBytes + header.paddingSize, pUserData)) { + return DRWAV_FALSE; + } + *pRunningBytesReadOut += header.sizeInBytes + header.paddingSize; + + /* Try the next header. */ + if (drwav__read_chunk_header(onRead, pUserData, container, pRunningBytesReadOut, &header) != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + } + + + /* Validation. */ + if (container == drwav_container_riff) { + if (!drwav__fourcc_equal(header.id.fourcc, "fmt ")) { + return DRWAV_FALSE; + } + } else { + if (!drwav__guid_equal(header.id.guid, drwavGUID_W64_FMT)) { + return DRWAV_FALSE; + } + } + + + if (onRead(pUserData, fmt, sizeof(fmt)) != sizeof(fmt)) { + return DRWAV_FALSE; + } + *pRunningBytesReadOut += sizeof(fmt); + + fmtOut->formatTag = drwav__bytes_to_u16(fmt + 0); + fmtOut->channels = drwav__bytes_to_u16(fmt + 2); + fmtOut->sampleRate = drwav__bytes_to_u32(fmt + 4); + fmtOut->avgBytesPerSec = drwav__bytes_to_u32(fmt + 8); + fmtOut->blockAlign = drwav__bytes_to_u16(fmt + 12); + fmtOut->bitsPerSample = drwav__bytes_to_u16(fmt + 14); + + fmtOut->extendedSize = 0; + fmtOut->validBitsPerSample = 0; + fmtOut->channelMask = 0; + memset(fmtOut->subFormat, 0, sizeof(fmtOut->subFormat)); + + if (header.sizeInBytes > 16) { + unsigned char fmt_cbSize[2]; + int bytesReadSoFar = 0; + + if (onRead(pUserData, fmt_cbSize, sizeof(fmt_cbSize)) != sizeof(fmt_cbSize)) { + return DRWAV_FALSE; /* Expecting more data. */ + } + *pRunningBytesReadOut += sizeof(fmt_cbSize); + + bytesReadSoFar = 18; + + fmtOut->extendedSize = drwav__bytes_to_u16(fmt_cbSize); + if (fmtOut->extendedSize > 0) { + /* Simple validation. */ + if (fmtOut->formatTag == DR_WAVE_FORMAT_EXTENSIBLE) { + if (fmtOut->extendedSize != 22) { + return DRWAV_FALSE; + } + } + + if (fmtOut->formatTag == DR_WAVE_FORMAT_EXTENSIBLE) { + unsigned char fmtext[22]; + if (onRead(pUserData, fmtext, fmtOut->extendedSize) != fmtOut->extendedSize) { + return DRWAV_FALSE; /* Expecting more data. */ + } + + fmtOut->validBitsPerSample = drwav__bytes_to_u16(fmtext + 0); + fmtOut->channelMask = drwav__bytes_to_u32(fmtext + 2); + drwav__bytes_to_guid(fmtext + 6, fmtOut->subFormat); + } else { + if (!onSeek(pUserData, fmtOut->extendedSize, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + } + *pRunningBytesReadOut += fmtOut->extendedSize; + + bytesReadSoFar += fmtOut->extendedSize; + } + + /* Seek past any leftover bytes. For w64 the leftover will be defined based on the chunk size. */ + if (!onSeek(pUserData, (int)(header.sizeInBytes - bytesReadSoFar), drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + *pRunningBytesReadOut += (header.sizeInBytes - bytesReadSoFar); + } + + if (header.paddingSize > 0) { + if (!onSeek(pUserData, header.paddingSize, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + *pRunningBytesReadOut += header.paddingSize; + } + + return DRWAV_TRUE; +} + + +#ifndef DR_WAV_NO_STDIO +FILE* drwav_fopen(const char* filePath, const char* openMode) +{ + FILE* pFile; +#if defined(_MSC_VER) && _MSC_VER >= 1400 + if (fopen_s(&pFile, filePath, openMode) != 0) { + return DRWAV_FALSE; + } +#else + pFile = fopen(filePath, openMode); + if (pFile == NULL) { + return DRWAV_FALSE; + } +#endif + + return pFile; +} + +static size_t drwav__on_read_stdio(void* pUserData, void* pBufferOut, size_t bytesToRead) +{ + return fread(pBufferOut, 1, bytesToRead, (FILE*)pUserData); +} + +static size_t drwav__on_write_stdio(void* pUserData, const void* pData, size_t bytesToWrite) +{ + return fwrite(pData, 1, bytesToWrite, (FILE*)pUserData); +} + +static drwav_bool32 drwav__on_seek_stdio(void* pUserData, int offset, drwav_seek_origin origin) +{ + return fseek((FILE*)pUserData, offset, (origin == drwav_seek_origin_current) ? SEEK_CUR : SEEK_SET) == 0; +} + +drwav_bool32 drwav_init_file(drwav* pWav, const char* filename) +{ + return drwav_init_file_ex(pWav, filename, NULL, NULL, 0); +} + +drwav_bool32 drwav_init_file_ex(drwav* pWav, const char* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags) +{ + FILE* pFile = drwav_fopen(filename, "rb"); + if (pFile == NULL) { + return DRWAV_FALSE; + } + + return drwav_init_ex(pWav, drwav__on_read_stdio, drwav__on_seek_stdio, onChunk, (void*)pFile, pChunkUserData, flags); +} + + +drwav_bool32 drwav_init_file_write__internal(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential) +{ + FILE* pFile = drwav_fopen(filename, "wb"); + if (pFile == NULL) { + return DRWAV_FALSE; + } + + return drwav_init_write__internal(pWav, pFormat, totalSampleCount, isSequential, drwav__on_write_stdio, drwav__on_seek_stdio, (void*)pFile); +} + +drwav_bool32 drwav_init_file_write(drwav* pWav, const char* filename, const drwav_data_format* pFormat) +{ + return drwav_init_file_write__internal(pWav, filename, pFormat, 0, DRWAV_FALSE); +} + +drwav_bool32 drwav_init_file_write_sequential(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount) +{ + return drwav_init_file_write__internal(pWav, filename, pFormat, totalSampleCount, DRWAV_TRUE); +} + +drwav* drwav_open_file(const char* filename) +{ + return drwav_open_file_ex(filename, NULL, NULL, 0); +} + +drwav* drwav_open_file_ex(const char* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags) +{ + FILE* pFile; + drwav* pWav; + + pFile = drwav_fopen(filename, "rb"); + if (pFile == NULL) { + return DRWAV_FALSE; + } + + pWav = drwav_open_ex(drwav__on_read_stdio, drwav__on_seek_stdio, onChunk, (void*)pFile, pChunkUserData, flags); + if (pWav == NULL) { + fclose(pFile); + return NULL; + } + + return pWav; +} + + +drwav* drwav_open_file_write__internal(const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential) +{ + FILE* pFile; + drwav* pWav; + + pFile = drwav_fopen(filename, "wb"); + if (pFile == NULL) { + return DRWAV_FALSE; + } + + pWav = drwav_open_write__internal(pFormat, totalSampleCount, isSequential, drwav__on_write_stdio, drwav__on_seek_stdio, (void*)pFile); + if (pWav == NULL) { + fclose(pFile); + return NULL; + } + + return pWav; +} + +drwav* drwav_open_file_write(const char* filename, const drwav_data_format* pFormat) +{ + return drwav_open_file_write__internal(filename, pFormat, 0, DRWAV_FALSE); +} + +drwav* drwav_open_file_write_sequential(const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount) +{ + return drwav_open_file_write__internal(filename, pFormat, totalSampleCount, DRWAV_TRUE); +} +#endif /* DR_WAV_NO_STDIO */ + + +static size_t drwav__on_read_memory(void* pUserData, void* pBufferOut, size_t bytesToRead) +{ + drwav__memory_stream* memory = (drwav__memory_stream*)pUserData; + size_t bytesRemaining; + + drwav_assert(memory != NULL); + drwav_assert(memory->dataSize >= memory->currentReadPos); + + bytesRemaining = memory->dataSize - memory->currentReadPos; + if (bytesToRead > bytesRemaining) { + bytesToRead = bytesRemaining; + } + + if (bytesToRead > 0) { + DRWAV_COPY_MEMORY(pBufferOut, memory->data + memory->currentReadPos, bytesToRead); + memory->currentReadPos += bytesToRead; + } + + return bytesToRead; +} + +static drwav_bool32 drwav__on_seek_memory(void* pUserData, int offset, drwav_seek_origin origin) +{ + drwav__memory_stream* memory = (drwav__memory_stream*)pUserData; + drwav_assert(memory != NULL); + + if (origin == drwav_seek_origin_current) { + if (offset > 0) { + if (memory->currentReadPos + offset > memory->dataSize) { + return DRWAV_FALSE; /* Trying to seek too far forward. */ + } + } else { + if (memory->currentReadPos < (size_t)-offset) { + return DRWAV_FALSE; /* Trying to seek too far backwards. */ + } + } + + /* This will never underflow thanks to the clamps above. */ + memory->currentReadPos += offset; + } else { + if ((drwav_uint32)offset <= memory->dataSize) { + memory->currentReadPos = offset; + } else { + return DRWAV_FALSE; /* Trying to seek too far forward. */ + } + } + + return DRWAV_TRUE; +} + +static size_t drwav__on_write_memory(void* pUserData, const void* pDataIn, size_t bytesToWrite) +{ + drwav__memory_stream_write* memory = (drwav__memory_stream_write*)pUserData; + size_t bytesRemaining; + + drwav_assert(memory != NULL); + drwav_assert(memory->dataCapacity >= memory->currentWritePos); + + bytesRemaining = memory->dataCapacity - memory->currentWritePos; + if (bytesRemaining < bytesToWrite) { + /* Need to reallocate. */ + void* pNewData; + size_t newDataCapacity = (memory->dataCapacity == 0) ? 256 : memory->dataCapacity * 2; + + /* If doubling wasn't enough, just make it the minimum required size to write the data. */ + if ((newDataCapacity - memory->currentWritePos) < bytesToWrite) { + newDataCapacity = memory->currentWritePos + bytesToWrite; + } + + pNewData = DRWAV_REALLOC(*memory->ppData, newDataCapacity); + if (pNewData == NULL) { + return 0; + } + + *memory->ppData = pNewData; + memory->dataCapacity = newDataCapacity; + } + + DRWAV_COPY_MEMORY(((drwav_uint8*)(*memory->ppData)) + memory->currentWritePos, pDataIn, bytesToWrite); + + memory->currentWritePos += bytesToWrite; + if (memory->dataSize < memory->currentWritePos) { + memory->dataSize = memory->currentWritePos; + } + + *memory->pDataSize = memory->dataSize; + + return bytesToWrite; +} + +static drwav_bool32 drwav__on_seek_memory_write(void* pUserData, int offset, drwav_seek_origin origin) +{ + drwav__memory_stream_write* memory = (drwav__memory_stream_write*)pUserData; + drwav_assert(memory != NULL); + + if (origin == drwav_seek_origin_current) { + if (offset > 0) { + if (memory->currentWritePos + offset > memory->dataSize) { + offset = (int)(memory->dataSize - memory->currentWritePos); /* Trying to seek too far forward. */ + } + } else { + if (memory->currentWritePos < (size_t)-offset) { + offset = -(int)memory->currentWritePos; /* Trying to seek too far backwards. */ + } + } + + /* This will never underflow thanks to the clamps above. */ + memory->currentWritePos += offset; + } else { + if ((drwav_uint32)offset <= memory->dataSize) { + memory->currentWritePos = offset; + } else { + memory->currentWritePos = memory->dataSize; /* Trying to seek too far forward. */ + } + } + + return DRWAV_TRUE; +} + +drwav_bool32 drwav_init_memory(drwav* pWav, const void* data, size_t dataSize) +{ + return drwav_init_memory_ex(pWav, data, dataSize, NULL, NULL, 0); +} + +drwav_bool32 drwav_init_memory_ex(drwav* pWav, const void* data, size_t dataSize, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags) +{ + drwav__memory_stream memoryStream; + + if (data == NULL || dataSize == 0) { + return DRWAV_FALSE; + } + + drwav_zero_memory(&memoryStream, sizeof(memoryStream)); + memoryStream.data = (const unsigned char*)data; + memoryStream.dataSize = dataSize; + memoryStream.currentReadPos = 0; + + if (!drwav_init_ex(pWav, drwav__on_read_memory, drwav__on_seek_memory, onChunk, (void*)&memoryStream, pChunkUserData, flags)) { + return DRWAV_FALSE; + } + + pWav->memoryStream = memoryStream; + pWav->pUserData = &pWav->memoryStream; + return DRWAV_TRUE; +} + + +drwav_bool32 drwav_init_memory_write__internal(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential) +{ + drwav__memory_stream_write memoryStreamWrite; + + if (ppData == NULL) { + return DRWAV_FALSE; + } + + *ppData = NULL; /* Important because we're using realloc()! */ + *pDataSize = 0; + + drwav_zero_memory(&memoryStreamWrite, sizeof(memoryStreamWrite)); + memoryStreamWrite.ppData = ppData; + memoryStreamWrite.pDataSize = pDataSize; + memoryStreamWrite.dataSize = 0; + memoryStreamWrite.dataCapacity = 0; + memoryStreamWrite.currentWritePos = 0; + + if (!drwav_init_write__internal(pWav, pFormat, totalSampleCount, isSequential, drwav__on_write_memory, drwav__on_seek_memory_write, (void*)&memoryStreamWrite)) { + return DRWAV_FALSE; + } + + pWav->memoryStreamWrite = memoryStreamWrite; + pWav->pUserData = &pWav->memoryStreamWrite; + return DRWAV_TRUE; +} + +drwav_bool32 drwav_init_memory_write(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat) +{ + return drwav_init_memory_write__internal(pWav, ppData, pDataSize, pFormat, 0, DRWAV_FALSE); +} + +drwav_bool32 drwav_init_memory_write_sequential(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount) +{ + return drwav_init_memory_write__internal(pWav, ppData, pDataSize, pFormat, totalSampleCount, DRWAV_TRUE); +} + + +drwav* drwav_open_memory(const void* data, size_t dataSize) +{ + return drwav_open_memory_ex(data, dataSize, NULL, NULL, 0); +} + +drwav* drwav_open_memory_ex(const void* data, size_t dataSize, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags) +{ + drwav__memory_stream memoryStream; + drwav* pWav; + + if (data == NULL || dataSize == 0) { + return NULL; + } + + drwav_zero_memory(&memoryStream, sizeof(memoryStream)); + memoryStream.data = (const unsigned char*)data; + memoryStream.dataSize = dataSize; + memoryStream.currentReadPos = 0; + + pWav = drwav_open_ex(drwav__on_read_memory, drwav__on_seek_memory, onChunk, (void*)&memoryStream, pChunkUserData, flags); + if (pWav == NULL) { + return NULL; + } + + pWav->memoryStream = memoryStream; + pWav->pUserData = &pWav->memoryStream; + return pWav; +} + + +drwav* drwav_open_memory_write__internal(void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential) +{ + drwav__memory_stream_write memoryStreamWrite; + drwav* pWav; + + if (ppData == NULL) { + return NULL; + } + + *ppData = NULL; /* Important because we're using realloc()! */ + *pDataSize = 0; + + drwav_zero_memory(&memoryStreamWrite, sizeof(memoryStreamWrite)); + memoryStreamWrite.ppData = ppData; + memoryStreamWrite.pDataSize = pDataSize; + memoryStreamWrite.dataSize = 0; + memoryStreamWrite.dataCapacity = 0; + memoryStreamWrite.currentWritePos = 0; + + pWav = drwav_open_write__internal(pFormat, totalSampleCount, isSequential, drwav__on_write_memory, drwav__on_seek_memory_write, (void*)&memoryStreamWrite); + if (pWav == NULL) { + return NULL; + } + + pWav->memoryStreamWrite = memoryStreamWrite; + pWav->pUserData = &pWav->memoryStreamWrite; + return pWav; +} + +drwav* drwav_open_memory_write(void** ppData, size_t* pDataSize, const drwav_data_format* pFormat) +{ + return drwav_open_memory_write__internal(ppData, pDataSize, pFormat, 0, DRWAV_FALSE); +} + +drwav* drwav_open_memory_write_sequential(void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount) +{ + return drwav_open_memory_write__internal(ppData, pDataSize, pFormat, totalSampleCount, DRWAV_TRUE); +} + + +size_t drwav__on_read(drwav_read_proc onRead, void* pUserData, void* pBufferOut, size_t bytesToRead, drwav_uint64* pCursor) +{ + size_t bytesRead; + + drwav_assert(onRead != NULL); + drwav_assert(pCursor != NULL); + + bytesRead = onRead(pUserData, pBufferOut, bytesToRead); + *pCursor += bytesRead; + return bytesRead; +} + +drwav_bool32 drwav__on_seek(drwav_seek_proc onSeek, void* pUserData, int offset, drwav_seek_origin origin, drwav_uint64* pCursor) +{ + drwav_assert(onSeek != NULL); + drwav_assert(pCursor != NULL); + + if (!onSeek(pUserData, offset, origin)) { + return DRWAV_FALSE; + } + + if (origin == drwav_seek_origin_start) { + *pCursor = offset; + } else { + *pCursor += offset; + } + + return DRWAV_TRUE; +} + + +static drwav_uint32 drwav_get_bytes_per_sample(drwav* pWav) +{ + /* + The number of bytes per sample is based on the bits per sample or the block align. We prioritize floor(bitsPerSample/8), but if + this is zero or the bits per sample is not a multiple of 8 we need to fall back to the block align. + */ + drwav_uint32 bytesPerSample = pWav->bitsPerSample >> 3; + if (bytesPerSample == 0 || (pWav->bitsPerSample & 0x7) != 0) { + bytesPerSample = pWav->fmt.blockAlign/pWav->fmt.channels; + } + + return bytesPerSample; +} + +static drwav_uint32 drwav_get_bytes_per_pcm_frame(drwav* pWav) +{ + /* + The number of bytes per frame is based on the bits per sample or the block align. We prioritize floor(bitsPerSample*channels/8), but if + this is zero or the bits per frame is not a multiple of 8 we need to fall back to the block align. + */ + drwav_uint32 bitsPerFrame = pWav->bitsPerSample * pWav->fmt.channels; + drwav_uint32 bytesPerFrame = bitsPerFrame >> 3; + if (bytesPerFrame == 0 || (bitsPerFrame & 0x7) != 0) { + bytesPerFrame = pWav->fmt.blockAlign; + } + + return bytesPerFrame; +} + + +drwav_bool32 drwav_init(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData) +{ + return drwav_init_ex(pWav, onRead, onSeek, NULL, pUserData, NULL, 0); +} + +drwav_bool32 drwav_init_ex(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_chunk_proc onChunk, void* pReadSeekUserData, void* pChunkUserData, drwav_uint32 flags) +{ + drwav_uint64 cursor; /* <-- Keeps track of the byte position so we can seek to specific locations. */ + drwav_bool32 sequential; + unsigned char riff[4]; + drwav_fmt fmt; + unsigned short translatedFormatTag; + drwav_uint64 sampleCountFromFactChunk; + drwav_bool32 foundDataChunk; + drwav_uint64 dataChunkSize; + drwav_uint64 chunkSize; + + if (onRead == NULL || onSeek == NULL) { + return DRWAV_FALSE; + } + + cursor = 0; + sequential = (flags & DRWAV_SEQUENTIAL) != 0; + + drwav_zero_memory(pWav, sizeof(*pWav)); + pWav->onRead = onRead; + pWav->onSeek = onSeek; + pWav->pUserData = pReadSeekUserData; + + /* The first 4 bytes should be the RIFF identifier. */ + if (drwav__on_read(onRead, pReadSeekUserData, riff, sizeof(riff), &cursor) != sizeof(riff)) { + return DRWAV_FALSE; + } + + /* + The first 4 bytes can be used to identify the container. For RIFF files it will start with "RIFF" and for + w64 it will start with "riff". + */ + if (drwav__fourcc_equal(riff, "RIFF")) { + pWav->container = drwav_container_riff; + } else if (drwav__fourcc_equal(riff, "riff")) { + int i; + drwav_uint8 riff2[12]; + + pWav->container = drwav_container_w64; + + /* Check the rest of the GUID for validity. */ + if (drwav__on_read(onRead, pReadSeekUserData, riff2, sizeof(riff2), &cursor) != sizeof(riff2)) { + return DRWAV_FALSE; + } + + for (i = 0; i < 12; ++i) { + if (riff2[i] != drwavGUID_W64_RIFF[i+4]) { + return DRWAV_FALSE; + } + } + } else { + return DRWAV_FALSE; /* Unknown or unsupported container. */ + } + + + if (pWav->container == drwav_container_riff) { + unsigned char chunkSizeBytes[4]; + unsigned char wave[4]; + + /* RIFF/WAVE */ + if (drwav__on_read(onRead, pReadSeekUserData, chunkSizeBytes, sizeof(chunkSizeBytes), &cursor) != sizeof(chunkSizeBytes)) { + return DRWAV_FALSE; + } + + if (drwav__bytes_to_u32(chunkSizeBytes) < 36) { + return DRWAV_FALSE; /* Chunk size should always be at least 36 bytes. */ + } + + if (drwav__on_read(onRead, pReadSeekUserData, wave, sizeof(wave), &cursor) != sizeof(wave)) { + return DRWAV_FALSE; + } + + if (!drwav__fourcc_equal(wave, "WAVE")) { + return DRWAV_FALSE; /* Expecting "WAVE". */ + } + } else { + unsigned char chunkSizeBytes[8]; + drwav_uint8 wave[16]; + + /* W64 */ + if (drwav__on_read(onRead, pReadSeekUserData, chunkSizeBytes, sizeof(chunkSizeBytes), &cursor) != sizeof(chunkSizeBytes)) { + return DRWAV_FALSE; + } + + if (drwav__bytes_to_u64(chunkSizeBytes) < 80) { + return DRWAV_FALSE; + } + + if (drwav__on_read(onRead, pReadSeekUserData, wave, sizeof(wave), &cursor) != sizeof(wave)) { + return DRWAV_FALSE; + } + + if (!drwav__guid_equal(wave, drwavGUID_W64_WAVE)) { + return DRWAV_FALSE; + } + } + + + /* The next bytes should be the "fmt " chunk. */ + if (!drwav__read_fmt(onRead, onSeek, pReadSeekUserData, pWav->container, &cursor, &fmt)) { + return DRWAV_FALSE; /* Failed to read the "fmt " chunk. */ + } + + /* Basic validation. */ + if (fmt.sampleRate == 0 || fmt.channels == 0 || fmt.bitsPerSample == 0 || fmt.blockAlign == 0) { + return DRWAV_FALSE; /* Invalid channel count. Probably an invalid WAV file. */ + } + + + /* Translate the internal format. */ + translatedFormatTag = fmt.formatTag; + if (translatedFormatTag == DR_WAVE_FORMAT_EXTENSIBLE) { + translatedFormatTag = drwav__bytes_to_u16(fmt.subFormat + 0); + } + + + + sampleCountFromFactChunk = 0; + + /* + We need to enumerate over each chunk for two reasons: + 1) The "data" chunk may not be the next one + 2) We may want to report each chunk back to the client + + In order to correctly report each chunk back to the client we will need to keep looping until the end of the file. + */ + foundDataChunk = DRWAV_FALSE; + dataChunkSize = 0; + + /* The next chunk we care about is the "data" chunk. This is not necessarily the next chunk so we'll need to loop. */ + chunkSize = 0; + for (;;) + { + drwav_chunk_header header; + drwav_result result = drwav__read_chunk_header(onRead, pReadSeekUserData, pWav->container, &cursor, &header); + if (result != DRWAV_SUCCESS) { + if (!foundDataChunk) { + return DRWAV_FALSE; + } else { + break; /* Probably at the end of the file. Get out of the loop. */ + } + } + + /* Tell the client about this chunk. */ + if (!sequential && onChunk != NULL) { + drwav_uint64 callbackBytesRead = onChunk(pChunkUserData, onRead, onSeek, pReadSeekUserData, &header); + + /* + dr_wav may need to read the contents of the chunk, so we now need to seek back to the position before + we called the callback. + */ + if (callbackBytesRead > 0) { + if (!drwav__seek_from_start(onSeek, cursor, pReadSeekUserData)) { + return DRWAV_FALSE; + } + } + } + + + if (!foundDataChunk) { + pWav->dataChunkDataPos = cursor; + } + + chunkSize = header.sizeInBytes; + if (pWav->container == drwav_container_riff) { + if (drwav__fourcc_equal(header.id.fourcc, "data")) { + foundDataChunk = DRWAV_TRUE; + dataChunkSize = chunkSize; + } + } else { + if (drwav__guid_equal(header.id.guid, drwavGUID_W64_DATA)) { + foundDataChunk = DRWAV_TRUE; + dataChunkSize = chunkSize; + } + } + + /* + If at this point we have found the data chunk and we're running in sequential mode, we need to break out of this loop. The reason for + this is that we would otherwise require a backwards seek which sequential mode forbids. + */ + if (foundDataChunk && sequential) { + break; + } + + /* Optional. Get the total sample count from the FACT chunk. This is useful for compressed formats. */ + if (pWav->container == drwav_container_riff) { + if (drwav__fourcc_equal(header.id.fourcc, "fact")) { + drwav_uint32 sampleCount; + if (drwav__on_read(onRead, pReadSeekUserData, &sampleCount, 4, &cursor) != 4) { + return DRWAV_FALSE; + } + chunkSize -= 4; + + if (!foundDataChunk) { + pWav->dataChunkDataPos = cursor; + } + + /* + The sample count in the "fact" chunk is either unreliable, or I'm not understanding it properly. For now I am only enabling this + for Microsoft ADPCM formats. + */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + sampleCountFromFactChunk = sampleCount; + } else { + sampleCountFromFactChunk = 0; + } + } + } else { + if (drwav__guid_equal(header.id.guid, drwavGUID_W64_FACT)) { + if (drwav__on_read(onRead, pReadSeekUserData, &sampleCountFromFactChunk, 8, &cursor) != 8) { + return DRWAV_FALSE; + } + chunkSize -= 8; + + if (!foundDataChunk) { + pWav->dataChunkDataPos = cursor; + } + } + } + + /* "smpl" chunk. */ + if (pWav->container == drwav_container_riff) { + if (drwav__fourcc_equal(header.id.fourcc, "smpl")) { + unsigned char smplHeaderData[36]; /* 36 = size of the smpl header section, not including the loop data. */ + if (chunkSize >= sizeof(smplHeaderData)) { + drwav_uint64 bytesJustRead = drwav__on_read(onRead, pReadSeekUserData, smplHeaderData, sizeof(smplHeaderData), &cursor); + chunkSize -= bytesJustRead; + + if (bytesJustRead == sizeof(smplHeaderData)) { + drwav_uint32 iLoop; + + pWav->smpl.manufacturer = drwav__bytes_to_u32(smplHeaderData+0); + pWav->smpl.product = drwav__bytes_to_u32(smplHeaderData+4); + pWav->smpl.samplePeriod = drwav__bytes_to_u32(smplHeaderData+8); + pWav->smpl.midiUnityNotes = drwav__bytes_to_u32(smplHeaderData+12); + pWav->smpl.midiPitchFraction = drwav__bytes_to_u32(smplHeaderData+16); + pWav->smpl.smpteFormat = drwav__bytes_to_u32(smplHeaderData+20); + pWav->smpl.smpteOffset = drwav__bytes_to_u32(smplHeaderData+24); + pWav->smpl.numSampleLoops = drwav__bytes_to_u32(smplHeaderData+28); + pWav->smpl.samplerData = drwav__bytes_to_u32(smplHeaderData+32); + + for (iLoop = 0; iLoop < pWav->smpl.numSampleLoops && iLoop < drwav_countof(pWav->smpl.loops); ++iLoop) { + unsigned char smplLoopData[24]; /* 24 = size of a loop section in the smpl chunk. */ + bytesJustRead = drwav__on_read(onRead, pReadSeekUserData, smplLoopData, sizeof(smplLoopData), &cursor); + chunkSize -= bytesJustRead; + + if (bytesJustRead == sizeof(smplLoopData)) { + pWav->smpl.loops[iLoop].cuePointId = drwav__bytes_to_u32(smplLoopData+0); + pWav->smpl.loops[iLoop].type = drwav__bytes_to_u32(smplLoopData+4); + pWav->smpl.loops[iLoop].start = drwav__bytes_to_u32(smplLoopData+8); + pWav->smpl.loops[iLoop].end = drwav__bytes_to_u32(smplLoopData+12); + pWav->smpl.loops[iLoop].fraction = drwav__bytes_to_u32(smplLoopData+16); + pWav->smpl.loops[iLoop].playCount = drwav__bytes_to_u32(smplLoopData+20); + } else { + break; /* Break from the smpl loop for loop. */ + } + } + } + } else { + /* Looks like invalid data. Ignore the chunk. */ + } + } + } else { + if (drwav__guid_equal(header.id.guid, drwavGUID_W64_SMPL)) { + /* + This path will be hit when a W64 WAV file contains a smpl chunk. I don't have a sample file to test this path, so a contribution + is welcome to add support for this. + */ + } + } + + /* Make sure we seek past the padding. */ + chunkSize += header.paddingSize; + if (!drwav__seek_forward(onSeek, chunkSize, pReadSeekUserData)) { + break; + } + cursor += chunkSize; + + if (!foundDataChunk) { + pWav->dataChunkDataPos = cursor; + } + } + + /* If we haven't found a data chunk, return an error. */ + if (!foundDataChunk) { + return DRWAV_FALSE; + } + + /* We may have moved passed the data chunk. If so we need to move back. If running in sequential mode we can assume we are already sitting on the data chunk. */ + if (!sequential) { + if (!drwav__seek_from_start(onSeek, pWav->dataChunkDataPos, pReadSeekUserData)) { + return DRWAV_FALSE; + } + cursor = pWav->dataChunkDataPos; + } + + + /* At this point we should be sitting on the first byte of the raw audio data. */ + + pWav->fmt = fmt; + pWav->sampleRate = fmt.sampleRate; + pWav->channels = fmt.channels; + pWav->bitsPerSample = fmt.bitsPerSample; + pWav->bytesRemaining = dataChunkSize; + pWav->translatedFormatTag = translatedFormatTag; + pWav->dataChunkDataSize = dataChunkSize; + + if (sampleCountFromFactChunk != 0) { + pWav->totalPCMFrameCount = sampleCountFromFactChunk; + } else { + pWav->totalPCMFrameCount = dataChunkSize / drwav_get_bytes_per_pcm_frame(pWav); + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign; + pWav->totalPCMFrameCount = (((blockCount * (fmt.blockAlign - (6*pWav->channels))) * 2)) / fmt.channels; /* x2 because two samples per byte. */ + } + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign; + pWav->totalPCMFrameCount = (((blockCount * (fmt.blockAlign - (4*pWav->channels))) * 2) + (blockCount * pWav->channels)) / fmt.channels; + } + } + + /* Some formats only support a certain number of channels. */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM || pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + if (pWav->channels > 2) { + return DRWAV_FALSE; + } + } + +#ifdef DR_WAV_LIBSNDFILE_COMPAT + /* + I use libsndfile as a benchmark for testing, however in the version I'm using (from the Windows installer on the libsndfile website), + it appears the total sample count libsndfile uses for MS-ADPCM is incorrect. It would seem they are computing the total sample count + from the number of blocks, however this results in the inclusion of extra silent samples at the end of the last block. The correct + way to know the total sample count is to inspect the "fact" chunk, which should always be present for compressed formats, and should + always include the sample count. This little block of code below is only used to emulate the libsndfile logic so I can properly run my + correctness tests against libsndfile, and is disabled by default. + */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign; + pWav->totalPCMFrameCount = (((blockCount * (fmt.blockAlign - (6*pWav->channels))) * 2)) / fmt.channels; /* x2 because two samples per byte. */ + } + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign; + pWav->totalPCMFrameCount = (((blockCount * (fmt.blockAlign - (4*pWav->channels))) * 2) + (blockCount * pWav->channels)) / fmt.channels; + } +#endif + + pWav->totalSampleCount = pWav->totalPCMFrameCount * pWav->channels; + + return DRWAV_TRUE; +} + + +drwav_uint32 drwav_riff_chunk_size_riff(drwav_uint64 dataChunkSize) +{ + if (dataChunkSize <= (0xFFFFFFFFUL - 36)) { + return 36 + (drwav_uint32)dataChunkSize; + } else { + return 0xFFFFFFFF; + } +} + +drwav_uint32 drwav_data_chunk_size_riff(drwav_uint64 dataChunkSize) +{ + if (dataChunkSize <= 0xFFFFFFFFUL) { + return (drwav_uint32)dataChunkSize; + } else { + return 0xFFFFFFFFUL; + } +} + +drwav_uint64 drwav_riff_chunk_size_w64(drwav_uint64 dataChunkSize) +{ + return 80 + 24 + dataChunkSize; /* +24 because W64 includes the size of the GUID and size fields. */ +} + +drwav_uint64 drwav_data_chunk_size_w64(drwav_uint64 dataChunkSize) +{ + return 24 + dataChunkSize; /* +24 because W64 includes the size of the GUID and size fields. */ +} + + +drwav_bool32 drwav_init_write__internal(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData) +{ + size_t runningPos = 0; + drwav_uint64 initialDataChunkSize = 0; + drwav_uint64 chunkSizeFMT; + + if (pWav == NULL) { + return DRWAV_FALSE; + } + + if (onWrite == NULL) { + return DRWAV_FALSE; + } + + if (!isSequential && onSeek == NULL) { + return DRWAV_FALSE; /* <-- onSeek is required when in non-sequential mode. */ + } + + + /* Not currently supporting compressed formats. Will need to add support for the "fact" chunk before we enable this. */ + if (pFormat->format == DR_WAVE_FORMAT_EXTENSIBLE) { + return DRWAV_FALSE; + } + if (pFormat->format == DR_WAVE_FORMAT_ADPCM || pFormat->format == DR_WAVE_FORMAT_DVI_ADPCM) { + return DRWAV_FALSE; + } + + + drwav_zero_memory(pWav, sizeof(*pWav)); + pWav->onWrite = onWrite; + pWav->onSeek = onSeek; + pWav->pUserData = pUserData; + pWav->fmt.formatTag = (drwav_uint16)pFormat->format; + pWav->fmt.channels = (drwav_uint16)pFormat->channels; + pWav->fmt.sampleRate = pFormat->sampleRate; + pWav->fmt.avgBytesPerSec = (drwav_uint32)((pFormat->bitsPerSample * pFormat->sampleRate * pFormat->channels) / 8); + pWav->fmt.blockAlign = (drwav_uint16)((pFormat->channels * pFormat->bitsPerSample) / 8); + pWav->fmt.bitsPerSample = (drwav_uint16)pFormat->bitsPerSample; + pWav->fmt.extendedSize = 0; + pWav->isSequentialWrite = isSequential; + + /* + The initial values for the "RIFF" and "data" chunks depends on whether or not we are initializing in sequential mode or not. In + sequential mode we set this to its final values straight away since they can be calculated from the total sample count. In non- + sequential mode we initialize it all to zero and fill it out in drwav_uninit() using a backwards seek. + */ + if (isSequential) { + initialDataChunkSize = (totalSampleCount * pWav->fmt.bitsPerSample) / 8; + + /* + The RIFF container has a limit on the number of samples. drwav is not allowing this. There's no practical limits for Wave64 + so for the sake of simplicity I'm not doing any validation for that. + */ + if (pFormat->container == drwav_container_riff) { + if (initialDataChunkSize > (0xFFFFFFFFUL - 36)) { + return DRWAV_FALSE; /* Not enough room to store every sample. */ + } + } + } + + pWav->dataChunkDataSizeTargetWrite = initialDataChunkSize; + + + /* "RIFF" chunk. */ + if (pFormat->container == drwav_container_riff) { + drwav_uint32 chunkSizeRIFF = 36 + (drwav_uint32)initialDataChunkSize; /* +36 = "RIFF"+[RIFF Chunk Size]+"WAVE" + [sizeof "fmt " chunk] */ + runningPos += pWav->onWrite(pUserData, "RIFF", 4); + runningPos += pWav->onWrite(pUserData, &chunkSizeRIFF, 4); + runningPos += pWav->onWrite(pUserData, "WAVE", 4); + } else { + drwav_uint64 chunkSizeRIFF = 80 + 24 + initialDataChunkSize; /* +24 because W64 includes the size of the GUID and size fields. */ + runningPos += pWav->onWrite(pUserData, drwavGUID_W64_RIFF, 16); + runningPos += pWav->onWrite(pUserData, &chunkSizeRIFF, 8); + runningPos += pWav->onWrite(pUserData, drwavGUID_W64_WAVE, 16); + } + + /* "fmt " chunk. */ + if (pFormat->container == drwav_container_riff) { + chunkSizeFMT = 16; + runningPos += pWav->onWrite(pUserData, "fmt ", 4); + runningPos += pWav->onWrite(pUserData, &chunkSizeFMT, 4); + } else { + chunkSizeFMT = 40; + runningPos += pWav->onWrite(pUserData, drwavGUID_W64_FMT, 16); + runningPos += pWav->onWrite(pUserData, &chunkSizeFMT, 8); + } + + runningPos += pWav->onWrite(pUserData, &pWav->fmt.formatTag, 2); + runningPos += pWav->onWrite(pUserData, &pWav->fmt.channels, 2); + runningPos += pWav->onWrite(pUserData, &pWav->fmt.sampleRate, 4); + runningPos += pWav->onWrite(pUserData, &pWav->fmt.avgBytesPerSec, 4); + runningPos += pWav->onWrite(pUserData, &pWav->fmt.blockAlign, 2); + runningPos += pWav->onWrite(pUserData, &pWav->fmt.bitsPerSample, 2); + + pWav->dataChunkDataPos = runningPos; + + /* "data" chunk. */ + if (pFormat->container == drwav_container_riff) { + drwav_uint32 chunkSizeDATA = (drwav_uint32)initialDataChunkSize; + runningPos += pWav->onWrite(pUserData, "data", 4); + runningPos += pWav->onWrite(pUserData, &chunkSizeDATA, 4); + } else { + drwav_uint64 chunkSizeDATA = 24 + initialDataChunkSize; /* +24 because W64 includes the size of the GUID and size fields. */ + runningPos += pWav->onWrite(pUserData, drwavGUID_W64_DATA, 16); + runningPos += pWav->onWrite(pUserData, &chunkSizeDATA, 8); + } + + + /* Simple validation. */ + if (pFormat->container == drwav_container_riff) { + if (runningPos != 20 + chunkSizeFMT + 8) { + return DRWAV_FALSE; + } + } else { + if (runningPos != 40 + chunkSizeFMT + 24) { + return DRWAV_FALSE; + } + } + + + + /* Set some properties for the client's convenience. */ + pWav->container = pFormat->container; + pWav->channels = (drwav_uint16)pFormat->channels; + pWav->sampleRate = pFormat->sampleRate; + pWav->bitsPerSample = (drwav_uint16)pFormat->bitsPerSample; + pWav->translatedFormatTag = (drwav_uint16)pFormat->format; + + return DRWAV_TRUE; +} + + +drwav_bool32 drwav_init_write(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData) +{ + return drwav_init_write__internal(pWav, pFormat, 0, DRWAV_FALSE, onWrite, onSeek, pUserData); /* DRWAV_FALSE = Not Sequential */ +} + +drwav_bool32 drwav_init_write_sequential(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData) +{ + return drwav_init_write__internal(pWav, pFormat, totalSampleCount, DRWAV_TRUE, onWrite, NULL, pUserData); /* DRWAV_TRUE = Sequential */ +} + +void drwav_uninit(drwav* pWav) +{ + if (pWav == NULL) { + return; + } + + /* + If the drwav object was opened in write mode we'll need to finalize a few things: + - Make sure the "data" chunk is aligned to 16-bits for RIFF containers, or 64 bits for W64 containers. + - Set the size of the "data" chunk. + */ + if (pWav->onWrite != NULL) { + drwav_uint32 paddingSize = 0; + + /* Validation for sequential mode. */ + if (pWav->isSequentialWrite) { + drwav_assert(pWav->dataChunkDataSize == pWav->dataChunkDataSizeTargetWrite); + } + + /* Padding. Do not adjust pWav->dataChunkDataSize - this should not include the padding. */ + if (pWav->container == drwav_container_riff) { + paddingSize = (drwav_uint32)(pWav->dataChunkDataSize % 2); + } else { + paddingSize = (drwav_uint32)(pWav->dataChunkDataSize % 8); + } + + if (paddingSize > 0) { + drwav_uint64 paddingData = 0; + pWav->onWrite(pWav->pUserData, &paddingData, paddingSize); + } + + /* + Chunk sizes. When using sequential mode, these will have been filled in at initialization time. We only need + to do this when using non-sequential mode. + */ + if (pWav->onSeek && !pWav->isSequentialWrite) { + if (pWav->container == drwav_container_riff) { + /* The "RIFF" chunk size. */ + if (pWav->onSeek(pWav->pUserData, 4, drwav_seek_origin_start)) { + drwav_uint32 riffChunkSize = drwav_riff_chunk_size_riff(pWav->dataChunkDataSize); + pWav->onWrite(pWav->pUserData, &riffChunkSize, 4); + } + + /* the "data" chunk size. */ + if (pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos + 4, drwav_seek_origin_start)) { + drwav_uint32 dataChunkSize = drwav_data_chunk_size_riff(pWav->dataChunkDataSize); + pWav->onWrite(pWav->pUserData, &dataChunkSize, 4); + } + } else { + /* The "RIFF" chunk size. */ + if (pWav->onSeek(pWav->pUserData, 16, drwav_seek_origin_start)) { + drwav_uint64 riffChunkSize = drwav_riff_chunk_size_w64(pWav->dataChunkDataSize); + pWav->onWrite(pWav->pUserData, &riffChunkSize, 8); + } + + /* The "data" chunk size. */ + if (pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos + 16, drwav_seek_origin_start)) { + drwav_uint64 dataChunkSize = drwav_data_chunk_size_w64(pWav->dataChunkDataSize); + pWav->onWrite(pWav->pUserData, &dataChunkSize, 8); + } + } + } + } + +#ifndef DR_WAV_NO_STDIO + /* + If we opened the file with drwav_open_file() we will want to close the file handle. We can know whether or not drwav_open_file() + was used by looking at the onRead and onSeek callbacks. + */ + if (pWav->onRead == drwav__on_read_stdio || pWav->onWrite == drwav__on_write_stdio) { + fclose((FILE*)pWav->pUserData); + } +#endif +} + + +drwav* drwav_open(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData) +{ + return drwav_open_ex(onRead, onSeek, NULL, pUserData, NULL, 0); +} + +drwav* drwav_open_ex(drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_chunk_proc onChunk, void* pReadSeekUserData, void* pChunkUserData, drwav_uint32 flags) +{ + drwav* pWav = (drwav*)DRWAV_MALLOC(sizeof(*pWav)); + if (pWav == NULL) { + return NULL; + } + + if (!drwav_init_ex(pWav, onRead, onSeek, onChunk, pReadSeekUserData, pChunkUserData, flags)) { + DRWAV_FREE(pWav); + return NULL; + } + + return pWav; +} + + +drwav* drwav_open_write__internal(const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData) +{ + drwav* pWav = (drwav*)DRWAV_MALLOC(sizeof(*pWav)); + if (pWav == NULL) { + return NULL; + } + + if (!drwav_init_write__internal(pWav, pFormat, totalSampleCount, isSequential, onWrite, onSeek, pUserData)) { + DRWAV_FREE(pWav); + return NULL; + } + + return pWav; +} + +drwav* drwav_open_write(const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData) +{ + return drwav_open_write__internal(pFormat, 0, DRWAV_FALSE, onWrite, onSeek, pUserData); +} + +drwav* drwav_open_write_sequential(const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData) +{ + return drwav_open_write__internal(pFormat, totalSampleCount, DRWAV_TRUE, onWrite, NULL, pUserData); +} + +void drwav_close(drwav* pWav) +{ + drwav_uninit(pWav); + DRWAV_FREE(pWav); +} + + +size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOut) +{ + size_t bytesRead; + + if (pWav == NULL || bytesToRead == 0 || pBufferOut == NULL) { + return 0; + } + + if (bytesToRead > pWav->bytesRemaining) { + bytesToRead = (size_t)pWav->bytesRemaining; + } + + bytesRead = pWav->onRead(pWav->pUserData, pBufferOut, bytesToRead); + + pWav->bytesRemaining -= bytesRead; + return bytesRead; +} + +drwav_uint64 drwav_read(drwav* pWav, drwav_uint64 samplesToRead, void* pBufferOut) +{ + drwav_uint32 bytesPerSample; + size_t bytesRead; + + if (pWav == NULL || samplesToRead == 0 || pBufferOut == NULL) { + return 0; + } + + /* Cannot use this function for compressed formats. */ + if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) { + return 0; + } + + bytesPerSample = drwav_get_bytes_per_sample(pWav); + if (bytesPerSample == 0) { + return 0; + } + + /* Don't try to read more samples than can potentially fit in the output buffer. */ + if (samplesToRead * bytesPerSample > DRWAV_SIZE_MAX) { + samplesToRead = DRWAV_SIZE_MAX / bytesPerSample; + } + + bytesRead = drwav_read_raw(pWav, (size_t)(samplesToRead * bytesPerSample), pBufferOut); + return bytesRead / bytesPerSample; +} + +drwav_uint64 drwav_read_pcm_frames(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut) +{ + drwav_uint32 bytesPerFrame; + size_t bytesRead; + + if (pWav == NULL || framesToRead == 0 || pBufferOut == NULL) { + return 0; + } + + /* Cannot use this function for compressed formats. */ + if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) { + return 0; + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + /* Don't try to read more samples than can potentially fit in the output buffer. */ + if (framesToRead * bytesPerFrame > DRWAV_SIZE_MAX) { + framesToRead = DRWAV_SIZE_MAX / bytesPerFrame; + } + + bytesRead = drwav_read_raw(pWav, (size_t)(framesToRead * bytesPerFrame), pBufferOut); + return bytesRead / bytesPerFrame; +} + +drwav_bool32 drwav_seek_to_first_pcm_frame(drwav* pWav) +{ + if (pWav->onWrite != NULL) { + return DRWAV_FALSE; /* No seeking in write mode. */ + } + + if (!pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos, drwav_seek_origin_start)) { + return DRWAV_FALSE; + } + + if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) { + pWav->compressed.iCurrentSample = 0; + } + + pWav->bytesRemaining = pWav->dataChunkDataSize; + return DRWAV_TRUE; +} + +drwav_bool32 drwav_seek_to_sample(drwav* pWav, drwav_uint64 sample) +{ + /* Seeking should be compatible with wave files > 2GB. */ + + if (pWav->onWrite != NULL) { + return DRWAV_FALSE; /* No seeking in write mode. */ + } + + if (pWav == NULL || pWav->onSeek == NULL) { + return DRWAV_FALSE; + } + + /* If there are no samples, just return DRWAV_TRUE without doing anything. */ + if (pWav->totalSampleCount == 0) { + return DRWAV_TRUE; + } + + /* Make sure the sample is clamped. */ + if (sample >= pWav->totalSampleCount) { + sample = pWav->totalSampleCount - 1; + } + + /* + For compressed formats we just use a slow generic seek. If we are seeking forward we just seek forward. If we are going backwards we need + to seek back to the start. + */ + if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) { + /* TODO: This can be optimized. */ + + /* + If we're seeking forward it's simple - just keep reading samples until we hit the sample we're requesting. If we're seeking backwards, + we first need to seek back to the start and then just do the same thing as a forward seek. + */ + if (sample < pWav->compressed.iCurrentSample) { + if (!drwav_seek_to_first_pcm_frame(pWav)) { + return DRWAV_FALSE; + } + } + + if (sample > pWav->compressed.iCurrentSample) { + drwav_uint64 offset = sample - pWav->compressed.iCurrentSample; + + drwav_int16 devnull[2048]; + while (offset > 0) { + drwav_uint64 samplesRead = 0; + drwav_uint64 samplesToRead = offset; + if (samplesToRead > 2048) { + samplesToRead = 2048; + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + samplesRead = drwav_read_s16__msadpcm(pWav, samplesToRead, devnull); + } else if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + samplesRead = drwav_read_s16__ima(pWav, samplesToRead, devnull); + } else { + assert(DRWAV_FALSE); /* If this assertion is triggered it means I've implemented a new compressed format but forgot to add a branch for it here. */ + } + + if (samplesRead != samplesToRead) { + return DRWAV_FALSE; + } + + offset -= samplesRead; + } + } + } else { + drwav_uint64 totalSizeInBytes; + drwav_uint64 currentBytePos; + drwav_uint64 targetBytePos; + drwav_uint64 offset; + + totalSizeInBytes = pWav->totalPCMFrameCount * drwav_get_bytes_per_pcm_frame(pWav); + drwav_assert(totalSizeInBytes >= pWav->bytesRemaining); + + currentBytePos = totalSizeInBytes - pWav->bytesRemaining; + targetBytePos = sample * drwav_get_bytes_per_sample(pWav); + + if (currentBytePos < targetBytePos) { + /* Offset forwards. */ + offset = (targetBytePos - currentBytePos); + } else { + /* Offset backwards. */ + if (!drwav_seek_to_first_pcm_frame(pWav)) { + return DRWAV_FALSE; + } + offset = targetBytePos; + } + + while (offset > 0) { + int offset32 = ((offset > INT_MAX) ? INT_MAX : (int)offset); + if (!pWav->onSeek(pWav->pUserData, offset32, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + + pWav->bytesRemaining -= offset32; + offset -= offset32; + } + } + + return DRWAV_TRUE; +} + +drwav_bool32 drwav_seek_to_pcm_frame(drwav* pWav, drwav_uint64 targetFrameIndex) +{ + return drwav_seek_to_sample(pWav, targetFrameIndex * pWav->channels); +} + + +size_t drwav_write_raw(drwav* pWav, size_t bytesToWrite, const void* pData) +{ + size_t bytesWritten; + + if (pWav == NULL || bytesToWrite == 0 || pData == NULL) { + return 0; + } + + bytesWritten = pWav->onWrite(pWav->pUserData, pData, bytesToWrite); + pWav->dataChunkDataSize += bytesWritten; + + return bytesWritten; +} + +drwav_uint64 drwav_write(drwav* pWav, drwav_uint64 samplesToWrite, const void* pData) +{ + drwav_uint64 bytesToWrite; + drwav_uint64 bytesWritten; + const drwav_uint8* pRunningData; + + if (pWav == NULL || samplesToWrite == 0 || pData == NULL) { + return 0; + } + + bytesToWrite = ((samplesToWrite * pWav->bitsPerSample) / 8); + if (bytesToWrite > DRWAV_SIZE_MAX) { + return 0; + } + + bytesWritten = 0; + pRunningData = (const drwav_uint8*)pData; + while (bytesToWrite > 0) { + size_t bytesJustWritten; + drwav_uint64 bytesToWriteThisIteration = bytesToWrite; + if (bytesToWriteThisIteration > DRWAV_SIZE_MAX) { + bytesToWriteThisIteration = DRWAV_SIZE_MAX; + } + + bytesJustWritten = drwav_write_raw(pWav, (size_t)bytesToWriteThisIteration, pRunningData); + if (bytesJustWritten == 0) { + break; + } + + bytesToWrite -= bytesJustWritten; + bytesWritten += bytesJustWritten; + pRunningData += bytesJustWritten; + } + + return (bytesWritten * 8) / pWav->bitsPerSample; +} + +drwav_uint64 drwav_write_pcm_frames(drwav* pWav, drwav_uint64 framesToWrite, const void* pData) +{ + return drwav_write(pWav, framesToWrite * pWav->channels, pData) / pWav->channels; +} + + + +drwav_uint64 drwav_read_s16__msadpcm(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalSamplesRead = 0; + + drwav_assert(pWav != NULL); + drwav_assert(samplesToRead > 0); + drwav_assert(pBufferOut != NULL); + + /* TODO: Lots of room for optimization here. */ + + while (samplesToRead > 0 && pWav->compressed.iCurrentSample < pWav->totalSampleCount) { + /* If there are no cached samples we need to load a new block. */ + if (pWav->msadpcm.cachedSampleCount == 0 && pWav->msadpcm.bytesRemainingInBlock == 0) { + if (pWav->channels == 1) { + /* Mono. */ + drwav_uint8 header[7]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalSamplesRead; + } + pWav->msadpcm.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + pWav->msadpcm.predictor[0] = header[0]; + pWav->msadpcm.delta[0] = drwav__bytes_to_s16(header + 1); + pWav->msadpcm.prevSamples[0][1] = (drwav_int32)drwav__bytes_to_s16(header + 3); + pWav->msadpcm.prevSamples[0][0] = (drwav_int32)drwav__bytes_to_s16(header + 5); + pWav->msadpcm.cachedSamples[2] = pWav->msadpcm.prevSamples[0][0]; + pWav->msadpcm.cachedSamples[3] = pWav->msadpcm.prevSamples[0][1]; + pWav->msadpcm.cachedSampleCount = 2; + } else { + /* Stereo. */ + drwav_uint8 header[14]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalSamplesRead; + } + pWav->msadpcm.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + pWav->msadpcm.predictor[0] = header[0]; + pWav->msadpcm.predictor[1] = header[1]; + pWav->msadpcm.delta[0] = drwav__bytes_to_s16(header + 2); + pWav->msadpcm.delta[1] = drwav__bytes_to_s16(header + 4); + pWav->msadpcm.prevSamples[0][1] = (drwav_int32)drwav__bytes_to_s16(header + 6); + pWav->msadpcm.prevSamples[1][1] = (drwav_int32)drwav__bytes_to_s16(header + 8); + pWav->msadpcm.prevSamples[0][0] = (drwav_int32)drwav__bytes_to_s16(header + 10); + pWav->msadpcm.prevSamples[1][0] = (drwav_int32)drwav__bytes_to_s16(header + 12); + + pWav->msadpcm.cachedSamples[0] = pWav->msadpcm.prevSamples[0][0]; + pWav->msadpcm.cachedSamples[1] = pWav->msadpcm.prevSamples[1][0]; + pWav->msadpcm.cachedSamples[2] = pWav->msadpcm.prevSamples[0][1]; + pWav->msadpcm.cachedSamples[3] = pWav->msadpcm.prevSamples[1][1]; + pWav->msadpcm.cachedSampleCount = 4; + } + } + + /* Output anything that's cached. */ + while (samplesToRead > 0 && pWav->msadpcm.cachedSampleCount > 0 && pWav->compressed.iCurrentSample < pWav->totalSampleCount) { + pBufferOut[0] = (drwav_int16)pWav->msadpcm.cachedSamples[drwav_countof(pWav->msadpcm.cachedSamples) - pWav->msadpcm.cachedSampleCount]; + pWav->msadpcm.cachedSampleCount -= 1; + + pBufferOut += 1; + samplesToRead -= 1; + totalSamplesRead += 1; + pWav->compressed.iCurrentSample += 1; + } + + if (samplesToRead == 0) { + return totalSamplesRead; + } + + + /* + If there's nothing left in the cache, just go ahead and load more. If there's nothing left to load in the current block we just continue to the next + loop iteration which will trigger the loading of a new block. + */ + if (pWav->msadpcm.cachedSampleCount == 0) { + if (pWav->msadpcm.bytesRemainingInBlock == 0) { + continue; + } else { + static drwav_int32 adaptationTable[] = { + 230, 230, 230, 230, 307, 409, 512, 614, + 768, 614, 512, 409, 307, 230, 230, 230 + }; + static drwav_int32 coeff1Table[] = { 256, 512, 0, 192, 240, 460, 392 }; + static drwav_int32 coeff2Table[] = { 0, -256, 0, 64, 0, -208, -232 }; + + drwav_uint8 nibbles; + drwav_int32 nibble0; + drwav_int32 nibble1; + + if (pWav->onRead(pWav->pUserData, &nibbles, 1) != 1) { + return totalSamplesRead; + } + pWav->msadpcm.bytesRemainingInBlock -= 1; + + /* TODO: Optimize away these if statements. */ + nibble0 = ((nibbles & 0xF0) >> 4); if ((nibbles & 0x80)) { nibble0 |= 0xFFFFFFF0UL; } + nibble1 = ((nibbles & 0x0F) >> 0); if ((nibbles & 0x08)) { nibble1 |= 0xFFFFFFF0UL; } + + if (pWav->channels == 1) { + /* Mono. */ + drwav_int32 newSample0; + drwav_int32 newSample1; + + newSample0 = ((pWav->msadpcm.prevSamples[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevSamples[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8; + newSample0 += nibble0 * pWav->msadpcm.delta[0]; + newSample0 = drwav_clamp(newSample0, -32768, 32767); + + pWav->msadpcm.delta[0] = (adaptationTable[((nibbles & 0xF0) >> 4)] * pWav->msadpcm.delta[0]) >> 8; + if (pWav->msadpcm.delta[0] < 16) { + pWav->msadpcm.delta[0] = 16; + } + + pWav->msadpcm.prevSamples[0][0] = pWav->msadpcm.prevSamples[0][1]; + pWav->msadpcm.prevSamples[0][1] = newSample0; + + + newSample1 = ((pWav->msadpcm.prevSamples[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevSamples[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8; + newSample1 += nibble1 * pWav->msadpcm.delta[0]; + newSample1 = drwav_clamp(newSample1, -32768, 32767); + + pWav->msadpcm.delta[0] = (adaptationTable[((nibbles & 0x0F) >> 0)] * pWav->msadpcm.delta[0]) >> 8; + if (pWav->msadpcm.delta[0] < 16) { + pWav->msadpcm.delta[0] = 16; + } + + pWav->msadpcm.prevSamples[0][0] = pWav->msadpcm.prevSamples[0][1]; + pWav->msadpcm.prevSamples[0][1] = newSample1; + + + pWav->msadpcm.cachedSamples[2] = newSample0; + pWav->msadpcm.cachedSamples[3] = newSample1; + pWav->msadpcm.cachedSampleCount = 2; + } else { + /* Stereo. */ + drwav_int32 newSample0; + drwav_int32 newSample1; + + /* Left. */ + newSample0 = ((pWav->msadpcm.prevSamples[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevSamples[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8; + newSample0 += nibble0 * pWav->msadpcm.delta[0]; + newSample0 = drwav_clamp(newSample0, -32768, 32767); + + pWav->msadpcm.delta[0] = (adaptationTable[((nibbles & 0xF0) >> 4)] * pWav->msadpcm.delta[0]) >> 8; + if (pWav->msadpcm.delta[0] < 16) { + pWav->msadpcm.delta[0] = 16; + } + + pWav->msadpcm.prevSamples[0][0] = pWav->msadpcm.prevSamples[0][1]; + pWav->msadpcm.prevSamples[0][1] = newSample0; + + + /* Right. */ + newSample1 = ((pWav->msadpcm.prevSamples[1][1] * coeff1Table[pWav->msadpcm.predictor[1]]) + (pWav->msadpcm.prevSamples[1][0] * coeff2Table[pWav->msadpcm.predictor[1]])) >> 8; + newSample1 += nibble1 * pWav->msadpcm.delta[1]; + newSample1 = drwav_clamp(newSample1, -32768, 32767); + + pWav->msadpcm.delta[1] = (adaptationTable[((nibbles & 0x0F) >> 0)] * pWav->msadpcm.delta[1]) >> 8; + if (pWav->msadpcm.delta[1] < 16) { + pWav->msadpcm.delta[1] = 16; + } + + pWav->msadpcm.prevSamples[1][0] = pWav->msadpcm.prevSamples[1][1]; + pWav->msadpcm.prevSamples[1][1] = newSample1; + + pWav->msadpcm.cachedSamples[2] = newSample0; + pWav->msadpcm.cachedSamples[3] = newSample1; + pWav->msadpcm.cachedSampleCount = 2; + } + } + } + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s16__ima(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalSamplesRead = 0; + + drwav_assert(pWav != NULL); + drwav_assert(samplesToRead > 0); + drwav_assert(pBufferOut != NULL); + + /* TODO: Lots of room for optimization here. */ + + while (samplesToRead > 0 && pWav->compressed.iCurrentSample < pWav->totalSampleCount) { + /* If there are no cached samples we need to load a new block. */ + if (pWav->ima.cachedSampleCount == 0 && pWav->ima.bytesRemainingInBlock == 0) { + if (pWav->channels == 1) { + /* Mono. */ + drwav_uint8 header[4]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalSamplesRead; + } + pWav->ima.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + pWav->ima.predictor[0] = drwav__bytes_to_s16(header + 0); + pWav->ima.stepIndex[0] = header[2]; + pWav->ima.cachedSamples[drwav_countof(pWav->ima.cachedSamples) - 1] = pWav->ima.predictor[0]; + pWav->ima.cachedSampleCount = 1; + } else { + /* Stereo. */ + drwav_uint8 header[8]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalSamplesRead; + } + pWav->ima.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + pWav->ima.predictor[0] = drwav__bytes_to_s16(header + 0); + pWav->ima.stepIndex[0] = header[2]; + pWav->ima.predictor[1] = drwav__bytes_to_s16(header + 4); + pWav->ima.stepIndex[1] = header[6]; + + pWav->ima.cachedSamples[drwav_countof(pWav->ima.cachedSamples) - 2] = pWav->ima.predictor[0]; + pWav->ima.cachedSamples[drwav_countof(pWav->ima.cachedSamples) - 1] = pWav->ima.predictor[1]; + pWav->ima.cachedSampleCount = 2; + } + } + + /* Output anything that's cached. */ + while (samplesToRead > 0 && pWav->ima.cachedSampleCount > 0 && pWav->compressed.iCurrentSample < pWav->totalSampleCount) { + pBufferOut[0] = (drwav_int16)pWav->ima.cachedSamples[drwav_countof(pWav->ima.cachedSamples) - pWav->ima.cachedSampleCount]; + pWav->ima.cachedSampleCount -= 1; + + pBufferOut += 1; + samplesToRead -= 1; + totalSamplesRead += 1; + pWav->compressed.iCurrentSample += 1; + } + + if (samplesToRead == 0) { + return totalSamplesRead; + } + + /* + If there's nothing left in the cache, just go ahead and load more. If there's nothing left to load in the current block we just continue to the next + loop iteration which will trigger the loading of a new block. + */ + if (pWav->ima.cachedSampleCount == 0) { + if (pWav->ima.bytesRemainingInBlock == 0) { + continue; + } else { + static drwav_int32 indexTable[16] = { + -1, -1, -1, -1, 2, 4, 6, 8, + -1, -1, -1, -1, 2, 4, 6, 8 + }; + + static drwav_int32 stepTable[89] = { + 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, + 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, + 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, + 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, + 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, + 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, + 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, + 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, + 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767 + }; + + drwav_uint32 iChannel; + + /* + From what I can tell with stereo streams, it looks like every 4 bytes (8 samples) is for one channel. So it goes 4 bytes for the + left channel, 4 bytes for the right channel. + */ + pWav->ima.cachedSampleCount = 8 * pWav->channels; + for (iChannel = 0; iChannel < pWav->channels; ++iChannel) { + drwav_uint32 iByte; + drwav_uint8 nibbles[4]; + if (pWav->onRead(pWav->pUserData, &nibbles, 4) != 4) { + return totalSamplesRead; + } + pWav->ima.bytesRemainingInBlock -= 4; + + for (iByte = 0; iByte < 4; ++iByte) { + drwav_uint8 nibble0 = ((nibbles[iByte] & 0x0F) >> 0); + drwav_uint8 nibble1 = ((nibbles[iByte] & 0xF0) >> 4); + + drwav_int32 step = stepTable[pWav->ima.stepIndex[iChannel]]; + drwav_int32 predictor = pWav->ima.predictor[iChannel]; + + drwav_int32 diff = step >> 3; + if (nibble0 & 1) diff += step >> 2; + if (nibble0 & 2) diff += step >> 1; + if (nibble0 & 4) diff += step; + if (nibble0 & 8) diff = -diff; + + predictor = drwav_clamp(predictor + diff, -32768, 32767); + pWav->ima.predictor[iChannel] = predictor; + pWav->ima.stepIndex[iChannel] = drwav_clamp(pWav->ima.stepIndex[iChannel] + indexTable[nibble0], 0, (drwav_int32)drwav_countof(stepTable)-1); + pWav->ima.cachedSamples[(drwav_countof(pWav->ima.cachedSamples) - pWav->ima.cachedSampleCount) + (iByte*2+0)*pWav->channels + iChannel] = predictor; + + + step = stepTable[pWav->ima.stepIndex[iChannel]]; + predictor = pWav->ima.predictor[iChannel]; + + diff = step >> 3; + if (nibble1 & 1) diff += step >> 2; + if (nibble1 & 2) diff += step >> 1; + if (nibble1 & 4) diff += step; + if (nibble1 & 8) diff = -diff; + + predictor = drwav_clamp(predictor + diff, -32768, 32767); + pWav->ima.predictor[iChannel] = predictor; + pWav->ima.stepIndex[iChannel] = drwav_clamp(pWav->ima.stepIndex[iChannel] + indexTable[nibble1], 0, (drwav_int32)drwav_countof(stepTable)-1); + pWav->ima.cachedSamples[(drwav_countof(pWav->ima.cachedSamples) - pWav->ima.cachedSampleCount) + (iByte*2+1)*pWav->channels + iChannel] = predictor; + } + } + } + } + } + + return totalSamplesRead; +} + + +#ifndef DR_WAV_NO_CONVERSION_API +static unsigned short g_drwavAlawTable[256] = { + 0xEA80, 0xEB80, 0xE880, 0xE980, 0xEE80, 0xEF80, 0xEC80, 0xED80, 0xE280, 0xE380, 0xE080, 0xE180, 0xE680, 0xE780, 0xE480, 0xE580, + 0xF540, 0xF5C0, 0xF440, 0xF4C0, 0xF740, 0xF7C0, 0xF640, 0xF6C0, 0xF140, 0xF1C0, 0xF040, 0xF0C0, 0xF340, 0xF3C0, 0xF240, 0xF2C0, + 0xAA00, 0xAE00, 0xA200, 0xA600, 0xBA00, 0xBE00, 0xB200, 0xB600, 0x8A00, 0x8E00, 0x8200, 0x8600, 0x9A00, 0x9E00, 0x9200, 0x9600, + 0xD500, 0xD700, 0xD100, 0xD300, 0xDD00, 0xDF00, 0xD900, 0xDB00, 0xC500, 0xC700, 0xC100, 0xC300, 0xCD00, 0xCF00, 0xC900, 0xCB00, + 0xFEA8, 0xFEB8, 0xFE88, 0xFE98, 0xFEE8, 0xFEF8, 0xFEC8, 0xFED8, 0xFE28, 0xFE38, 0xFE08, 0xFE18, 0xFE68, 0xFE78, 0xFE48, 0xFE58, + 0xFFA8, 0xFFB8, 0xFF88, 0xFF98, 0xFFE8, 0xFFF8, 0xFFC8, 0xFFD8, 0xFF28, 0xFF38, 0xFF08, 0xFF18, 0xFF68, 0xFF78, 0xFF48, 0xFF58, + 0xFAA0, 0xFAE0, 0xFA20, 0xFA60, 0xFBA0, 0xFBE0, 0xFB20, 0xFB60, 0xF8A0, 0xF8E0, 0xF820, 0xF860, 0xF9A0, 0xF9E0, 0xF920, 0xF960, + 0xFD50, 0xFD70, 0xFD10, 0xFD30, 0xFDD0, 0xFDF0, 0xFD90, 0xFDB0, 0xFC50, 0xFC70, 0xFC10, 0xFC30, 0xFCD0, 0xFCF0, 0xFC90, 0xFCB0, + 0x1580, 0x1480, 0x1780, 0x1680, 0x1180, 0x1080, 0x1380, 0x1280, 0x1D80, 0x1C80, 0x1F80, 0x1E80, 0x1980, 0x1880, 0x1B80, 0x1A80, + 0x0AC0, 0x0A40, 0x0BC0, 0x0B40, 0x08C0, 0x0840, 0x09C0, 0x0940, 0x0EC0, 0x0E40, 0x0FC0, 0x0F40, 0x0CC0, 0x0C40, 0x0DC0, 0x0D40, + 0x5600, 0x5200, 0x5E00, 0x5A00, 0x4600, 0x4200, 0x4E00, 0x4A00, 0x7600, 0x7200, 0x7E00, 0x7A00, 0x6600, 0x6200, 0x6E00, 0x6A00, + 0x2B00, 0x2900, 0x2F00, 0x2D00, 0x2300, 0x2100, 0x2700, 0x2500, 0x3B00, 0x3900, 0x3F00, 0x3D00, 0x3300, 0x3100, 0x3700, 0x3500, + 0x0158, 0x0148, 0x0178, 0x0168, 0x0118, 0x0108, 0x0138, 0x0128, 0x01D8, 0x01C8, 0x01F8, 0x01E8, 0x0198, 0x0188, 0x01B8, 0x01A8, + 0x0058, 0x0048, 0x0078, 0x0068, 0x0018, 0x0008, 0x0038, 0x0028, 0x00D8, 0x00C8, 0x00F8, 0x00E8, 0x0098, 0x0088, 0x00B8, 0x00A8, + 0x0560, 0x0520, 0x05E0, 0x05A0, 0x0460, 0x0420, 0x04E0, 0x04A0, 0x0760, 0x0720, 0x07E0, 0x07A0, 0x0660, 0x0620, 0x06E0, 0x06A0, + 0x02B0, 0x0290, 0x02F0, 0x02D0, 0x0230, 0x0210, 0x0270, 0x0250, 0x03B0, 0x0390, 0x03F0, 0x03D0, 0x0330, 0x0310, 0x0370, 0x0350 +}; + +static unsigned short g_drwavMulawTable[256] = { + 0x8284, 0x8684, 0x8A84, 0x8E84, 0x9284, 0x9684, 0x9A84, 0x9E84, 0xA284, 0xA684, 0xAA84, 0xAE84, 0xB284, 0xB684, 0xBA84, 0xBE84, + 0xC184, 0xC384, 0xC584, 0xC784, 0xC984, 0xCB84, 0xCD84, 0xCF84, 0xD184, 0xD384, 0xD584, 0xD784, 0xD984, 0xDB84, 0xDD84, 0xDF84, + 0xE104, 0xE204, 0xE304, 0xE404, 0xE504, 0xE604, 0xE704, 0xE804, 0xE904, 0xEA04, 0xEB04, 0xEC04, 0xED04, 0xEE04, 0xEF04, 0xF004, + 0xF0C4, 0xF144, 0xF1C4, 0xF244, 0xF2C4, 0xF344, 0xF3C4, 0xF444, 0xF4C4, 0xF544, 0xF5C4, 0xF644, 0xF6C4, 0xF744, 0xF7C4, 0xF844, + 0xF8A4, 0xF8E4, 0xF924, 0xF964, 0xF9A4, 0xF9E4, 0xFA24, 0xFA64, 0xFAA4, 0xFAE4, 0xFB24, 0xFB64, 0xFBA4, 0xFBE4, 0xFC24, 0xFC64, + 0xFC94, 0xFCB4, 0xFCD4, 0xFCF4, 0xFD14, 0xFD34, 0xFD54, 0xFD74, 0xFD94, 0xFDB4, 0xFDD4, 0xFDF4, 0xFE14, 0xFE34, 0xFE54, 0xFE74, + 0xFE8C, 0xFE9C, 0xFEAC, 0xFEBC, 0xFECC, 0xFEDC, 0xFEEC, 0xFEFC, 0xFF0C, 0xFF1C, 0xFF2C, 0xFF3C, 0xFF4C, 0xFF5C, 0xFF6C, 0xFF7C, + 0xFF88, 0xFF90, 0xFF98, 0xFFA0, 0xFFA8, 0xFFB0, 0xFFB8, 0xFFC0, 0xFFC8, 0xFFD0, 0xFFD8, 0xFFE0, 0xFFE8, 0xFFF0, 0xFFF8, 0x0000, + 0x7D7C, 0x797C, 0x757C, 0x717C, 0x6D7C, 0x697C, 0x657C, 0x617C, 0x5D7C, 0x597C, 0x557C, 0x517C, 0x4D7C, 0x497C, 0x457C, 0x417C, + 0x3E7C, 0x3C7C, 0x3A7C, 0x387C, 0x367C, 0x347C, 0x327C, 0x307C, 0x2E7C, 0x2C7C, 0x2A7C, 0x287C, 0x267C, 0x247C, 0x227C, 0x207C, + 0x1EFC, 0x1DFC, 0x1CFC, 0x1BFC, 0x1AFC, 0x19FC, 0x18FC, 0x17FC, 0x16FC, 0x15FC, 0x14FC, 0x13FC, 0x12FC, 0x11FC, 0x10FC, 0x0FFC, + 0x0F3C, 0x0EBC, 0x0E3C, 0x0DBC, 0x0D3C, 0x0CBC, 0x0C3C, 0x0BBC, 0x0B3C, 0x0ABC, 0x0A3C, 0x09BC, 0x093C, 0x08BC, 0x083C, 0x07BC, + 0x075C, 0x071C, 0x06DC, 0x069C, 0x065C, 0x061C, 0x05DC, 0x059C, 0x055C, 0x051C, 0x04DC, 0x049C, 0x045C, 0x041C, 0x03DC, 0x039C, + 0x036C, 0x034C, 0x032C, 0x030C, 0x02EC, 0x02CC, 0x02AC, 0x028C, 0x026C, 0x024C, 0x022C, 0x020C, 0x01EC, 0x01CC, 0x01AC, 0x018C, + 0x0174, 0x0164, 0x0154, 0x0144, 0x0134, 0x0124, 0x0114, 0x0104, 0x00F4, 0x00E4, 0x00D4, 0x00C4, 0x00B4, 0x00A4, 0x0094, 0x0084, + 0x0078, 0x0070, 0x0068, 0x0060, 0x0058, 0x0050, 0x0048, 0x0040, 0x0038, 0x0030, 0x0028, 0x0020, 0x0018, 0x0010, 0x0008, 0x0000 +}; + +static DRWAV_INLINE drwav_int16 drwav__alaw_to_s16(drwav_uint8 sampleIn) +{ + return (short)g_drwavAlawTable[sampleIn]; +} + +static DRWAV_INLINE drwav_int16 drwav__mulaw_to_s16(drwav_uint8 sampleIn) +{ + return (short)g_drwavMulawTable[sampleIn]; +} + + + +static void drwav__pcm_to_s16(drwav_int16* pOut, const unsigned char* pIn, size_t totalSampleCount, unsigned int bytesPerSample) +{ + unsigned int i; + + /* Special case for 8-bit sample data because it's treated as unsigned. */ + if (bytesPerSample == 1) { + drwav_u8_to_s16(pOut, pIn, totalSampleCount); + return; + } + + + /* Slightly more optimal implementation for common formats. */ + if (bytesPerSample == 2) { + for (i = 0; i < totalSampleCount; ++i) { + *pOut++ = ((const drwav_int16*)pIn)[i]; + } + return; + } + if (bytesPerSample == 3) { + drwav_s24_to_s16(pOut, pIn, totalSampleCount); + return; + } + if (bytesPerSample == 4) { + drwav_s32_to_s16(pOut, (const drwav_int32*)pIn, totalSampleCount); + return; + } + + + /* Anything more than 64 bits per sample is not supported. */ + if (bytesPerSample > 8) { + drwav_zero_memory(pOut, totalSampleCount * sizeof(*pOut)); + return; + } + + + /* Generic, slow converter. */ + for (i = 0; i < totalSampleCount; ++i) { + drwav_uint64 sample = 0; + unsigned int shift = (8 - bytesPerSample) * 8; + + unsigned int j; + for (j = 0; j < bytesPerSample && j < 8; j += 1) { + sample |= (drwav_uint64)(pIn[j]) << shift; + shift += 8; + } + + pIn += j; + *pOut++ = (drwav_int16)((drwav_int64)sample >> 48); + } +} + +static void drwav__ieee_to_s16(drwav_int16* pOut, const unsigned char* pIn, size_t totalSampleCount, unsigned int bytesPerSample) +{ + if (bytesPerSample == 4) { + drwav_f32_to_s16(pOut, (const float*)pIn, totalSampleCount); + return; + } else if (bytesPerSample == 8) { + drwav_f64_to_s16(pOut, (const double*)pIn, totalSampleCount); + return; + } else { + /* Only supporting 32- and 64-bit float. Output silence in all other cases. Contributions welcome for 16-bit float. */ + drwav_zero_memory(pOut, totalSampleCount * sizeof(*pOut)); + return; + } +} + +drwav_uint64 drwav_read_s16__pcm(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut) +{ + drwav_uint32 bytesPerSample; + drwav_uint64 totalSamplesRead; + unsigned char sampleData[4096]; + + /* Fast path. */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM && pWav->bitsPerSample == 16) { + return drwav_read(pWav, samplesToRead, pBufferOut); + } + + bytesPerSample = drwav_get_bytes_per_sample(pWav); + if (bytesPerSample == 0) { + return 0; + } + + totalSamplesRead = 0; + + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav__pcm_to_s16(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s16__ieee(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalSamplesRead; + unsigned char sampleData[4096]; + + drwav_uint32 bytesPerSample = drwav_get_bytes_per_sample(pWav); + if (bytesPerSample == 0) { + return 0; + } + + totalSamplesRead = 0; + + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav__ieee_to_s16(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s16__alaw(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalSamplesRead; + unsigned char sampleData[4096]; + + drwav_uint32 bytesPerSample = drwav_get_bytes_per_sample(pWav); + if (bytesPerSample == 0) { + return 0; + } + + totalSamplesRead = 0; + + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav_alaw_to_s16(pBufferOut, sampleData, (size_t)samplesRead); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s16__mulaw(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalSamplesRead; + unsigned char sampleData[4096]; + + drwav_uint32 bytesPerSample = drwav_get_bytes_per_sample(pWav); + if (bytesPerSample == 0) { + return 0; + } + + totalSamplesRead = 0; + + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav_mulaw_to_s16(pBufferOut, sampleData, (size_t)samplesRead); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s16(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut) +{ + if (pWav == NULL || samplesToRead == 0 || pBufferOut == NULL) { + return 0; + } + + /* Don't try to read more samples than can potentially fit in the output buffer. */ + if (samplesToRead * sizeof(drwav_int16) > DRWAV_SIZE_MAX) { + samplesToRead = DRWAV_SIZE_MAX / sizeof(drwav_int16); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) { + return drwav_read_s16__pcm(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + return drwav_read_s16__msadpcm(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) { + return drwav_read_s16__ieee(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) { + return drwav_read_s16__alaw(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) { + return drwav_read_s16__mulaw(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + return drwav_read_s16__ima(pWav, samplesToRead, pBufferOut); + } + + return 0; +} + +drwav_uint64 drwav_read_pcm_frames_s16(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + return drwav_read_s16(pWav, framesToRead * pWav->channels, pBufferOut) / pWav->channels; +} + +void drwav_u8_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + int r; + size_t i; + for (i = 0; i < sampleCount; ++i) { + int x = pIn[i]; + r = x << 8; + r = r - 32768; + pOut[i] = (short)r; + } +} + +void drwav_s24_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + int r; + size_t i; + for (i = 0; i < sampleCount; ++i) { + int x = ((int)(((unsigned int)(((const unsigned char*)pIn)[i*3+0]) << 8) | ((unsigned int)(((const unsigned char*)pIn)[i*3+1]) << 16) | ((unsigned int)(((const unsigned char*)pIn)[i*3+2])) << 24)) >> 8; + r = x >> 8; + pOut[i] = (short)r; + } +} + +void drwav_s32_to_s16(drwav_int16* pOut, const drwav_int32* pIn, size_t sampleCount) +{ + int r; + size_t i; + for (i = 0; i < sampleCount; ++i) { + int x = pIn[i]; + r = x >> 16; + pOut[i] = (short)r; + } +} + +void drwav_f32_to_s16(drwav_int16* pOut, const float* pIn, size_t sampleCount) +{ + int r; + size_t i; + for (i = 0; i < sampleCount; ++i) { + float x = pIn[i]; + float c; + c = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); + c = c + 1; + r = (int)(c * 32767.5f); + r = r - 32768; + pOut[i] = (short)r; + } +} + +void drwav_f64_to_s16(drwav_int16* pOut, const double* pIn, size_t sampleCount) +{ + int r; + size_t i; + for (i = 0; i < sampleCount; ++i) { + double x = pIn[i]; + double c; + c = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); + c = c + 1; + r = (int)(c * 32767.5); + r = r - 32768; + pOut[i] = (short)r; + } +} + +void drwav_alaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + for (i = 0; i < sampleCount; ++i) { + pOut[i] = drwav__alaw_to_s16(pIn[i]); + } +} + +void drwav_mulaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + for (i = 0; i < sampleCount; ++i) { + pOut[i] = drwav__mulaw_to_s16(pIn[i]); + } +} + + + +static void drwav__pcm_to_f32(float* pOut, const unsigned char* pIn, size_t sampleCount, unsigned int bytesPerSample) +{ + unsigned int i; + + /* Special case for 8-bit sample data because it's treated as unsigned. */ + if (bytesPerSample == 1) { + drwav_u8_to_f32(pOut, pIn, sampleCount); + return; + } + + /* Slightly more optimal implementation for common formats. */ + if (bytesPerSample == 2) { + drwav_s16_to_f32(pOut, (const drwav_int16*)pIn, sampleCount); + return; + } + if (bytesPerSample == 3) { + drwav_s24_to_f32(pOut, pIn, sampleCount); + return; + } + if (bytesPerSample == 4) { + drwav_s32_to_f32(pOut, (const drwav_int32*)pIn, sampleCount); + return; + } + + + /* Anything more than 64 bits per sample is not supported. */ + if (bytesPerSample > 8) { + drwav_zero_memory(pOut, sampleCount * sizeof(*pOut)); + return; + } + + + /* Generic, slow converter. */ + for (i = 0; i < sampleCount; ++i) { + drwav_uint64 sample = 0; + unsigned int shift = (8 - bytesPerSample) * 8; + + unsigned int j; + for (j = 0; j < bytesPerSample && j < 8; j += 1) { + sample |= (drwav_uint64)(pIn[j]) << shift; + shift += 8; + } + + pIn += j; + *pOut++ = (float)((drwav_int64)sample / 9223372036854775807.0); + } +} + +static void drwav__ieee_to_f32(float* pOut, const unsigned char* pIn, size_t sampleCount, unsigned int bytesPerSample) +{ + if (bytesPerSample == 4) { + unsigned int i; + for (i = 0; i < sampleCount; ++i) { + *pOut++ = ((const float*)pIn)[i]; + } + return; + } else if (bytesPerSample == 8) { + drwav_f64_to_f32(pOut, (const double*)pIn, sampleCount); + return; + } else { + /* Only supporting 32- and 64-bit float. Output silence in all other cases. Contributions welcome for 16-bit float. */ + drwav_zero_memory(pOut, sampleCount * sizeof(*pOut)); + return; + } +} + + +drwav_uint64 drwav_read_f32__pcm(drwav* pWav, drwav_uint64 samplesToRead, float* pBufferOut) +{ + drwav_uint64 totalSamplesRead; + unsigned char sampleData[4096]; + + drwav_uint32 bytesPerSample = drwav_get_bytes_per_sample(pWav); + if (bytesPerSample == 0) { + return 0; + } + + totalSamplesRead = 0; + + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav__pcm_to_f32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); + pBufferOut += samplesRead; + + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_f32__msadpcm(drwav* pWav, drwav_uint64 samplesToRead, float* pBufferOut) +{ + /* + We're just going to borrow the implementation from the drwav_read_s16() since ADPCM is a little bit more complicated than other formats and I don't + want to duplicate that code. + */ + drwav_uint64 totalSamplesRead = 0; + drwav_int16 samples16[2048]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read_s16(pWav, drwav_min(samplesToRead, 2048), samples16); + if (samplesRead == 0) { + break; + } + + drwav_s16_to_f32(pBufferOut, samples16, (size_t)samplesRead); /* <-- Safe cast because we're clamping to 2048. */ + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_f32__ima(drwav* pWav, drwav_uint64 samplesToRead, float* pBufferOut) +{ + /* + We're just going to borrow the implementation from the drwav_read_s16() since IMA-ADPCM is a little bit more complicated than other formats and I don't + want to duplicate that code. + */ + drwav_uint64 totalSamplesRead = 0; + drwav_int16 samples16[2048]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read_s16(pWav, drwav_min(samplesToRead, 2048), samples16); + if (samplesRead == 0) { + break; + } + + drwav_s16_to_f32(pBufferOut, samples16, (size_t)samplesRead); /* <-- Safe cast because we're clamping to 2048. */ + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_f32__ieee(drwav* pWav, drwav_uint64 samplesToRead, float* pBufferOut) +{ + drwav_uint64 totalSamplesRead; + unsigned char sampleData[4096]; + drwav_uint32 bytesPerSample; + + /* Fast path. */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT && pWav->bitsPerSample == 32) { + return drwav_read(pWav, samplesToRead, pBufferOut); + } + + bytesPerSample = drwav_get_bytes_per_sample(pWav); + if (bytesPerSample == 0) { + return 0; + } + + totalSamplesRead = 0; + + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav__ieee_to_f32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_f32__alaw(drwav* pWav, drwav_uint64 samplesToRead, float* pBufferOut) +{ + drwav_uint64 totalSamplesRead; + unsigned char sampleData[4096]; + drwav_uint32 bytesPerSample = drwav_get_bytes_per_sample(pWav); + if (bytesPerSample == 0) { + return 0; + } + + totalSamplesRead = 0; + + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav_alaw_to_f32(pBufferOut, sampleData, (size_t)samplesRead); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_f32__mulaw(drwav* pWav, drwav_uint64 samplesToRead, float* pBufferOut) +{ + drwav_uint64 totalSamplesRead; + unsigned char sampleData[4096]; + + drwav_uint32 bytesPerSample = drwav_get_bytes_per_sample(pWav); + if (bytesPerSample == 0) { + return 0; + } + + totalSamplesRead = 0; + + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav_mulaw_to_f32(pBufferOut, sampleData, (size_t)samplesRead); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_f32(drwav* pWav, drwav_uint64 samplesToRead, float* pBufferOut) +{ + if (pWav == NULL || samplesToRead == 0 || pBufferOut == NULL) { + return 0; + } + + /* Don't try to read more samples than can potentially fit in the output buffer. */ + if (samplesToRead * sizeof(float) > DRWAV_SIZE_MAX) { + samplesToRead = DRWAV_SIZE_MAX / sizeof(float); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) { + return drwav_read_f32__pcm(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + return drwav_read_f32__msadpcm(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) { + return drwav_read_f32__ieee(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) { + return drwav_read_f32__alaw(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) { + return drwav_read_f32__mulaw(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + return drwav_read_f32__ima(pWav, samplesToRead, pBufferOut); + } + + return 0; +} + +drwav_uint64 drwav_read_pcm_frames_f32(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + return drwav_read_f32(pWav, framesToRead * pWav->channels, pBufferOut) / pWav->channels; +} + +void drwav_u8_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + +#ifdef DR_WAV_LIBSNDFILE_COMPAT + /* + It appears libsndfile uses slightly different logic for the u8 -> f32 conversion to dr_wav, which in my opinion is incorrect. It appears + libsndfile performs the conversion something like "f32 = (u8 / 256) * 2 - 1", however I think it should be "f32 = (u8 / 255) * 2 - 1" (note + the divisor of 256 vs 255). I use libsndfile as a benchmark for testing, so I'm therefore leaving this block here just for my automated + correctness testing. This is disabled by default. + */ + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (pIn[i] / 256.0f) * 2 - 1; + } +#else + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (pIn[i] / 255.0f) * 2 - 1; + } +#endif +} + +void drwav_s16_to_f32(float* pOut, const drwav_int16* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = pIn[i] / 32768.0f; + } +} + +void drwav_s24_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + unsigned int s0 = pIn[i*3 + 0]; + unsigned int s1 = pIn[i*3 + 1]; + unsigned int s2 = pIn[i*3 + 2]; + + int sample32 = (int)((s0 << 8) | (s1 << 16) | (s2 << 24)); + *pOut++ = (float)(sample32 / 2147483648.0); + } +} + +void drwav_s32_to_f32(float* pOut, const drwav_int32* pIn, size_t sampleCount) +{ + size_t i; + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (float)(pIn[i] / 2147483648.0); + } +} + +void drwav_f64_to_f32(float* pOut, const double* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (float)pIn[i]; + } +} + +void drwav_alaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = drwav__alaw_to_s16(pIn[i]) / 32768.0f; + } +} + +void drwav_mulaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = drwav__mulaw_to_s16(pIn[i]) / 32768.0f; + } +} + + + +static void drwav__pcm_to_s32(drwav_int32* pOut, const unsigned char* pIn, size_t totalSampleCount, unsigned int bytesPerSample) +{ + unsigned int i; + + /* Special case for 8-bit sample data because it's treated as unsigned. */ + if (bytesPerSample == 1) { + drwav_u8_to_s32(pOut, pIn, totalSampleCount); + return; + } + + /* Slightly more optimal implementation for common formats. */ + if (bytesPerSample == 2) { + drwav_s16_to_s32(pOut, (const drwav_int16*)pIn, totalSampleCount); + return; + } + if (bytesPerSample == 3) { + drwav_s24_to_s32(pOut, pIn, totalSampleCount); + return; + } + if (bytesPerSample == 4) { + for (i = 0; i < totalSampleCount; ++i) { + *pOut++ = ((const drwav_int32*)pIn)[i]; + } + return; + } + + + /* Anything more than 64 bits per sample is not supported. */ + if (bytesPerSample > 8) { + drwav_zero_memory(pOut, totalSampleCount * sizeof(*pOut)); + return; + } + + + /* Generic, slow converter. */ + for (i = 0; i < totalSampleCount; ++i) { + drwav_uint64 sample = 0; + unsigned int shift = (8 - bytesPerSample) * 8; + + unsigned int j; + for (j = 0; j < bytesPerSample && j < 8; j += 1) { + sample |= (drwav_uint64)(pIn[j]) << shift; + shift += 8; + } + + pIn += j; + *pOut++ = (drwav_int32)((drwav_int64)sample >> 32); + } +} + +static void drwav__ieee_to_s32(drwav_int32* pOut, const unsigned char* pIn, size_t totalSampleCount, unsigned int bytesPerSample) +{ + if (bytesPerSample == 4) { + drwav_f32_to_s32(pOut, (const float*)pIn, totalSampleCount); + return; + } else if (bytesPerSample == 8) { + drwav_f64_to_s32(pOut, (const double*)pIn, totalSampleCount); + return; + } else { + /* Only supporting 32- and 64-bit float. Output silence in all other cases. Contributions welcome for 16-bit float. */ + drwav_zero_memory(pOut, totalSampleCount * sizeof(*pOut)); + return; + } +} + + +drwav_uint64 drwav_read_s32__pcm(drwav* pWav, drwav_uint64 samplesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 totalSamplesRead; + unsigned char sampleData[4096]; + drwav_uint32 bytesPerSample; + + /* Fast path. */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM && pWav->bitsPerSample == 32) { + return drwav_read(pWav, samplesToRead, pBufferOut); + } + + bytesPerSample = drwav_get_bytes_per_sample(pWav); + if (bytesPerSample == 0) { + return 0; + } + + totalSamplesRead = 0; + + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav__pcm_to_s32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s32__msadpcm(drwav* pWav, drwav_uint64 samplesToRead, drwav_int32* pBufferOut) +{ + /* + We're just going to borrow the implementation from the drwav_read_s16() since ADPCM is a little bit more complicated than other formats and I don't + want to duplicate that code. + */ + drwav_uint64 totalSamplesRead = 0; + drwav_int16 samples16[2048]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read_s16(pWav, drwav_min(samplesToRead, 2048), samples16); + if (samplesRead == 0) { + break; + } + + drwav_s16_to_s32(pBufferOut, samples16, (size_t)samplesRead); /* <-- Safe cast because we're clamping to 2048. */ + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s32__ima(drwav* pWav, drwav_uint64 samplesToRead, drwav_int32* pBufferOut) +{ + /* + We're just going to borrow the implementation from the drwav_read_s16() since IMA-ADPCM is a little bit more complicated than other formats and I don't + want to duplicate that code. + */ + drwav_uint64 totalSamplesRead = 0; + drwav_int16 samples16[2048]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read_s16(pWav, drwav_min(samplesToRead, 2048), samples16); + if (samplesRead == 0) { + break; + } + + drwav_s16_to_s32(pBufferOut, samples16, (size_t)samplesRead); /* <-- Safe cast because we're clamping to 2048. */ + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s32__ieee(drwav* pWav, drwav_uint64 samplesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 totalSamplesRead; + unsigned char sampleData[4096]; + + drwav_uint32 bytesPerSample = drwav_get_bytes_per_sample(pWav); + if (bytesPerSample == 0) { + return 0; + } + + totalSamplesRead = 0; + + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav__ieee_to_s32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s32__alaw(drwav* pWav, drwav_uint64 samplesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 totalSamplesRead; + unsigned char sampleData[4096]; + + drwav_uint32 bytesPerSample = drwav_get_bytes_per_sample(pWav); + if (bytesPerSample == 0) { + return 0; + } + + totalSamplesRead = 0; + + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav_alaw_to_s32(pBufferOut, sampleData, (size_t)samplesRead); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s32__mulaw(drwav* pWav, drwav_uint64 samplesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 totalSamplesRead; + unsigned char sampleData[4096]; + + drwav_uint32 bytesPerSample = drwav_get_bytes_per_sample(pWav); + if (bytesPerSample == 0) { + return 0; + } + + totalSamplesRead = 0; + + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav_mulaw_to_s32(pBufferOut, sampleData, (size_t)samplesRead); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s32(drwav* pWav, drwav_uint64 samplesToRead, drwav_int32* pBufferOut) +{ + if (pWav == NULL || samplesToRead == 0 || pBufferOut == NULL) { + return 0; + } + + /* Don't try to read more samples than can potentially fit in the output buffer. */ + if (samplesToRead * sizeof(drwav_int32) > DRWAV_SIZE_MAX) { + samplesToRead = DRWAV_SIZE_MAX / sizeof(drwav_int32); + } + + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) { + return drwav_read_s32__pcm(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + return drwav_read_s32__msadpcm(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) { + return drwav_read_s32__ieee(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) { + return drwav_read_s32__alaw(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) { + return drwav_read_s32__mulaw(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + return drwav_read_s32__ima(pWav, samplesToRead, pBufferOut); + } + + return 0; +} + +drwav_uint64 drwav_read_pcm_frames_s32(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + return drwav_read_s32(pWav, framesToRead * pWav->channels, pBufferOut) / pWav->channels; +} + +void drwav_u8_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = ((int)pIn[i] - 128) << 24; + } +} + +void drwav_s16_to_s32(drwav_int32* pOut, const drwav_int16* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = pIn[i] << 16; + } +} + +void drwav_s24_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + unsigned int s0 = pIn[i*3 + 0]; + unsigned int s1 = pIn[i*3 + 1]; + unsigned int s2 = pIn[i*3 + 2]; + + drwav_int32 sample32 = (drwav_int32)((s0 << 8) | (s1 << 16) | (s2 << 24)); + *pOut++ = sample32; + } +} + +void drwav_f32_to_s32(drwav_int32* pOut, const float* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (drwav_int32)(2147483648.0 * pIn[i]); + } +} + +void drwav_f64_to_s32(drwav_int32* pOut, const double* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (drwav_int32)(2147483648.0 * pIn[i]); + } +} + +void drwav_alaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = ((drwav_int32)drwav__alaw_to_s16(pIn[i])) << 16; + } +} + +void drwav_mulaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i= 0; i < sampleCount; ++i) { + *pOut++ = ((drwav_int32)drwav__mulaw_to_s16(pIn[i])) << 16; + } +} + + + +drwav_int16* drwav__read_and_close_s16(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + drwav_uint64 sampleDataSize; + drwav_int16* pSampleData; + drwav_uint64 samplesRead; + + drwav_assert(pWav != NULL); + + sampleDataSize = pWav->totalSampleCount * sizeof(drwav_int16); + if (sampleDataSize > DRWAV_SIZE_MAX) { + drwav_uninit(pWav); + return NULL; /* File's too big. */ + } + + pSampleData = (drwav_int16*)DRWAV_MALLOC((size_t)sampleDataSize); /* <-- Safe cast due to the check above. */ + if (pSampleData == NULL) { + drwav_uninit(pWav); + return NULL; /* Failed to allocate memory. */ + } + + samplesRead = drwav_read_s16(pWav, (size_t)pWav->totalSampleCount, pSampleData); + if (samplesRead != pWav->totalSampleCount) { + DRWAV_FREE(pSampleData); + drwav_uninit(pWav); + return NULL; /* There was an error reading the samples. */ + } + + drwav_uninit(pWav); + + if (sampleRate) { + *sampleRate = pWav->sampleRate; + } + if (channels) { + *channels = pWav->channels; + } + if (totalSampleCount) { + *totalSampleCount = pWav->totalSampleCount; + } + + return pSampleData; +} + +float* drwav__read_and_close_f32(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + drwav_uint64 sampleDataSize; + float* pSampleData; + drwav_uint64 samplesRead; + + drwav_assert(pWav != NULL); + + sampleDataSize = pWav->totalSampleCount * sizeof(float); + if (sampleDataSize > DRWAV_SIZE_MAX) { + drwav_uninit(pWav); + return NULL; /* File's too big. */ + } + + pSampleData = (float*)DRWAV_MALLOC((size_t)sampleDataSize); /* <-- Safe cast due to the check above. */ + if (pSampleData == NULL) { + drwav_uninit(pWav); + return NULL; /* Failed to allocate memory. */ + } + + samplesRead = drwav_read_f32(pWav, (size_t)pWav->totalSampleCount, pSampleData); + if (samplesRead != pWav->totalSampleCount) { + DRWAV_FREE(pSampleData); + drwav_uninit(pWav); + return NULL; /* There was an error reading the samples. */ + } + + drwav_uninit(pWav); + + if (sampleRate) { + *sampleRate = pWav->sampleRate; + } + if (channels) { + *channels = pWav->channels; + } + if (totalSampleCount) { + *totalSampleCount = pWav->totalSampleCount; + } + + return pSampleData; +} + +drwav_int32* drwav__read_and_close_s32(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + drwav_uint64 sampleDataSize; + drwav_int32* pSampleData; + drwav_uint64 samplesRead; + + drwav_assert(pWav != NULL); + + sampleDataSize = pWav->totalSampleCount * sizeof(drwav_int32); + if (sampleDataSize > DRWAV_SIZE_MAX) { + drwav_uninit(pWav); + return NULL; /* File's too big. */ + } + + pSampleData = (drwav_int32*)DRWAV_MALLOC((size_t)sampleDataSize); /* <-- Safe cast due to the check above. */ + if (pSampleData == NULL) { + drwav_uninit(pWav); + return NULL; /* Failed to allocate memory. */ + } + + samplesRead = drwav_read_s32(pWav, (size_t)pWav->totalSampleCount, pSampleData); + if (samplesRead != pWav->totalSampleCount) { + DRWAV_FREE(pSampleData); + drwav_uninit(pWav); + return NULL; /* There was an error reading the samples. */ + } + + drwav_uninit(pWav); + + if (sampleRate) { + *sampleRate = pWav->sampleRate; + } + if (channels) { + *channels = pWav->channels; + } + if (totalSampleCount) { + *totalSampleCount = pWav->totalSampleCount; + } + + return pSampleData; +} + + +drwav_int16* drwav_open_and_read_s16(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + drwav wav; + + if (channels) { + *channels = 0; + } + if (sampleRate) { + *sampleRate = 0; + } + if (totalSampleCount) { + *totalSampleCount = 0; + } + + if (!drwav_init(&wav, onRead, onSeek, pUserData)) { + return NULL; + } + + return drwav__read_and_close_s16(&wav, channels, sampleRate, totalSampleCount); +} + +drwav_int16* drwav_open_and_read_pcm_frames_s16(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut) +{ + unsigned int channels; + unsigned int sampleRate; + drwav_uint64 totalSampleCount; + drwav_int16* result; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + result = drwav_open_and_read_s16(onRead, onSeek, pUserData, &channels, &sampleRate, &totalSampleCount); + if (result == NULL) { + return NULL; + } + + if (channelsOut) { + *channelsOut = channels; + } + if (sampleRateOut) { + *sampleRateOut = sampleRate; + } + if (totalFrameCountOut) { + *totalFrameCountOut = totalSampleCount / channels; + } + + return result; +} + +float* drwav_open_and_read_f32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + drwav wav; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalSampleCount) { + *totalSampleCount = 0; + } + + if (!drwav_init(&wav, onRead, onSeek, pUserData)) { + return NULL; + } + + return drwav__read_and_close_f32(&wav, channels, sampleRate, totalSampleCount); +} + +float* drwav_open_and_read_pcm_frames_f32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut) +{ + unsigned int channels; + unsigned int sampleRate; + drwav_uint64 totalSampleCount; + float* result; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + result = drwav_open_and_read_f32(onRead, onSeek, pUserData, &channels, &sampleRate, &totalSampleCount); + if (result == NULL) { + return NULL; + } + + if (channelsOut) { + *channelsOut = channels; + } + if (sampleRateOut) { + *sampleRateOut = sampleRate; + } + if (totalFrameCountOut) { + *totalFrameCountOut = totalSampleCount / channels; + } + + return result; +} + +drwav_int32* drwav_open_and_read_s32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + drwav wav; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalSampleCount) { + *totalSampleCount = 0; + } + + if (!drwav_init(&wav, onRead, onSeek, pUserData)) { + return NULL; + } + + return drwav__read_and_close_s32(&wav, channels, sampleRate, totalSampleCount); +} + +drwav_int32* drwav_open_and_read_pcm_frames_s32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut) +{ + unsigned int channels; + unsigned int sampleRate; + drwav_uint64 totalSampleCount; + drwav_int32* result; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + result = drwav_open_and_read_s32(onRead, onSeek, pUserData, &channels, &sampleRate, &totalSampleCount); + if (result == NULL) { + return NULL; + } + + if (channelsOut) { + *channelsOut = channels; + } + if (sampleRateOut) { + *sampleRateOut = sampleRate; + } + if (totalFrameCountOut) { + *totalFrameCountOut = totalSampleCount / channels; + } + + return result; +} + +#ifndef DR_WAV_NO_STDIO +drwav_int16* drwav_open_file_and_read_s16(const char* filename, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + drwav wav; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalSampleCount) { + *totalSampleCount = 0; + } + + if (!drwav_init_file(&wav, filename)) { + return NULL; + } + + return drwav__read_and_close_s16(&wav, channels, sampleRate, totalSampleCount); +} + +drwav_int16* drwav_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut) +{ + unsigned int channels; + unsigned int sampleRate; + drwav_uint64 totalSampleCount; + drwav_int16* result; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + result = drwav_open_file_and_read_s16(filename, &channels, &sampleRate, &totalSampleCount); + if (result == NULL) { + return NULL; + } + + if (channelsOut) { + *channelsOut = channels; + } + if (sampleRateOut) { + *sampleRateOut = sampleRate; + } + if (totalFrameCountOut) { + *totalFrameCountOut = totalSampleCount / channels; + } + + return result; +} + +float* drwav_open_file_and_read_f32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + drwav wav; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalSampleCount) { + *totalSampleCount = 0; + } + + if (!drwav_init_file(&wav, filename)) { + return NULL; + } + + return drwav__read_and_close_f32(&wav, channels, sampleRate, totalSampleCount); +} + +float* drwav_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut) +{ + unsigned int channels; + unsigned int sampleRate; + drwav_uint64 totalSampleCount; + float* result; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + result = drwav_open_file_and_read_f32(filename, &channels, &sampleRate, &totalSampleCount); + if (result == NULL) { + return NULL; + } + + if (channelsOut) { + *channelsOut = channels; + } + if (sampleRateOut) { + *sampleRateOut = sampleRate; + } + if (totalFrameCountOut) { + *totalFrameCountOut = totalSampleCount / channels; + } + + return result; +} + +drwav_int32* drwav_open_file_and_read_s32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + drwav wav; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalSampleCount) { + *totalSampleCount = 0; + } + + if (!drwav_init_file(&wav, filename)) { + return NULL; + } + + return drwav__read_and_close_s32(&wav, channels, sampleRate, totalSampleCount); +} + +drwav_int32* drwav_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut) +{ + unsigned int channels; + unsigned int sampleRate; + drwav_uint64 totalSampleCount; + drwav_int32* result; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + result = drwav_open_file_and_read_s32(filename, &channels, &sampleRate, &totalSampleCount); + if (result == NULL) { + return NULL; + } + + if (channelsOut) { + *channelsOut = channels; + } + if (sampleRateOut) { + *sampleRateOut = sampleRate; + } + if (totalFrameCountOut) { + *totalFrameCountOut = totalSampleCount / channels; + } + + return result; +} +#endif + +drwav_int16* drwav_open_memory_and_read_s16(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + drwav wav; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalSampleCount) { + *totalSampleCount = 0; + } + + if (!drwav_init_memory(&wav, data, dataSize)) { + return NULL; + } + + return drwav__read_and_close_s16(&wav, channels, sampleRate, totalSampleCount); +} + +drwav_int16* drwav_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut) +{ + unsigned int channels; + unsigned int sampleRate; + drwav_uint64 totalSampleCount; + drwav_int16* result; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + result = drwav_open_memory_and_read_s16(data, dataSize, &channels, &sampleRate, &totalSampleCount); + if (result == NULL) { + return NULL; + } + + if (channelsOut) { + *channelsOut = channels; + } + if (sampleRateOut) { + *sampleRateOut = sampleRate; + } + if (totalFrameCountOut) { + *totalFrameCountOut = totalSampleCount / channels; + } + + return result; +} + +float* drwav_open_memory_and_read_f32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + drwav wav; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalSampleCount) { + *totalSampleCount = 0; + } + + if (!drwav_init_memory(&wav, data, dataSize)) { + return NULL; + } + + return drwav__read_and_close_f32(&wav, channels, sampleRate, totalSampleCount); +} + +float* drwav_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut) +{ + unsigned int channels; + unsigned int sampleRate; + drwav_uint64 totalSampleCount; + float* result; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + result = drwav_open_memory_and_read_f32(data, dataSize, &channels, &sampleRate, &totalSampleCount); + if (result == NULL) { + return NULL; + } + + if (channelsOut) { + *channelsOut = channels; + } + if (sampleRateOut) { + *sampleRateOut = sampleRate; + } + if (totalFrameCountOut) { + *totalFrameCountOut = totalSampleCount / channels; + } + + return result; +} + +drwav_int32* drwav_open_memory_and_read_s32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + drwav wav; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalSampleCount) { + *totalSampleCount = 0; + } + + if (!drwav_init_memory(&wav, data, dataSize)) { + return NULL; + } + + return drwav__read_and_close_s32(&wav, channels, sampleRate, totalSampleCount); +} + +drwav_int32* drwav_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut) +{ + unsigned int channels; + unsigned int sampleRate; + drwav_uint64 totalSampleCount; + drwav_int32* result; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + result = drwav_open_memory_and_read_s32(data, dataSize, &channels, &sampleRate, &totalSampleCount); + if (result == NULL) { + return NULL; + } + + if (channelsOut) { + *channelsOut = channels; + } + if (sampleRateOut) { + *sampleRateOut = sampleRate; + } + if (totalFrameCountOut) { + *totalFrameCountOut = totalSampleCount / channels; + } + + return result; +} +#endif /* DR_WAV_NO_CONVERSION_API */ + + +void drwav_free(void* pDataReturnedByOpenAndRead) +{ + DRWAV_FREE(pDataReturnedByOpenAndRead); +} + +#endif /* DR_WAV_IMPLEMENTATION */ + + +/* +REVISION HISTORY +================ +v0.9.2 - 2019-05-21 + - Fix warnings. + +v0.9.1 - 2019-05-05 + - Add support for C89. + - Change license to choice of public domain or MIT-0. + +v0.9.0 - 2018-12-16 + - API CHANGE: Add new reading APIs for reading by PCM frames instead of samples. Old APIs have been deprecated and + will be removed in v0.10.0. Deprecated APIs and their replacements: + drwav_read() -> drwav_read_pcm_frames() + drwav_read_s16() -> drwav_read_pcm_frames_s16() + drwav_read_f32() -> drwav_read_pcm_frames_f32() + drwav_read_s32() -> drwav_read_pcm_frames_s32() + drwav_seek_to_sample() -> drwav_seek_to_pcm_frame() + drwav_write() -> drwav_write_pcm_frames() + drwav_open_and_read_s16() -> drwav_open_and_read_pcm_frames_s16() + drwav_open_and_read_f32() -> drwav_open_and_read_pcm_frames_f32() + drwav_open_and_read_s32() -> drwav_open_and_read_pcm_frames_s32() + drwav_open_file_and_read_s16() -> drwav_open_file_and_read_pcm_frames_s16() + drwav_open_file_and_read_f32() -> drwav_open_file_and_read_pcm_frames_f32() + drwav_open_file_and_read_s32() -> drwav_open_file_and_read_pcm_frames_s32() + drwav_open_memory_and_read_s16() -> drwav_open_memory_and_read_pcm_frames_s16() + drwav_open_memory_and_read_f32() -> drwav_open_memory_and_read_pcm_frames_f32() + drwav_open_memory_and_read_s32() -> drwav_open_memory_and_read_pcm_frames_s32() + drwav::totalSampleCount -> drwav::totalPCMFrameCount + - API CHANGE: Rename drwav_open_and_read_file_*() to drwav_open_file_and_read_*(). + - API CHANGE: Rename drwav_open_and_read_memory_*() to drwav_open_memory_and_read_*(). + - Add built-in support for smpl chunks. + - Add support for firing a callback for each chunk in the file at initialization time. + - This is enabled through the drwav_init_ex(), etc. family of APIs. + - Handle invalid FMT chunks more robustly. + +v0.8.5 - 2018-09-11 + - Const correctness. + - Fix a potential stack overflow. + +v0.8.4 - 2018-08-07 + - Improve 64-bit detection. + +v0.8.3 - 2018-08-05 + - Fix C++ build on older versions of GCC. + +v0.8.2 - 2018-08-02 + - Fix some big-endian bugs. + +v0.8.1 - 2018-06-29 + - Add support for sequential writing APIs. + - Disable seeking in write mode. + - Fix bugs with Wave64. + - Fix typos. + +v0.8 - 2018-04-27 + - Bug fix. + - Start using major.minor.revision versioning. + +v0.7f - 2018-02-05 + - Restrict ADPCM formats to a maximum of 2 channels. + +v0.7e - 2018-02-02 + - Fix a crash. + +v0.7d - 2018-02-01 + - Fix a crash. + +v0.7c - 2018-02-01 + - Set drwav.bytesPerSample to 0 for all compressed formats. + - Fix a crash when reading 16-bit floating point WAV files. In this case dr_wav will output silence for + all format conversion reading APIs (*_s16, *_s32, *_f32 APIs). + - Fix some divide-by-zero errors. + +v0.7b - 2018-01-22 + - Fix errors with seeking of compressed formats. + - Fix compilation error when DR_WAV_NO_CONVERSION_API + +v0.7a - 2017-11-17 + - Fix some GCC warnings. + +v0.7 - 2017-11-04 + - Add writing APIs. + +v0.6 - 2017-08-16 + - API CHANGE: Rename dr_* types to drwav_*. + - Add support for custom implementations of malloc(), realloc(), etc. + - Add support for Microsoft ADPCM. + - Add support for IMA ADPCM (DVI, format code 0x11). + - Optimizations to drwav_read_s16(). + - Bug fixes. + +v0.5g - 2017-07-16 + - Change underlying type for booleans to unsigned. + +v0.5f - 2017-04-04 + - Fix a minor bug with drwav_open_and_read_s16() and family. + +v0.5e - 2016-12-29 + - Added support for reading samples as signed 16-bit integers. Use the _s16() family of APIs for this. + - Minor fixes to documentation. + +v0.5d - 2016-12-28 + - Use drwav_int* and drwav_uint* sized types to improve compiler support. + +v0.5c - 2016-11-11 + - Properly handle JUNK chunks that come before the FMT chunk. + +v0.5b - 2016-10-23 + - A minor change to drwav_bool8 and drwav_bool32 types. + +v0.5a - 2016-10-11 + - Fixed a bug with drwav_open_and_read() and family due to incorrect argument ordering. + - Improve A-law and mu-law efficiency. + +v0.5 - 2016-09-29 + - API CHANGE. Swap the order of "channels" and "sampleRate" parameters in drwav_open_and_read*(). Rationale for this is to + keep it consistent with dr_audio and dr_flac. + +v0.4b - 2016-09-18 + - Fixed a typo in documentation. + +v0.4a - 2016-09-18 + - Fixed a typo. + - Change date format to ISO 8601 (YYYY-MM-DD) + +v0.4 - 2016-07-13 + - API CHANGE. Make onSeek consistent with dr_flac. + - API CHANGE. Rename drwav_seek() to drwav_seek_to_sample() for clarity and consistency with dr_flac. + - Added support for Sony Wave64. + +v0.3a - 2016-05-28 + - API CHANGE. Return drwav_bool32 instead of int in onSeek callback. + - Fixed a memory leak. + +v0.3 - 2016-05-22 + - Lots of API changes for consistency. + +v0.2a - 2016-05-16 + - Fixed Linux/GCC build. + +v0.2 - 2016-05-11 + - Added support for reading data as signed 32-bit PCM for consistency with dr_flac. + +v0.1a - 2016-05-07 + - Fixed a bug in drwav_open_file() where the file handle would not be closed if the loader failed to initialize. + +v0.1 - 2016-05-04 + - Initial versioned release. +*/ + +/* +This software is available as a choice of the following licenses. Choose +whichever you prefer. + +=============================================================================== +ALTERNATIVE 1 - Public Domain (www.unlicense.org) +=============================================================================== +This is free and unencumbered software released into the public domain. + +Anyone is free to copy, modify, publish, use, compile, sell, or distribute this +software, either in source code form or as a compiled binary, for any purpose, +commercial or non-commercial, and by any means. + +In jurisdictions that recognize copyright laws, the author or authors of this +software dedicate any and all copyright interest in the software to the public +domain. We make this dedication for the benefit of the public at large and to +the detriment of our heirs and successors. We intend this dedication to be an +overt act of relinquishment in perpetuity of all present and future rights to +this software under copyright law. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN +ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION +WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + +For more information, please refer to <http://unlicense.org/> + +=============================================================================== +ALTERNATIVE 2 - MIT No Attribution +=============================================================================== +Copyright 2018 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 +the Software without restriction, including without limitation the rights to +use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies +of the Software, and to permit persons to whom the Software is furnished to do +so. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +SOFTWARE. +*/ |