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Diffstat (limited to 'libs/raylib/src/raudio.c')
| -rw-r--r-- | libs/raylib/src/raudio.c | 2414 |
1 files changed, 0 insertions, 2414 deletions
diff --git a/libs/raylib/src/raudio.c b/libs/raylib/src/raudio.c deleted file mode 100644 index 2b6b0a1..0000000 --- a/libs/raylib/src/raudio.c +++ /dev/null @@ -1,2414 +0,0 @@ -/********************************************************************************************** -* -* raudio v1.0 - A simple and easy-to-use audio library based on miniaudio -* -* FEATURES: -* - Manage audio device (init/close) -* - Manage raw audio context -* - Manage mixing channels -* - Load and unload audio files -* - Format wave data (sample rate, size, channels) -* - Play/Stop/Pause/Resume loaded audio -* -* CONFIGURATION: -* -* #define RAUDIO_STANDALONE -* Define to use the module as standalone library (independently of raylib). -* Required types and functions are defined in the same module. -* -* #define SUPPORT_FILEFORMAT_WAV -* #define SUPPORT_FILEFORMAT_OGG -* #define SUPPORT_FILEFORMAT_XM -* #define SUPPORT_FILEFORMAT_MOD -* #define SUPPORT_FILEFORMAT_FLAC -* #define SUPPORT_FILEFORMAT_MP3 -* Selected desired fileformats to be supported for loading. Some of those formats are -* supported by default, to remove support, just comment unrequired #define in this module -* -* DEPENDENCIES: -* miniaudio.h - Audio device management lib (https://github.com/dr-soft/miniaudio) -* stb_vorbis.h - Ogg audio files loading (http://www.nothings.org/stb_vorbis/) -* dr_mp3.h - MP3 audio file loading (https://github.com/mackron/dr_libs) -* dr_flac.h - FLAC audio file loading (https://github.com/mackron/dr_libs) -* jar_xm.h - XM module file loading -* jar_mod.h - MOD audio file loading -* -* CONTRIBUTORS: -* David Reid (github: @mackron) (Nov. 2017): -* - Complete port to miniaudio library -* -* Joshua Reisenauer (github: @kd7tck) (2015) -* - XM audio module support (jar_xm) -* - MOD audio module support (jar_mod) -* - Mixing channels support -* - Raw audio context support -* -* -* LICENSE: zlib/libpng -* -* Copyright (c) 2013-2021 Ramon Santamaria (@raysan5) -* -* This software is provided "as-is", without any express or implied warranty. In no event -* will the authors be held liable for any damages arising from the use of this software. -* -* Permission is granted to anyone to use this software for any purpose, including commercial -* applications, and to alter it and redistribute it freely, subject to the following restrictions: -* -* 1. The origin of this software must not be misrepresented; you must not claim that you -* wrote the original software. If you use this software in a product, an acknowledgment -* in the product documentation would be appreciated but is not required. -* -* 2. Altered source versions must be plainly marked as such, and must not be misrepresented -* as being the original software. -* -* 3. This notice may not be removed or altered from any source distribution. -* -**********************************************************************************************/ - -#if defined(RAUDIO_STANDALONE) - #include "raudio.h" - #include <stdarg.h> // Required for: va_list, va_start(), vfprintf(), va_end() -#else - #include "raylib.h" // Declares module functions - -// Check if config flags have been externally provided on compilation line -#if !defined(EXTERNAL_CONFIG_FLAGS) - #include "config.h" // Defines module configuration flags -#endif - #include "utils.h" // Required for: fopen() Android mapping -#endif - -#if defined(_WIN32) -// To avoid conflicting windows.h symbols with raylib, some flags are defined -// WARNING: Those flags avoid inclusion of some Win32 headers that could be required -// by user at some point and won't be included... -//------------------------------------------------------------------------------------- - -// If defined, the following flags inhibit definition of the indicated items. -#define NOGDICAPMASKS // CC_*, LC_*, PC_*, CP_*, TC_*, RC_ -#define NOVIRTUALKEYCODES // VK_* -#define NOWINMESSAGES // WM_*, EM_*, LB_*, CB_* -#define NOWINSTYLES // WS_*, CS_*, ES_*, LBS_*, SBS_*, CBS_* -#define NOSYSMETRICS // SM_* -#define NOMENUS // MF_* -#define NOICONS // IDI_* -#define NOKEYSTATES // MK_* -#define NOSYSCOMMANDS // SC_* -#define NORASTEROPS // Binary and Tertiary raster ops -#define NOSHOWWINDOW // SW_* -#define OEMRESOURCE // OEM Resource values -#define NOATOM // Atom Manager routines -#define NOCLIPBOARD // Clipboard routines -#define NOCOLOR // Screen colors -#define NOCTLMGR // Control and Dialog routines -#define NODRAWTEXT // DrawText() and DT_* -#define NOGDI // All GDI defines and routines -#define NOKERNEL // All KERNEL defines and routines -#define NOUSER // All USER defines and routines -//#define NONLS // All NLS defines and routines -#define NOMB // MB_* and MessageBox() -#define NOMEMMGR // GMEM_*, LMEM_*, GHND, LHND, associated routines -#define NOMETAFILE // typedef METAFILEPICT -#define NOMINMAX // Macros min(a,b) and max(a,b) -#define NOMSG // typedef MSG and associated routines -#define NOOPENFILE // OpenFile(), OemToAnsi, AnsiToOem, and OF_* -#define NOSCROLL // SB_* and scrolling routines -#define NOSERVICE // All Service Controller routines, SERVICE_ equates, etc. -#define NOSOUND // Sound driver routines -#define NOTEXTMETRIC // typedef TEXTMETRIC and associated routines -#define NOWH // SetWindowsHook and WH_* -#define NOWINOFFSETS // GWL_*, GCL_*, associated routines -#define NOCOMM // COMM driver routines -#define NOKANJI // Kanji support stuff. -#define NOHELP // Help engine interface. -#define NOPROFILER // Profiler interface. -#define NODEFERWINDOWPOS // DeferWindowPos routines -#define NOMCX // Modem Configuration Extensions - -// Type required before windows.h inclusion -typedef struct tagMSG *LPMSG; - -#include <windows.h> - -// Type required by some unused function... -typedef struct tagBITMAPINFOHEADER { - DWORD biSize; - LONG biWidth; - LONG biHeight; - WORD biPlanes; - WORD biBitCount; - DWORD biCompression; - DWORD biSizeImage; - LONG biXPelsPerMeter; - LONG biYPelsPerMeter; - DWORD biClrUsed; - DWORD biClrImportant; -} BITMAPINFOHEADER, *PBITMAPINFOHEADER; - -#include <objbase.h> -#include <mmreg.h> -#include <mmsystem.h> - -// Some required types defined for MSVC/TinyC compiler -#if defined(_MSC_VER) || defined(__TINYC__) - #include "propidl.h" -#endif -#endif - -#define MA_MALLOC RL_MALLOC -#define MA_FREE RL_FREE - -#define MA_NO_JACK -#define MA_NO_WAV -#define MA_NO_FLAC -#define MA_NO_MP3 -#define MINIAUDIO_IMPLEMENTATION -//#define MA_DEBUG_OUTPUT -#include "external/miniaudio.h" // miniaudio library -#undef PlaySound // Win32 API: windows.h > mmsystem.h defines PlaySound macro - -#include <stdlib.h> // Required for: malloc(), free() -#include <stdio.h> // Required for: FILE, fopen(), fclose(), fread() - -#if defined(RAUDIO_STANDALONE) - #include <string.h> // Required for: strcmp() [Used in IsFileExtension()] - - #if !defined(TRACELOG) - #define TRACELOG(level, ...) (void)0 - #endif - - // Allow custom memory allocators - #ifndef RL_MALLOC - #define RL_MALLOC(sz) malloc(sz) - #endif - #ifndef RL_CALLOC - #define RL_CALLOC(n,sz) calloc(n,sz) - #endif - #ifndef RL_REALLOC - #define RL_REALLOC(ptr,sz) realloc(ptr,sz) - #endif - #ifndef RL_FREE - #define RL_FREE(ptr) free(ptr) - #endif -#endif - -#if defined(SUPPORT_FILEFORMAT_OGG) - // TODO: Remap malloc()/free() calls to RL_MALLOC/RL_FREE - - #define STB_VORBIS_IMPLEMENTATION - #include "external/stb_vorbis.h" // OGG loading functions -#endif - -#if defined(SUPPORT_FILEFORMAT_XM) - #define JARXM_MALLOC RL_MALLOC - #define JARXM_FREE RL_FREE - - #define JAR_XM_IMPLEMENTATION - #include "external/jar_xm.h" // XM loading functions -#endif - -#if defined(SUPPORT_FILEFORMAT_MOD) - #define JARMOD_MALLOC RL_MALLOC - #define JARMOD_FREE RL_FREE - - #define JAR_MOD_IMPLEMENTATION - #include "external/jar_mod.h" // MOD loading functions -#endif - -#if defined(SUPPORT_FILEFORMAT_WAV) - #define DRWAV_MALLOC RL_MALLOC - #define DRWAV_REALLOC RL_REALLOC - #define DRWAV_FREE RL_FREE - - #define DR_WAV_IMPLEMENTATION - #include "external/dr_wav.h" // WAV loading functions -#endif - -#if defined(SUPPORT_FILEFORMAT_MP3) - #define DRMP3_MALLOC RL_MALLOC - #define DRMP3_REALLOC RL_REALLOC - #define DRMP3_FREE RL_FREE - - #define DR_MP3_IMPLEMENTATION - #include "external/dr_mp3.h" // MP3 loading functions -#endif - -#if defined(SUPPORT_FILEFORMAT_FLAC) - #define DRFLAC_MALLOC RL_MALLOC - #define DRFLAC_REALLOC RL_REALLOC - #define DRFLAC_FREE RL_FREE - - #define DR_FLAC_IMPLEMENTATION - #define DR_FLAC_NO_WIN32_IO - #include "external/dr_flac.h" // FLAC loading functions -#endif - -#if defined(_MSC_VER) - #undef bool -#endif - -//---------------------------------------------------------------------------------- -// Defines and Macros -//---------------------------------------------------------------------------------- -#ifndef AUDIO_DEVICE_FORMAT - #define AUDIO_DEVICE_FORMAT ma_format_f32 // Device output format (float-32bit) -#endif -#ifndef AUDIO_DEVICE_CHANNELS - #define AUDIO_DEVICE_CHANNELS 2 // Device output channels: stereo -#endif - -#ifndef AUDIO_DEVICE_SAMPLE_RATE - #define AUDIO_DEVICE_SAMPLE_RATE 0 // Device output channels: stereo -#endif -#ifndef MAX_AUDIO_BUFFER_POOL_CHANNELS - #define MAX_AUDIO_BUFFER_POOL_CHANNELS 16 // Audio pool channels -#endif -#ifndef DEFAULT_AUDIO_BUFFER_SIZE - #define DEFAULT_AUDIO_BUFFER_SIZE 4096 // Default audio buffer size -#endif - - -//---------------------------------------------------------------------------------- -// Types and Structures Definition -//---------------------------------------------------------------------------------- - -// Music context type -// NOTE: Depends on data structure provided by the library -// in charge of reading the different file types -typedef enum { - MUSIC_AUDIO_NONE = 0, - MUSIC_AUDIO_WAV, - MUSIC_AUDIO_OGG, - MUSIC_AUDIO_FLAC, - MUSIC_AUDIO_MP3, - MUSIC_MODULE_XM, - MUSIC_MODULE_MOD -} MusicContextType; - -#if defined(RAUDIO_STANDALONE) -typedef enum { - LOG_ALL, - LOG_TRACE, - LOG_DEBUG, - LOG_INFO, - LOG_WARNING, - LOG_ERROR, - LOG_FATAL, - LOG_NONE -} TraceLogLevel; -#endif - -// NOTE: Different logic is used when feeding data to the playback device -// depending on whether or not data is streamed (Music vs Sound) -typedef enum { - AUDIO_BUFFER_USAGE_STATIC = 0, - AUDIO_BUFFER_USAGE_STREAM -} AudioBufferUsage; - -// Audio buffer structure -struct rAudioBuffer { - ma_data_converter converter; // Audio data converter - - float volume; // Audio buffer volume - float pitch; // Audio buffer pitch - - bool playing; // Audio buffer state: AUDIO_PLAYING - bool paused; // Audio buffer state: AUDIO_PAUSED - bool looping; // Audio buffer looping, always true for AudioStreams - int usage; // Audio buffer usage mode: STATIC or STREAM - - bool isSubBufferProcessed[2]; // SubBuffer processed (virtual double buffer) - unsigned int sizeInFrames; // Total buffer size in frames - unsigned int frameCursorPos; // Frame cursor position - unsigned int totalFramesProcessed; // Total frames processed in this buffer (required for play timing) - - unsigned char *data; // Data buffer, on music stream keeps filling - - rAudioBuffer *next; // Next audio buffer on the list - rAudioBuffer *prev; // Previous audio buffer on the list -}; - -#define AudioBuffer rAudioBuffer // HACK: To avoid CoreAudio (macOS) symbol collision - -// Audio data context -typedef struct AudioData { - struct { - ma_context context; // miniaudio context data - ma_device device; // miniaudio device - ma_mutex lock; // miniaudio mutex lock - bool isReady; // Check if audio device is ready - } System; - struct { - AudioBuffer *first; // Pointer to first AudioBuffer in the list - AudioBuffer *last; // Pointer to last AudioBuffer in the list - int defaultSize; // Default audio buffer size for audio streams - } Buffer; - struct { - unsigned int poolCounter; // AudioBuffer pointers pool counter - AudioBuffer *pool[MAX_AUDIO_BUFFER_POOL_CHANNELS]; // Multichannel AudioBuffer pointers pool - unsigned int channels[MAX_AUDIO_BUFFER_POOL_CHANNELS]; // AudioBuffer pool channels - } MultiChannel; -} AudioData; - -//---------------------------------------------------------------------------------- -// Global Variables Definition -//---------------------------------------------------------------------------------- -static AudioData AUDIO = { // Global AUDIO context - - // NOTE: Music buffer size is defined by number of samples, independent of sample size and channels number - // After some math, considering a sampleRate of 48000, a buffer refill rate of 1/60 seconds and a - // standard double-buffering system, a 4096 samples buffer has been chosen, it should be enough - // In case of music-stalls, just increase this number - .Buffer.defaultSize = 0 -}; - -//---------------------------------------------------------------------------------- -// Module specific Functions Declaration -//---------------------------------------------------------------------------------- -static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message); -static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount); -static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume); - -#if defined(SUPPORT_FILEFORMAT_WAV) -static Wave LoadWAV(const unsigned char *fileData, unsigned int fileSize); // Load WAV file -static int SaveWAV(Wave wave, const char *fileName); // Save wave data as WAV file -#endif -#if defined(SUPPORT_FILEFORMAT_OGG) -static Wave LoadOGG(const unsigned char *fileData, unsigned int fileSize); // Load OGG file -#endif -#if defined(SUPPORT_FILEFORMAT_FLAC) -static Wave LoadFLAC(const unsigned char *fileData, unsigned int fileSize); // Load FLAC file -#endif -#if defined(SUPPORT_FILEFORMAT_MP3) -static Wave LoadMP3(const unsigned char *fileData, unsigned int fileSize); // Load MP3 file -#endif - -#if defined(RAUDIO_STANDALONE) -static bool IsFileExtension(const char *fileName, const char *ext); // Check file extension -static unsigned char *LoadFileData(const char *fileName, unsigned int *bytesRead); // Load file data as byte array (read) -static bool SaveFileData(const char *fileName, void *data, unsigned int bytesToWrite); // Save data to file from byte array (write) -static bool SaveFileText(const char *fileName, char *text); // Save text data to file (write), string must be '\0' terminated -#endif - -//---------------------------------------------------------------------------------- -// AudioBuffer management functions declaration -// NOTE: Those functions are not exposed by raylib... for the moment -//---------------------------------------------------------------------------------- -AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage); -void UnloadAudioBuffer(AudioBuffer *buffer); - -bool IsAudioBufferPlaying(AudioBuffer *buffer); -void PlayAudioBuffer(AudioBuffer *buffer); -void StopAudioBuffer(AudioBuffer *buffer); -void PauseAudioBuffer(AudioBuffer *buffer); -void ResumeAudioBuffer(AudioBuffer *buffer); -void SetAudioBufferVolume(AudioBuffer *buffer, float volume); -void SetAudioBufferPitch(AudioBuffer *buffer, float pitch); -void TrackAudioBuffer(AudioBuffer *buffer); -void UntrackAudioBuffer(AudioBuffer *buffer); -int GetAudioStreamBufferSizeDefault(); - -//---------------------------------------------------------------------------------- -// Module Functions Definition - Audio Device initialization and Closing -//---------------------------------------------------------------------------------- -// Initialize audio device -void InitAudioDevice(void) -{ - // TODO: Load AUDIO context memory dynamically? - - // Init audio context - ma_context_config ctxConfig = ma_context_config_init(); - ctxConfig.logCallback = OnLog; - - ma_result result = ma_context_init(NULL, 0, &ctxConfig, &AUDIO.System.context); - if (result != MA_SUCCESS) - { - TRACELOG(LOG_WARNING, "AUDIO: Failed to initialize context"); - return; - } - - // Init audio device - // NOTE: Using the default device. Format is floating point because it simplifies mixing. - ma_device_config config = ma_device_config_init(ma_device_type_playback); - config.playback.pDeviceID = NULL; // NULL for the default playback AUDIO.System.device. - config.playback.format = AUDIO_DEVICE_FORMAT; - config.playback.channels = AUDIO_DEVICE_CHANNELS; - config.capture.pDeviceID = NULL; // NULL for the default capture AUDIO.System.device. - config.capture.format = ma_format_s16; - config.capture.channels = 1; - config.sampleRate = AUDIO_DEVICE_SAMPLE_RATE; - config.dataCallback = OnSendAudioDataToDevice; - config.pUserData = NULL; - - result = ma_device_init(&AUDIO.System.context, &config, &AUDIO.System.device); - if (result != MA_SUCCESS) - { - TRACELOG(LOG_WARNING, "AUDIO: Failed to initialize playback device"); - ma_context_uninit(&AUDIO.System.context); - return; - } - - // Keep the device running the whole time. May want to consider doing something a bit smarter and only have the device running - // while there's at least one sound being played. - result = ma_device_start(&AUDIO.System.device); - if (result != MA_SUCCESS) - { - TRACELOG(LOG_WARNING, "AUDIO: Failed to start playback device"); - ma_device_uninit(&AUDIO.System.device); - ma_context_uninit(&AUDIO.System.context); - return; - } - - // Mixing happens on a seperate thread which means we need to synchronize. I'm using a mutex here to make things simple, but may - // want to look at something a bit smarter later on to keep everything real-time, if that's necessary. - if (ma_mutex_init(&AUDIO.System.lock) != MA_SUCCESS) - { - TRACELOG(LOG_WARNING, "AUDIO: Failed to create mutex for mixing"); - ma_device_uninit(&AUDIO.System.device); - ma_context_uninit(&AUDIO.System.context); - return; - } - - // Init dummy audio buffers pool for multichannel sound playing - for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) - { - // WARNING: An empty audioBuffer is created (data = 0) - // AudioBuffer data just points to loaded sound data - AUDIO.MultiChannel.pool[i] = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, 0, AUDIO_BUFFER_USAGE_STATIC); - } - - TRACELOG(LOG_INFO, "AUDIO: Device initialized successfully"); - TRACELOG(LOG_INFO, " > Backend: miniaudio / %s", ma_get_backend_name(AUDIO.System.context.backend)); - TRACELOG(LOG_INFO, " > Format: %s -> %s", ma_get_format_name(AUDIO.System.device.playback.format), ma_get_format_name(AUDIO.System.device.playback.internalFormat)); - TRACELOG(LOG_INFO, " > Channels: %d -> %d", AUDIO.System.device.playback.channels, AUDIO.System.device.playback.internalChannels); - TRACELOG(LOG_INFO, " > Sample rate: %d -> %d", AUDIO.System.device.sampleRate, AUDIO.System.device.playback.internalSampleRate); - TRACELOG(LOG_INFO, " > Periods size: %d", AUDIO.System.device.playback.internalPeriodSizeInFrames*AUDIO.System.device.playback.internalPeriods); - - AUDIO.System.isReady = true; -} - -// Close the audio device for all contexts -void CloseAudioDevice(void) -{ - if (AUDIO.System.isReady) - { - // Unload dummy audio buffers pool - // WARNING: They can be pointing to already unloaded data - for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) - { - //UnloadAudioBuffer(AUDIO.MultiChannel.pool[i]); - if (AUDIO.MultiChannel.pool[i] != NULL) - { - ma_data_converter_uninit(&AUDIO.MultiChannel.pool[i]->converter); - UntrackAudioBuffer(AUDIO.MultiChannel.pool[i]); - //RL_FREE(buffer->data); // Already unloaded by UnloadSound() - RL_FREE(AUDIO.MultiChannel.pool[i]); - } - } - - ma_mutex_uninit(&AUDIO.System.lock); - ma_device_uninit(&AUDIO.System.device); - ma_context_uninit(&AUDIO.System.context); - - AUDIO.System.isReady = false; - - TRACELOG(LOG_INFO, "AUDIO: Device closed successfully"); - } - else TRACELOG(LOG_WARNING, "AUDIO: Device could not be closed, not currently initialized"); -} - -// Check if device has been initialized successfully -bool IsAudioDeviceReady(void) -{ - return AUDIO.System.isReady; -} - -// Set master volume (listener) -void SetMasterVolume(float volume) -{ - ma_device_set_master_volume(&AUDIO.System.device, volume); -} - -//---------------------------------------------------------------------------------- -// Module Functions Definition - Audio Buffer management -//---------------------------------------------------------------------------------- - -// Initialize a new audio buffer (filled with silence) -AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage) -{ - AudioBuffer *audioBuffer = (AudioBuffer *)RL_CALLOC(1, sizeof(AudioBuffer)); - - if (audioBuffer == NULL) - { - TRACELOG(LOG_WARNING, "AUDIO: Failed to allocate memory for buffer"); - return NULL; - } - - if (sizeInFrames > 0) audioBuffer->data = RL_CALLOC(sizeInFrames*channels*ma_get_bytes_per_sample(format), 1); - - // Audio data runs through a format converter - ma_data_converter_config converterConfig = ma_data_converter_config_init(format, AUDIO_DEVICE_FORMAT, channels, AUDIO_DEVICE_CHANNELS, sampleRate, AUDIO.System.device.sampleRate); - converterConfig.resampling.allowDynamicSampleRate = true; // Required for pitch shifting - - ma_result result = ma_data_converter_init(&converterConfig, &audioBuffer->converter); - - if (result != MA_SUCCESS) - { - TRACELOG(LOG_WARNING, "AUDIO: Failed to create data conversion pipeline"); - RL_FREE(audioBuffer); - return NULL; - } - - // Init audio buffer values - audioBuffer->volume = 1.0f; - audioBuffer->pitch = 1.0f; - audioBuffer->playing = false; - audioBuffer->paused = false; - audioBuffer->looping = false; - audioBuffer->usage = usage; - audioBuffer->frameCursorPos = 0; - audioBuffer->sizeInFrames = sizeInFrames; - - // Buffers should be marked as processed by default so that a call to - // UpdateAudioStream() immediately after initialization works correctly - audioBuffer->isSubBufferProcessed[0] = true; - audioBuffer->isSubBufferProcessed[1] = true; - - // Track audio buffer to linked list next position - TrackAudioBuffer(audioBuffer); - - return audioBuffer; -} - -// Delete an audio buffer -void UnloadAudioBuffer(AudioBuffer *buffer) -{ - if (buffer != NULL) - { - ma_data_converter_uninit(&buffer->converter); - UntrackAudioBuffer(buffer); - RL_FREE(buffer->data); - RL_FREE(buffer); - } -} - -// Check if an audio buffer is playing -bool IsAudioBufferPlaying(AudioBuffer *buffer) -{ - bool result = false; - - if (buffer != NULL) result = (buffer->playing && !buffer->paused); - - return result; -} - -// Play an audio buffer -// NOTE: Buffer is restarted to the start. -// Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position should be maintained. -void PlayAudioBuffer(AudioBuffer *buffer) -{ - if (buffer != NULL) - { - buffer->playing = true; - buffer->paused = false; - buffer->frameCursorPos = 0; - } -} - -// Stop an audio buffer -void StopAudioBuffer(AudioBuffer *buffer) -{ - if (buffer != NULL) - { - if (IsAudioBufferPlaying(buffer)) - { - buffer->playing = false; - buffer->paused = false; - buffer->frameCursorPos = 0; - buffer->totalFramesProcessed = 0; - buffer->isSubBufferProcessed[0] = true; - buffer->isSubBufferProcessed[1] = true; - } - } -} - -// Pause an audio buffer -void PauseAudioBuffer(AudioBuffer *buffer) -{ - if (buffer != NULL) buffer->paused = true; -} - -// Resume an audio buffer -void ResumeAudioBuffer(AudioBuffer *buffer) -{ - if (buffer != NULL) buffer->paused = false; -} - -// Set volume for an audio buffer -void SetAudioBufferVolume(AudioBuffer *buffer, float volume) -{ - if (buffer != NULL) buffer->volume = volume; -} - -// Set pitch for an audio buffer -void SetAudioBufferPitch(AudioBuffer *buffer, float pitch) -{ - if ((buffer != NULL) && (pitch > 0.0f)) - { - // Pitching is just an adjustment of the sample rate. - // Note that this changes the duration of the sound: - // - higher pitches will make the sound faster - // - lower pitches make it slower - ma_uint32 outputSampleRate = (ma_uint32)((float)buffer->converter.config.sampleRateOut/pitch); - ma_data_converter_set_rate(&buffer->converter, buffer->converter.config.sampleRateIn, outputSampleRate); - - buffer->pitch = pitch; - } -} - -// Track audio buffer to linked list next position -void TrackAudioBuffer(AudioBuffer *buffer) -{ - ma_mutex_lock(&AUDIO.System.lock); - { - if (AUDIO.Buffer.first == NULL) AUDIO.Buffer.first = buffer; - else - { - AUDIO.Buffer.last->next = buffer; - buffer->prev = AUDIO.Buffer.last; - } - - AUDIO.Buffer.last = buffer; - } - ma_mutex_unlock(&AUDIO.System.lock); -} - -// Untrack audio buffer from linked list -void UntrackAudioBuffer(AudioBuffer *buffer) -{ - ma_mutex_lock(&AUDIO.System.lock); - { - if (buffer->prev == NULL) AUDIO.Buffer.first = buffer->next; - else buffer->prev->next = buffer->next; - - if (buffer->next == NULL) AUDIO.Buffer.last = buffer->prev; - else buffer->next->prev = buffer->prev; - - buffer->prev = NULL; - buffer->next = NULL; - } - ma_mutex_unlock(&AUDIO.System.lock); -} - -//---------------------------------------------------------------------------------- -// Module Functions Definition - Sounds loading and playing (.WAV) -//---------------------------------------------------------------------------------- - -// Load wave data from file -Wave LoadWave(const char *fileName) -{ - Wave wave = { 0 }; - - // Loading file to memory - unsigned int fileSize = 0; - unsigned char *fileData = LoadFileData(fileName, &fileSize); - - if (fileData != NULL) - { - // Loading wave from memory data - wave = LoadWaveFromMemory(GetFileExtension(fileName), fileData, fileSize); - - RL_FREE(fileData); - } - - return wave; -} - -// Load wave from memory buffer, fileType refers to extension: i.e. ".wav" -Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int dataSize) -{ - Wave wave = { 0 }; - - char fileExtLower[16] = { 0 }; - strcpy(fileExtLower, TextToLower(fileType)); - - if (false) { } -#if defined(SUPPORT_FILEFORMAT_WAV) - else if (TextIsEqual(fileExtLower, ".wav")) wave = LoadWAV(fileData, dataSize); -#endif -#if defined(SUPPORT_FILEFORMAT_OGG) - else if (TextIsEqual(fileExtLower, ".ogg")) wave = LoadOGG(fileData, dataSize); -#endif -#if defined(SUPPORT_FILEFORMAT_FLAC) - else if (TextIsEqual(fileExtLower, ".flac")) wave = LoadFLAC(fileData, dataSize); -#endif -#if defined(SUPPORT_FILEFORMAT_MP3) - else if (TextIsEqual(fileExtLower, ".mp3")) wave = LoadMP3(fileData, dataSize); -#endif - else TRACELOG(LOG_WARNING, "WAVE: File format not supported"); - - return wave; -} - -// Load sound from file -// NOTE: The entire file is loaded to memory to be played (no-streaming) -Sound LoadSound(const char *fileName) -{ - Wave wave = LoadWave(fileName); - - Sound sound = LoadSoundFromWave(wave); - - UnloadWave(wave); // Sound is loaded, we can unload wave - - return sound; -} - -// Load sound from wave data -// NOTE: Wave data must be unallocated manually -Sound LoadSoundFromWave(Wave wave) -{ - Sound sound = { 0 }; - - if (wave.data != NULL) - { - // When using miniaudio we need to do our own mixing. - // To simplify this we need convert the format of each sound to be consistent with - // the format used to open the playback AUDIO.System.device. We can do this two ways: - // - // 1) Convert the whole sound in one go at load time (here). - // 2) Convert the audio data in chunks at mixing time. - // - // First option has been selected, format conversion is done on the loading stage. - // The downside is that it uses more memory if the original sound is u8 or s16. - ma_format formatIn = ((wave.sampleSize == 8)? ma_format_u8 : ((wave.sampleSize == 16)? ma_format_s16 : ma_format_f32)); - ma_uint32 frameCountIn = wave.sampleCount/wave.channels; - - ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, NULL, frameCountIn, formatIn, wave.channels, wave.sampleRate); - if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed to get frame count for format conversion"); - - AudioBuffer *audioBuffer = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, frameCount, AUDIO_BUFFER_USAGE_STATIC); - if (audioBuffer == NULL) - { - TRACELOG(LOG_WARNING, "SOUND: Failed to create buffer"); - return sound; // early return to avoid dereferencing the audioBuffer null pointer - } - - frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, frameCount, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, wave.data, frameCountIn, formatIn, wave.channels, wave.sampleRate); - if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed format conversion"); - - sound.sampleCount = frameCount*AUDIO_DEVICE_CHANNELS; - sound.stream.sampleRate = AUDIO.System.device.sampleRate; - sound.stream.sampleSize = 32; - sound.stream.channels = AUDIO_DEVICE_CHANNELS; - sound.stream.buffer = audioBuffer; - } - - return sound; -} - -// Unload wave data -void UnloadWave(Wave wave) -{ - if (wave.data != NULL) RL_FREE(wave.data); - - TRACELOG(LOG_INFO, "WAVE: Unloaded wave data from RAM"); -} - -// Unload sound -void UnloadSound(Sound sound) -{ - UnloadAudioBuffer(sound.stream.buffer); - - TRACELOG(LOG_INFO, "WAVE: Unloaded sound data from RAM"); -} - -// Update sound buffer with new data -void UpdateSound(Sound sound, const void *data, int samplesCount) -{ - if (sound.stream.buffer != NULL) - { - StopAudioBuffer(sound.stream.buffer); - - // TODO: May want to lock/unlock this since this data buffer is read at mixing time - memcpy(sound.stream.buffer->data, data, samplesCount*ma_get_bytes_per_frame(sound.stream.buffer->converter.config.formatIn, sound.stream.buffer->converter.config.channelsIn)); - } -} - -// Export wave data to file -bool ExportWave(Wave wave, const char *fileName) -{ - bool success = false; - - if (false) { } -#if defined(SUPPORT_FILEFORMAT_WAV) - else if (IsFileExtension(fileName, ".wav")) success = SaveWAV(wave, fileName); -#endif - else if (IsFileExtension(fileName, ".raw")) - { - // Export raw sample data (without header) - // NOTE: It's up to the user to track wave parameters - success = SaveFileData(fileName, wave.data, wave.sampleCount*wave.sampleSize/8); - } - - if (success) TRACELOG(LOG_INFO, "FILEIO: [%s] Wave data exported successfully", fileName); - else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to export wave data", fileName); - - return success; -} - -// Export wave sample data to code (.h) -bool ExportWaveAsCode(Wave wave, const char *fileName) -{ - bool success = false; - -#ifndef TEXT_BYTES_PER_LINE - #define TEXT_BYTES_PER_LINE 20 -#endif - - int waveDataSize = wave.sampleCount*wave.channels*wave.sampleSize/8; - - // NOTE: Text data buffer size is estimated considering wave data size in bytes - // and requiring 6 char bytes for every byte: "0x00, " - char *txtData = (char *)RL_CALLOC(6*waveDataSize + 2000, sizeof(char)); - - int bytesCount = 0; - bytesCount += sprintf(txtData + bytesCount, "\n//////////////////////////////////////////////////////////////////////////////////\n"); - bytesCount += sprintf(txtData + bytesCount, "// //\n"); - bytesCount += sprintf(txtData + bytesCount, "// WaveAsCode exporter v1.0 - Wave data exported as an array of bytes //\n"); - bytesCount += sprintf(txtData + bytesCount, "// //\n"); - bytesCount += sprintf(txtData + bytesCount, "// more info and bugs-report: github.com/raysan5/raylib //\n"); - bytesCount += sprintf(txtData + bytesCount, "// feedback and support: ray[at]raylib.com //\n"); - bytesCount += sprintf(txtData + bytesCount, "// //\n"); - bytesCount += sprintf(txtData + bytesCount, "// Copyright (c) 2018 Ramon Santamaria (@raysan5) //\n"); - bytesCount += sprintf(txtData + bytesCount, "// //\n"); - bytesCount += sprintf(txtData + bytesCount, "//////////////////////////////////////////////////////////////////////////////////\n\n"); - - char varFileName[256] = { 0 }; -#if !defined(RAUDIO_STANDALONE) - // Get file name from path and convert variable name to uppercase - strcpy(varFileName, GetFileNameWithoutExt(fileName)); - for (int i = 0; varFileName[i] != '\0'; i++) if (varFileName[i] >= 'a' && varFileName[i] <= 'z') { varFileName[i] = varFileName[i] - 32; } -#else - strcpy(varFileName, fileName); -#endif - - bytesCount += sprintf(txtData + bytesCount, "// Wave data information\n"); - bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_COUNT %u\n", varFileName, wave.sampleCount); - bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_RATE %u\n", varFileName, wave.sampleRate); - bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_SIZE %u\n", varFileName, wave.sampleSize); - bytesCount += sprintf(txtData + bytesCount, "#define %s_CHANNELS %u\n\n", varFileName, wave.channels); - - // Write byte data as hexadecimal text - bytesCount += sprintf(txtData + bytesCount, "static unsigned char %s_DATA[%i] = { ", varFileName, waveDataSize); - for (int i = 0; i < waveDataSize - 1; i++) bytesCount += sprintf(txtData + bytesCount, ((i%TEXT_BYTES_PER_LINE == 0)? "0x%x,\n" : "0x%x, "), ((unsigned char *)wave.data)[i]); - bytesCount += sprintf(txtData + bytesCount, "0x%x };\n", ((unsigned char *)wave.data)[waveDataSize - 1]); - - // NOTE: Text data length exported is determined by '\0' (NULL) character - success = SaveFileText(fileName, txtData); - - RL_FREE(txtData); - - return success; -} - -// Play a sound -void PlaySound(Sound sound) -{ - PlayAudioBuffer(sound.stream.buffer); -} - -// Play a sound in the multichannel buffer pool -void PlaySoundMulti(Sound sound) -{ - int index = -1; - unsigned int oldAge = 0; - int oldIndex = -1; - - // find the first non playing pool entry - for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) - { - if (AUDIO.MultiChannel.channels[i] > oldAge) - { - oldAge = AUDIO.MultiChannel.channels[i]; - oldIndex = i; - } - - if (!IsAudioBufferPlaying(AUDIO.MultiChannel.pool[i])) - { - index = i; - break; - } - } - - // If no none playing pool members can be index choose the oldest - if (index == -1) - { - TRACELOG(LOG_WARNING, "SOUND: Buffer pool is already full, count: %i", AUDIO.MultiChannel.poolCounter); - - if (oldIndex == -1) - { - // Shouldn't be able to get here... but just in case something odd happens! - TRACELOG(LOG_WARNING, "SOUND: Buffer pool could not determine oldest buffer not playing sound"); - return; - } - - index = oldIndex; - - // Just in case... - StopAudioBuffer(AUDIO.MultiChannel.pool[index]); - } - - // Experimentally mutex lock doesn't seem to be needed this makes sense - // as pool[index] isn't playing and the only stuff we're copying - // shouldn't be changing... - - AUDIO.MultiChannel.channels[index] = AUDIO.MultiChannel.poolCounter; - AUDIO.MultiChannel.poolCounter++; - - AUDIO.MultiChannel.pool[index]->volume = sound.stream.buffer->volume; - AUDIO.MultiChannel.pool[index]->pitch = sound.stream.buffer->pitch; - AUDIO.MultiChannel.pool[index]->looping = sound.stream.buffer->looping; - AUDIO.MultiChannel.pool[index]->usage = sound.stream.buffer->usage; - AUDIO.MultiChannel.pool[index]->isSubBufferProcessed[0] = false; - AUDIO.MultiChannel.pool[index]->isSubBufferProcessed[1] = false; - AUDIO.MultiChannel.pool[index]->sizeInFrames = sound.stream.buffer->sizeInFrames; - AUDIO.MultiChannel.pool[index]->data = sound.stream.buffer->data; - - PlayAudioBuffer(AUDIO.MultiChannel.pool[index]); -} - -// Stop any sound played with PlaySoundMulti() -void StopSoundMulti(void) -{ - for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) StopAudioBuffer(AUDIO.MultiChannel.pool[i]); -} - -// Get number of sounds playing in the multichannel buffer pool -int GetSoundsPlaying(void) -{ - int counter = 0; - - for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) - { - if (IsAudioBufferPlaying(AUDIO.MultiChannel.pool[i])) counter++; - } - - return counter; -} - -// Pause a sound -void PauseSound(Sound sound) -{ - PauseAudioBuffer(sound.stream.buffer); -} - -// Resume a paused sound -void ResumeSound(Sound sound) -{ - ResumeAudioBuffer(sound.stream.buffer); -} - -// Stop reproducing a sound -void StopSound(Sound sound) -{ - StopAudioBuffer(sound.stream.buffer); -} - -// Check if a sound is playing -bool IsSoundPlaying(Sound sound) -{ - return IsAudioBufferPlaying(sound.stream.buffer); -} - -// Set volume for a sound -void SetSoundVolume(Sound sound, float volume) -{ - SetAudioBufferVolume(sound.stream.buffer, volume); -} - -// Set pitch for a sound -void SetSoundPitch(Sound sound, float pitch) -{ - SetAudioBufferPitch(sound.stream.buffer, pitch); -} - -// Convert wave data to desired format -void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels) -{ - ma_format formatIn = ((wave->sampleSize == 8)? ma_format_u8 : ((wave->sampleSize == 16)? ma_format_s16 : ma_format_f32)); - ma_format formatOut = ((sampleSize == 8)? ma_format_u8 : ((sampleSize == 16)? ma_format_s16 : ma_format_f32)); - - ma_uint32 frameCountIn = wave->sampleCount/wave->channels; - - ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, formatOut, channels, sampleRate, NULL, frameCountIn, formatIn, wave->channels, wave->sampleRate); - if (frameCount == 0) - { - TRACELOG(LOG_WARNING, "WAVE: Failed to get frame count for format conversion"); - return; - } - - void *data = RL_MALLOC(frameCount*channels*(sampleSize/8)); - - frameCount = (ma_uint32)ma_convert_frames(data, frameCount, formatOut, channels, sampleRate, wave->data, frameCountIn, formatIn, wave->channels, wave->sampleRate); - if (frameCount == 0) - { - TRACELOG(LOG_WARNING, "WAVE: Failed format conversion"); - return; - } - - wave->sampleCount = frameCount*channels; - wave->sampleSize = sampleSize; - wave->sampleRate = sampleRate; - wave->channels = channels; - RL_FREE(wave->data); - wave->data = data; -} - -// Copy a wave to a new wave -Wave WaveCopy(Wave wave) -{ - Wave newWave = { 0 }; - - newWave.data = RL_MALLOC(wave.sampleCount*wave.sampleSize/8); - - if (newWave.data != NULL) - { - // NOTE: Size must be provided in bytes - memcpy(newWave.data, wave.data, wave.sampleCount*wave.sampleSize/8); - - newWave.sampleCount = wave.sampleCount; - newWave.sampleRate = wave.sampleRate; - newWave.sampleSize = wave.sampleSize; - newWave.channels = wave.channels; - } - - return newWave; -} - -// Crop a wave to defined samples range -// NOTE: Security check in case of out-of-range -void WaveCrop(Wave *wave, int initSample, int finalSample) -{ - if ((initSample >= 0) && (initSample < finalSample) && - (finalSample > 0) && ((unsigned int)finalSample < wave->sampleCount)) - { - int sampleCount = finalSample - initSample; - - void *data = RL_MALLOC(sampleCount*wave->sampleSize/8); - - memcpy(data, (unsigned char *)wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->sampleSize/8); - - RL_FREE(wave->data); - wave->data = data; - } - else TRACELOG(LOG_WARNING, "WAVE: Crop range out of bounds"); -} - -// Load samples data from wave as a floats array -// NOTE 1: Returned sample values are normalized to range [-1..1] -// NOTE 2: Sample data allocated should be freed with UnloadWaveSamples() -float *LoadWaveSamples(Wave wave) -{ - float *samples = (float *)RL_MALLOC(wave.sampleCount*sizeof(float)); - - // NOTE: sampleCount is the total number of interlaced samples (including channels) - - for (unsigned int i = 0; i < wave.sampleCount; i++) - { - if (wave.sampleSize == 8) samples[i] = (float)(((unsigned char *)wave.data)[i] - 127)/256.0f; - else if (wave.sampleSize == 16) samples[i] = (float)(((short *)wave.data)[i])/32767.0f; - else if (wave.sampleSize == 32) samples[i] = ((float *)wave.data)[i]; - } - - return samples; -} - -// Unload samples data loaded with LoadWaveSamples() -void UnloadWaveSamples(float *samples) -{ - RL_FREE(samples); -} - -//---------------------------------------------------------------------------------- -// Module Functions Definition - Music loading and stream playing (.OGG) -//---------------------------------------------------------------------------------- - -// Load music stream from file -Music LoadMusicStream(const char *fileName) -{ - Music music = { 0 }; - bool musicLoaded = false; - - if (false) { } -#if defined(SUPPORT_FILEFORMAT_WAV) - else if (IsFileExtension(fileName, ".wav")) - { - drwav *ctxWav = RL_CALLOC(1, sizeof(drwav)); - bool success = drwav_init_file(ctxWav, fileName, NULL); - - music.ctxType = MUSIC_AUDIO_WAV; - music.ctxData = ctxWav; - - if (success) - { - int sampleSize = ctxWav->bitsPerSample; - if (ctxWav->bitsPerSample == 24) sampleSize = 16; // Forcing conversion to s16 on UpdateMusicStream() - - music.stream = InitAudioStream(ctxWav->sampleRate, sampleSize, ctxWav->channels); - music.sampleCount = (unsigned int)ctxWav->totalPCMFrameCount*ctxWav->channels; - music.looping = true; // Looping enabled by default - musicLoaded = true; - } - } -#endif -#if defined(SUPPORT_FILEFORMAT_OGG) - else if (IsFileExtension(fileName, ".ogg")) - { - // Open ogg audio stream - music.ctxType = MUSIC_AUDIO_OGG; - music.ctxData = stb_vorbis_open_filename(fileName, NULL, NULL); - - if (music.ctxData != NULL) - { - stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music.ctxData); // Get Ogg file info - - // OGG bit rate defaults to 16 bit, it's enough for compressed format - music.stream = InitAudioStream(info.sample_rate, 16, info.channels); - - // WARNING: It seems this function returns length in frames, not samples, so we multiply by channels - music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels; - music.looping = true; // Looping enabled by default - musicLoaded = true; - } - } -#endif -#if defined(SUPPORT_FILEFORMAT_FLAC) - else if (IsFileExtension(fileName, ".flac")) - { - music.ctxType = MUSIC_AUDIO_FLAC; - music.ctxData = drflac_open_file(fileName, NULL); - - if (music.ctxData != NULL) - { - drflac *ctxFlac = (drflac *)music.ctxData; - - music.stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels); - music.sampleCount = (unsigned int)ctxFlac->totalPCMFrameCount*ctxFlac->channels; - music.looping = true; // Looping enabled by default - musicLoaded = true; - } - } -#endif -#if defined(SUPPORT_FILEFORMAT_MP3) - else if (IsFileExtension(fileName, ".mp3")) - { - drmp3 *ctxMp3 = RL_CALLOC(1, sizeof(drmp3)); - int result = drmp3_init_file(ctxMp3, fileName, NULL); - - music.ctxType = MUSIC_AUDIO_MP3; - music.ctxData = ctxMp3; - - if (result > 0) - { - music.stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels); - music.sampleCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels; - music.looping = true; // Looping enabled by default - musicLoaded = true; - } - } -#endif -#if defined(SUPPORT_FILEFORMAT_XM) - else if (IsFileExtension(fileName, ".xm")) - { - jar_xm_context_t *ctxXm = NULL; - int result = jar_xm_create_context_from_file(&ctxXm, AUDIO.System.device.sampleRate, fileName); - - music.ctxType = MUSIC_MODULE_XM; - music.ctxData = ctxXm; - - if (result == 0) // XM AUDIO.System.context created successfully - { - jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops - - unsigned int bits = 32; - if (AUDIO_DEVICE_FORMAT == ma_format_s16) - bits = 16; - else if (AUDIO_DEVICE_FORMAT == ma_format_u8) - bits = 8; - - // NOTE: Only stereo is supported for XM - music.stream = InitAudioStream(AUDIO.System.device.sampleRate, bits, AUDIO_DEVICE_CHANNELS); - music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)*2; // 2 channels - music.looping = true; // Looping enabled by default - jar_xm_reset(ctxXm); // make sure we start at the beginning of the song - musicLoaded = true; - } - } -#endif -#if defined(SUPPORT_FILEFORMAT_MOD) - else if (IsFileExtension(fileName, ".mod")) - { - jar_mod_context_t *ctxMod = RL_CALLOC(1, sizeof(jar_mod_context_t)); - jar_mod_init(ctxMod); - int result = jar_mod_load_file(ctxMod, fileName); - - music.ctxType = MUSIC_MODULE_MOD; - music.ctxData = ctxMod; - - if (result > 0) - { - // NOTE: Only stereo is supported for MOD - music.stream = InitAudioStream(AUDIO.System.device.sampleRate, 16, AUDIO_DEVICE_CHANNELS); - music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2; // 2 channels - music.looping = true; // Looping enabled by default - musicLoaded = true; - } - } -#endif - else TRACELOG(LOG_WARNING, "STREAM: [%s] Fileformat not supported", fileName); - - if (!musicLoaded) - { - if (false) { } - #if defined(SUPPORT_FILEFORMAT_WAV) - else if (music.ctxType == MUSIC_AUDIO_WAV) drwav_uninit((drwav *)music.ctxData); - #endif - #if defined(SUPPORT_FILEFORMAT_OGG) - else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData); - #endif - #if defined(SUPPORT_FILEFORMAT_FLAC) - else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL); - #endif - #if defined(SUPPORT_FILEFORMAT_MP3) - else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); } - #endif - #if defined(SUPPORT_FILEFORMAT_XM) - else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData); - #endif - #if defined(SUPPORT_FILEFORMAT_MOD) - else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); } - #endif - - music.ctxData = NULL; - TRACELOG(LOG_WARNING, "FILEIO: [%s] Music file could not be opened", fileName); - } - else - { - // Show some music stream info - TRACELOG(LOG_INFO, "FILEIO: [%s] Music file successfully loaded:", fileName); - TRACELOG(LOG_INFO, " > Total samples: %i", music.sampleCount); - TRACELOG(LOG_INFO, " > Sample rate: %i Hz", music.stream.sampleRate); - TRACELOG(LOG_INFO, " > Sample size: %i bits", music.stream.sampleSize); - TRACELOG(LOG_INFO, " > Channels: %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi"); - } - - return music; -} - -// extension including period ".mod" -Music LoadMusicStreamFromMemory(const char *fileType, unsigned char* data, int dataSize) -{ - Music music = { 0 }; - bool musicLoaded = false; - - char fileExtLower[16] = { 0 }; - strcpy(fileExtLower, TextToLower(fileType)); - - if (false) { } -#if defined(SUPPORT_FILEFORMAT_WAV) - else if (TextIsEqual(fileExtLower, ".wav")) - { - drwav *ctxWav = RL_CALLOC(1, sizeof(drwav)); - - bool success = drwav_init_memory(ctxWav, (const void*)data, dataSize, NULL); - - music.ctxType = MUSIC_AUDIO_WAV; - music.ctxData = ctxWav; - - if (success) - { - int sampleSize = ctxWav->bitsPerSample; - if (ctxWav->bitsPerSample == 24) sampleSize = 16; // Forcing conversion to s16 on UpdateMusicStream() - - music.stream = InitAudioStream(ctxWav->sampleRate, sampleSize, ctxWav->channels); - music.sampleCount = (unsigned int)ctxWav->totalPCMFrameCount*ctxWav->channels; - music.looping = true; // Looping enabled by default - musicLoaded = true; - } - } -#endif -#if defined(SUPPORT_FILEFORMAT_FLAC) - else if (TextIsEqual(fileExtLower, ".flac")) - { - music.ctxType = MUSIC_AUDIO_FLAC; - music.ctxData = drflac_open_memory((const void*)data, dataSize, NULL); - - if (music.ctxData != NULL) - { - drflac *ctxFlac = (drflac *)music.ctxData; - - music.stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels); - music.sampleCount = (unsigned int)ctxFlac->totalPCMFrameCount*ctxFlac->channels; - music.looping = true; // Looping enabled by default - musicLoaded = true; - } - } -#endif -#if defined(SUPPORT_FILEFORMAT_MP3) - else if (TextIsEqual(fileExtLower, ".mp3")) - { - drmp3 *ctxMp3 = RL_CALLOC(1, sizeof(drmp3)); - int success = drmp3_init_memory(ctxMp3, (const void*)data, dataSize, NULL); - - music.ctxType = MUSIC_AUDIO_MP3; - music.ctxData = ctxMp3; - - if (success) - { - music.stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels); - music.sampleCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels; - music.looping = true; // Looping enabled by default - musicLoaded = true; - } - } -#endif -#if defined(SUPPORT_FILEFORMAT_OGG) - else if (TextIsEqual(fileExtLower, ".ogg")) - { - // Open ogg audio stream - music.ctxType = MUSIC_AUDIO_OGG; - //music.ctxData = stb_vorbis_open_filename(fileName, NULL, NULL); - music.ctxData = stb_vorbis_open_memory((const unsigned char*)data, dataSize, NULL, NULL); - - if (music.ctxData != NULL) - { - stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music.ctxData); // Get Ogg file info - - // OGG bit rate defaults to 16 bit, it's enough for compressed format - music.stream = InitAudioStream(info.sample_rate, 16, info.channels); - - // WARNING: It seems this function returns length in frames, not samples, so we multiply by channels - music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels; - music.looping = true; // Looping enabled by default - musicLoaded = true; - } - } -#endif -#if defined(SUPPORT_FILEFORMAT_XM) - else if (TextIsEqual(fileExtLower, ".xm")) - { - jar_xm_context_t *ctxXm = NULL; - int result = jar_xm_create_context_safe(&ctxXm, (const char*)data, dataSize, AUDIO.System.device.sampleRate); - if (result == 0) // XM AUDIO.System.context created successfully - { - music.ctxType = MUSIC_MODULE_XM; - jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops - - unsigned int bits = 32; - if (AUDIO_DEVICE_FORMAT == ma_format_s16) - bits = 16; - else if (AUDIO_DEVICE_FORMAT == ma_format_u8) - bits = 8; - - // NOTE: Only stereo is supported for XM - music.stream = InitAudioStream(AUDIO.System.device.sampleRate, bits, 2); - music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)*2; // 2 channels - music.looping = true; // Looping enabled by default - jar_xm_reset(ctxXm); // make sure we start at the beginning of the song - - music.ctxData = ctxXm; - musicLoaded = true; - } - } -#endif -#if defined(SUPPORT_FILEFORMAT_MOD) - else if (TextIsEqual(fileExtLower, ".mod")) - { - jar_mod_context_t *ctxMod = RL_MALLOC(sizeof(jar_mod_context_t)); - int result = 0; - - jar_mod_init(ctxMod); - - // copy data to allocated memory for default UnloadMusicStream - unsigned char *newData = RL_MALLOC(dataSize); - int it = dataSize/sizeof(unsigned char); - for (int i = 0; i < it; i++){ - newData[i] = data[i]; - } - - // Memory loaded version for jar_mod_load_file() - if (dataSize && dataSize < 32*1024*1024) - { - ctxMod->modfilesize = dataSize; - ctxMod->modfile = newData; - if (jar_mod_load(ctxMod, (void *)ctxMod->modfile, dataSize)) result = dataSize; - } - - if (result > 0) - { - music.ctxType = MUSIC_MODULE_MOD; - - // NOTE: Only stereo is supported for MOD - music.stream = InitAudioStream(AUDIO.System.device.sampleRate, 16, 2); - music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2; // 2 channels - music.looping = true; // Looping enabled by default - musicLoaded = true; - - music.ctxData = ctxMod; - musicLoaded = true; - } - } -#endif - else TRACELOG(LOG_WARNING, "STREAM: [%s] Fileformat not supported", fileType); - - if (!musicLoaded) - { - if (false) { } - #if defined(SUPPORT_FILEFORMAT_WAV) - else if (music.ctxType == MUSIC_AUDIO_WAV) drwav_uninit((drwav *)music.ctxData); - #endif - #if defined(SUPPORT_FILEFORMAT_FLAC) - else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL); - #endif - #if defined(SUPPORT_FILEFORMAT_MP3) - else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); } - #endif - #if defined(SUPPORT_FILEFORMAT_OGG) - else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData); - #endif - #if defined(SUPPORT_FILEFORMAT_XM) - else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData); - #endif - #if defined(SUPPORT_FILEFORMAT_MOD) - else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); } - #endif - - music.ctxData = NULL; - TRACELOG(LOG_WARNING, "FILEIO: [%s] Music memory could not be opened", fileType); - } - else - { - // Show some music stream info - TRACELOG(LOG_INFO, "FILEIO: [%s] Music memory successfully loaded:", fileType); - TRACELOG(LOG_INFO, " > Total samples: %i", music.sampleCount); - TRACELOG(LOG_INFO, " > Sample rate: %i Hz", music.stream.sampleRate); - TRACELOG(LOG_INFO, " > Sample size: %i bits", music.stream.sampleSize); - TRACELOG(LOG_INFO, " > Channels: %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi"); - } - - return music; -} - -// Unload music stream -void UnloadMusicStream(Music music) -{ - CloseAudioStream(music.stream); - - if (music.ctxData != NULL) - { - if (false) { } -#if defined(SUPPORT_FILEFORMAT_WAV) - else if (music.ctxType == MUSIC_AUDIO_WAV) drwav_uninit((drwav *)music.ctxData); -#endif -#if defined(SUPPORT_FILEFORMAT_OGG) - else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData); -#endif -#if defined(SUPPORT_FILEFORMAT_FLAC) - else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL); -#endif -#if defined(SUPPORT_FILEFORMAT_MP3) - else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); } -#endif -#if defined(SUPPORT_FILEFORMAT_XM) - else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData); -#endif -#if defined(SUPPORT_FILEFORMAT_MOD) - else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); } -#endif - } -} - -// Start music playing (open stream) -void PlayMusicStream(Music music) -{ - if (music.stream.buffer != NULL) - { - // For music streams, we need to make sure we maintain the frame cursor position - // This is a hack for this section of code in UpdateMusicStream() - // NOTE: In case window is minimized, music stream is stopped, just make sure to - // play again on window restore: if (IsMusicPlaying(music)) PlayMusicStream(music); - ma_uint32 frameCursorPos = music.stream.buffer->frameCursorPos; - PlayAudioStream(music.stream); // WARNING: This resets the cursor position. - music.stream.buffer->frameCursorPos = frameCursorPos; - } -} - -// Pause music playing -void PauseMusicStream(Music music) -{ - PauseAudioStream(music.stream); -} - -// Resume music playing -void ResumeMusicStream(Music music) -{ - ResumeAudioStream(music.stream); -} - -// Stop music playing (close stream) -void StopMusicStream(Music music) -{ - StopAudioStream(music.stream); - - switch (music.ctxType) - { -#if defined(SUPPORT_FILEFORMAT_WAV) - case MUSIC_AUDIO_WAV: drwav_seek_to_pcm_frame((drwav *)music.ctxData, 0); break; -#endif -#if defined(SUPPORT_FILEFORMAT_OGG) - case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break; -#endif -#if defined(SUPPORT_FILEFORMAT_FLAC) - case MUSIC_AUDIO_FLAC: drflac_seek_to_pcm_frame((drflac *)music.ctxData, 0); break; -#endif -#if defined(SUPPORT_FILEFORMAT_MP3) - case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, 0); break; -#endif -#if defined(SUPPORT_FILEFORMAT_XM) - case MUSIC_MODULE_XM: jar_xm_reset((jar_xm_context_t *)music.ctxData); break; -#endif -#if defined(SUPPORT_FILEFORMAT_MOD) - case MUSIC_MODULE_MOD: jar_mod_seek_start((jar_mod_context_t *)music.ctxData); break; -#endif - default: break; - } -} - -// Update (re-fill) music buffers if data already processed -void UpdateMusicStream(Music music) -{ - if (music.stream.buffer == NULL) - return; - - if (music.ctxType == MUSIC_MODULE_XM) - jar_xm_set_max_loop_count(music.ctxData, music.looping ? 0 : 1); - - bool streamEnding = false; - - unsigned int subBufferSizeInFrames = music.stream.buffer->sizeInFrames/2; - - // NOTE: Using dynamic allocation because it could require more than 16KB - void *pcm = RL_CALLOC(subBufferSizeInFrames*music.stream.channels*music.stream.sampleSize/8, 1); - - int samplesCount = 0; // Total size of data streamed in L+R samples for xm floats, individual L or R for ogg shorts - - // TODO: Get the sampleLeft using totalFramesProcessed... but first, get total frames processed correctly... - //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels; - int sampleLeft = music.sampleCount - (music.stream.buffer->totalFramesProcessed*music.stream.channels); - - if (music.ctxType == MUSIC_MODULE_XM && music.looping) sampleLeft = subBufferSizeInFrames*4; - - while (IsAudioStreamProcessed(music.stream)) - { - if ((sampleLeft/music.stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music.stream.channels; - else samplesCount = sampleLeft; - - switch (music.ctxType) - { - #if defined(SUPPORT_FILEFORMAT_WAV) - case MUSIC_AUDIO_WAV: - { - // NOTE: Returns the number of samples to process (not required) - if (music.stream.sampleSize == 16) drwav_read_pcm_frames_s16((drwav *)music.ctxData, samplesCount/music.stream.channels, (short *)pcm); - else if (music.stream.sampleSize == 32) drwav_read_pcm_frames_f32((drwav *)music.ctxData, samplesCount/music.stream.channels, (float *)pcm); - - } break; - #endif - #if defined(SUPPORT_FILEFORMAT_OGG) - case MUSIC_AUDIO_OGG: - { - // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!) - stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)pcm, samplesCount); - - } break; - #endif - #if defined(SUPPORT_FILEFORMAT_FLAC) - case MUSIC_AUDIO_FLAC: - { - // NOTE: Returns the number of samples to process (not required) - drflac_read_pcm_frames_s16((drflac *)music.ctxData, samplesCount, (short *)pcm); - - } break; - #endif - #if defined(SUPPORT_FILEFORMAT_MP3) - case MUSIC_AUDIO_MP3: - { - // NOTE: samplesCount, actually refers to framesCount and returns the number of frames processed - drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, samplesCount/music.stream.channels, (float *)pcm); - - } break; - #endif - #if defined(SUPPORT_FILEFORMAT_XM) - case MUSIC_MODULE_XM: - { - switch (AUDIO_DEVICE_FORMAT) - { - case ma_format_f32: - // NOTE: Internally this function considers 2 channels generation, so samplesCount/2 - jar_xm_generate_samples((jar_xm_context_t*)music.ctxData, (float*)pcm, samplesCount / 2); - break; - - case ma_format_s16: - // NOTE: Internally this function considers 2 channels generation, so samplesCount/2 - jar_xm_generate_samples_16bit((jar_xm_context_t*)music.ctxData, (short*)pcm, samplesCount / 2); - break; - - case ma_format_u8: - // NOTE: Internally this function considers 2 channels generation, so samplesCount/2 - jar_xm_generate_samples_8bit((jar_xm_context_t*)music.ctxData, (char*)pcm, samplesCount / 2); - break; - } - - } break; - #endif - #if defined(SUPPORT_FILEFORMAT_MOD) - case MUSIC_MODULE_MOD: - { - // NOTE: 3rd parameter (nbsample) specify the number of stereo 16bits samples you want, so sampleCount/2 - jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)pcm, samplesCount/2, 0); - } break; - #endif - default: break; - } - - UpdateAudioStream(music.stream, pcm, samplesCount); - - if ((music.ctxType == MUSIC_MODULE_XM) || music.ctxType == MUSIC_MODULE_MOD) - { - if (samplesCount > 1) sampleLeft -= samplesCount/2; - else sampleLeft -= samplesCount; - } - else sampleLeft -= samplesCount; - - if (sampleLeft <= 0) - { - streamEnding = true; - break; - } - } - - // Free allocated pcm data - RL_FREE(pcm); - - // Reset audio stream for looping - if (streamEnding) - { - StopMusicStream(music); // Stop music (and reset) - if (music.looping) PlayMusicStream(music); // Play again - } - else - { - // NOTE: In case window is minimized, music stream is stopped, - // just make sure to play again on window restore - if (IsMusicPlaying(music)) PlayMusicStream(music); - } -} - -// Check if any music is playing -bool IsMusicPlaying(Music music) -{ - return IsAudioStreamPlaying(music.stream); -} - -// Set volume for music -void SetMusicVolume(Music music, float volume) -{ - SetAudioStreamVolume(music.stream, volume); -} - -// Set pitch for music -void SetMusicPitch(Music music, float pitch) -{ - SetAudioBufferPitch(music.stream.buffer, pitch); -} - -// Get music time length (in seconds) -float GetMusicTimeLength(Music music) -{ - float totalSeconds = 0.0f; - - totalSeconds = (float)music.sampleCount/(music.stream.sampleRate*music.stream.channels); - - return totalSeconds; -} - -// Get current music time played (in seconds) -float GetMusicTimePlayed(Music music) -{ -#if defined(SUPPORT_FILEFORMAT_XM) - if (music.ctxType == MUSIC_MODULE_XM) - { - uint64_t samples = 0; - jar_xm_get_position(music.ctxData, NULL, NULL, NULL, &samples); - samples = samples % (music.sampleCount); - - return (float)(samples)/(music.stream.sampleRate*music.stream.channels); - } -#endif - float secondsPlayed = 0.0f; - if (music.stream.buffer != NULL) - { - //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels; - unsigned int samplesPlayed = music.stream.buffer->totalFramesProcessed*music.stream.channels; - secondsPlayed = (float)samplesPlayed/(music.stream.sampleRate*music.stream.channels); - } - - return secondsPlayed; -} - -// Init audio stream (to stream audio pcm data) -AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels) -{ - AudioStream stream = { 0 }; - - stream.sampleRate = sampleRate; - stream.sampleSize = sampleSize; - stream.channels = channels; - - ma_format formatIn = ((stream.sampleSize == 8)? ma_format_u8 : ((stream.sampleSize == 16)? ma_format_s16 : ma_format_f32)); - - // The size of a streaming buffer must be at least double the size of a period - unsigned int periodSize = AUDIO.System.device.playback.internalPeriodSizeInFrames; - unsigned int subBufferSize = GetAudioStreamBufferSizeDefault(); - - if (subBufferSize < periodSize) subBufferSize = periodSize; - - // Create a double audio buffer of defined size - stream.buffer = LoadAudioBuffer(formatIn, stream.channels, stream.sampleRate, subBufferSize*2, AUDIO_BUFFER_USAGE_STREAM); - - if (stream.buffer != NULL) - { - stream.buffer->looping = true; // Always loop for streaming buffers - TRACELOG(LOG_INFO, "STREAM: Initialized successfully (%i Hz, %i bit, %s)", stream.sampleRate, stream.sampleSize, (stream.channels == 1)? "Mono" : "Stereo"); - } - else TRACELOG(LOG_WARNING, "STREAM: Failed to load audio buffer, stream could not be created"); - - return stream; -} - -// Close audio stream and free memory -void CloseAudioStream(AudioStream stream) -{ - UnloadAudioBuffer(stream.buffer); - - TRACELOG(LOG_INFO, "STREAM: Unloaded audio stream data from RAM"); -} - -// Update audio stream buffers with data -// NOTE 1: Only updates one buffer of the stream source: unqueue -> update -> queue -// NOTE 2: To unqueue a buffer it needs to be processed: IsAudioStreamProcessed() -void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount) -{ - if (stream.buffer != NULL) - { - if (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1]) - { - ma_uint32 subBufferToUpdate = 0; - - if (stream.buffer->isSubBufferProcessed[0] && stream.buffer->isSubBufferProcessed[1]) - { - // Both buffers are available for updating. - // Update the first one and make sure the cursor is moved back to the front. - subBufferToUpdate = 0; - stream.buffer->frameCursorPos = 0; - } - else - { - // Just update whichever sub-buffer is processed. - subBufferToUpdate = (stream.buffer->isSubBufferProcessed[0])? 0 : 1; - } - - ma_uint32 subBufferSizeInFrames = stream.buffer->sizeInFrames/2; - unsigned char *subBuffer = stream.buffer->data + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate); - - // TODO: Get total frames processed on this buffer... DOES NOT WORK. - stream.buffer->totalFramesProcessed += subBufferSizeInFrames; - - // Does this API expect a whole buffer to be updated in one go? - // Assuming so, but if not will need to change this logic. - if (subBufferSizeInFrames >= (ma_uint32)samplesCount/stream.channels) - { - ma_uint32 framesToWrite = subBufferSizeInFrames; - - if (framesToWrite > ((ma_uint32)samplesCount/stream.channels)) framesToWrite = (ma_uint32)samplesCount/stream.channels; - - ma_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8); - memcpy(subBuffer, data, bytesToWrite); - - // Any leftover frames should be filled with zeros. - ma_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite; - - if (leftoverFrameCount > 0) memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8)); - - stream.buffer->isSubBufferProcessed[subBufferToUpdate] = false; - } - else TRACELOG(LOG_WARNING, "STREAM: Attempting to write too many frames to buffer"); - } - else TRACELOG(LOG_WARNING, "STREAM: Buffer not available for updating"); - } -} - -// Check if any audio stream buffers requires refill -bool IsAudioStreamProcessed(AudioStream stream) -{ - if (stream.buffer == NULL) return false; - - return (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1]); -} - -// Play audio stream -void PlayAudioStream(AudioStream stream) -{ - PlayAudioBuffer(stream.buffer); -} - -// Play audio stream -void PauseAudioStream(AudioStream stream) -{ - PauseAudioBuffer(stream.buffer); -} - -// Resume audio stream playing -void ResumeAudioStream(AudioStream stream) -{ - ResumeAudioBuffer(stream.buffer); -} - -// Check if audio stream is playing. -bool IsAudioStreamPlaying(AudioStream stream) -{ - return IsAudioBufferPlaying(stream.buffer); -} - -// Stop audio stream -void StopAudioStream(AudioStream stream) -{ - StopAudioBuffer(stream.buffer); -} - -// Set volume for audio stream (1.0 is max level) -void SetAudioStreamVolume(AudioStream stream, float volume) -{ - SetAudioBufferVolume(stream.buffer, volume); -} - -// Set pitch for audio stream (1.0 is base level) -void SetAudioStreamPitch(AudioStream stream, float pitch) -{ - SetAudioBufferPitch(stream.buffer, pitch); -} - -// Default size for new audio streams -void SetAudioStreamBufferSizeDefault(int size) -{ - AUDIO.Buffer.defaultSize = size; -} - -int GetAudioStreamBufferSizeDefault() -{ - // if the buffer is not set, compute one that would give us a buffer good enough for a decent frame rate - if (AUDIO.Buffer.defaultSize == 0) - AUDIO.Buffer.defaultSize = AUDIO.System.device.sampleRate/30; - - return AUDIO.Buffer.defaultSize; -} - -//---------------------------------------------------------------------------------- -// Module specific Functions Definition -//---------------------------------------------------------------------------------- - -// Log callback function -static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message) -{ - (void)pContext; - (void)pDevice; - - TRACELOG(LOG_WARNING, "miniaudio: %s", message); // All log messages from miniaudio are errors -} - -// Reads audio data from an AudioBuffer object in internal format. -static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer, void *framesOut, ma_uint32 frameCount) -{ - ma_uint32 subBufferSizeInFrames = (audioBuffer->sizeInFrames > 1)? audioBuffer->sizeInFrames/2 : audioBuffer->sizeInFrames; - ma_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos/subBufferSizeInFrames; - - if (currentSubBufferIndex > 1) return 0; - - // Another thread can update the processed state of buffers so - // we just take a copy here to try and avoid potential synchronization problems - bool isSubBufferProcessed[2]; - isSubBufferProcessed[0] = audioBuffer->isSubBufferProcessed[0]; - isSubBufferProcessed[1] = audioBuffer->isSubBufferProcessed[1]; - - ma_uint32 frameSizeInBytes = ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn); - - // Fill out every frame until we find a buffer that's marked as processed. Then fill the remainder with 0 - ma_uint32 framesRead = 0; - while (1) - { - // We break from this loop differently depending on the buffer's usage - // - For static buffers, we simply fill as much data as we can - // - For streaming buffers we only fill the halves of the buffer that are processed - // Unprocessed halves must keep their audio data in-tact - if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC) - { - if (framesRead >= frameCount) break; - } - else - { - if (isSubBufferProcessed[currentSubBufferIndex]) break; - } - - ma_uint32 totalFramesRemaining = (frameCount - framesRead); - if (totalFramesRemaining == 0) break; - - ma_uint32 framesRemainingInOutputBuffer; - if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC) - { - framesRemainingInOutputBuffer = audioBuffer->sizeInFrames - audioBuffer->frameCursorPos; - } - else - { - ma_uint32 firstFrameIndexOfThisSubBuffer = subBufferSizeInFrames*currentSubBufferIndex; - framesRemainingInOutputBuffer = subBufferSizeInFrames - (audioBuffer->frameCursorPos - firstFrameIndexOfThisSubBuffer); - } - - ma_uint32 framesToRead = totalFramesRemaining; - if (framesToRead > framesRemainingInOutputBuffer) framesToRead = framesRemainingInOutputBuffer; - - memcpy((unsigned char *)framesOut + (framesRead*frameSizeInBytes), audioBuffer->data + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes); - audioBuffer->frameCursorPos = (audioBuffer->frameCursorPos + framesToRead)%audioBuffer->sizeInFrames; - framesRead += framesToRead; - - // If we've read to the end of the buffer, mark it as processed - if (framesToRead == framesRemainingInOutputBuffer) - { - audioBuffer->isSubBufferProcessed[currentSubBufferIndex] = true; - isSubBufferProcessed[currentSubBufferIndex] = true; - - currentSubBufferIndex = (currentSubBufferIndex + 1)%2; - - // We need to break from this loop if we're not looping - if (!audioBuffer->looping) - { - StopAudioBuffer(audioBuffer); - break; - } - } - } - - // Zero-fill excess - ma_uint32 totalFramesRemaining = (frameCount - framesRead); - if (totalFramesRemaining > 0) - { - memset((unsigned char *)framesOut + (framesRead*frameSizeInBytes), 0, totalFramesRemaining*frameSizeInBytes); - - // For static buffers we can fill the remaining frames with silence for safety, but we don't want - // to report those frames as "read". The reason for this is that the caller uses the return value - // to know whether or not a non-looping sound has finished playback. - if (audioBuffer->usage != AUDIO_BUFFER_USAGE_STATIC) framesRead += totalFramesRemaining; - } - - return framesRead; -} - -// Reads audio data from an AudioBuffer object in device format. Returned data will be in a format appropriate for mixing. -static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, float *framesOut, ma_uint32 frameCount) -{ - // What's going on here is that we're continuously converting data from the AudioBuffer's internal format to the mixing format, which - // should be defined by the output format of the data converter. We do this until frameCount frames have been output. The important - // detail to remember here is that we never, ever attempt to read more input data than is required for the specified number of output - // frames. This can be achieved with ma_data_converter_get_required_input_frame_count(). - ma_uint8 inputBuffer[4096]; - ma_uint32 inputBufferFrameCap = sizeof(inputBuffer)/ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn); - - ma_uint32 totalOutputFramesProcessed = 0; - while (totalOutputFramesProcessed < frameCount) - { - ma_uint64 outputFramesToProcessThisIteration = frameCount - totalOutputFramesProcessed; - - ma_uint64 inputFramesToProcessThisIteration = ma_data_converter_get_required_input_frame_count(&audioBuffer->converter, outputFramesToProcessThisIteration); - if (inputFramesToProcessThisIteration > inputBufferFrameCap) - { - inputFramesToProcessThisIteration = inputBufferFrameCap; - } - - float *runningFramesOut = framesOut + (totalOutputFramesProcessed*audioBuffer->converter.config.channelsOut); - - /* At this point we can convert the data to our mixing format. */ - ma_uint64 inputFramesProcessedThisIteration = ReadAudioBufferFramesInInternalFormat(audioBuffer, inputBuffer, (ma_uint32)inputFramesToProcessThisIteration); /* Safe cast. */ - ma_uint64 outputFramesProcessedThisIteration = outputFramesToProcessThisIteration; - ma_data_converter_process_pcm_frames(&audioBuffer->converter, inputBuffer, &inputFramesProcessedThisIteration, runningFramesOut, &outputFramesProcessedThisIteration); - - totalOutputFramesProcessed += (ma_uint32)outputFramesProcessedThisIteration; /* Safe cast. */ - - if (inputFramesProcessedThisIteration < inputFramesToProcessThisIteration) - { - break; /* Ran out of input data. */ - } - - /* This should never be hit, but will add it here for safety. Ensures we get out of the loop when no input nor output frames are processed. */ - if (inputFramesProcessedThisIteration == 0 && outputFramesProcessedThisIteration == 0) - { - break; - } - } - - return totalOutputFramesProcessed; -} - - -// Sending audio data to device callback function -// NOTE: All the mixing takes place here -static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount) -{ - (void)pDevice; - - // Mixing is basically just an accumulation, we need to initialize the output buffer to 0 - memset(pFramesOut, 0, frameCount*pDevice->playback.channels*ma_get_bytes_per_sample(pDevice->playback.format)); - - // Using a mutex here for thread-safety which makes things not real-time - // This is unlikely to be necessary for this project, but may want to consider how you might want to avoid this - ma_mutex_lock(&AUDIO.System.lock); - { - for (AudioBuffer *audioBuffer = AUDIO.Buffer.first; audioBuffer != NULL; audioBuffer = audioBuffer->next) - { - // Ignore stopped or paused sounds - if (!audioBuffer->playing || audioBuffer->paused) continue; - - ma_uint32 framesRead = 0; - - while (1) - { - if (framesRead >= frameCount) break; - - // Just read as much data as we can from the stream - ma_uint32 framesToRead = (frameCount - framesRead); - - while (framesToRead > 0) - { - float tempBuffer[1024]; // 512 frames for stereo - - ma_uint32 framesToReadRightNow = framesToRead; - if (framesToReadRightNow > sizeof(tempBuffer)/sizeof(tempBuffer[0])/AUDIO_DEVICE_CHANNELS) - { - framesToReadRightNow = sizeof(tempBuffer)/sizeof(tempBuffer[0])/AUDIO_DEVICE_CHANNELS; - } - - ma_uint32 framesJustRead = ReadAudioBufferFramesInMixingFormat(audioBuffer, tempBuffer, framesToReadRightNow); - if (framesJustRead > 0) - { - float *framesOut = (float *)pFramesOut + (framesRead*AUDIO.System.device.playback.channels); - float *framesIn = tempBuffer; - - MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume); - - framesToRead -= framesJustRead; - framesRead += framesJustRead; - } - - if (!audioBuffer->playing) - { - framesRead = frameCount; - break; - } - - // If we weren't able to read all the frames we requested, break - if (framesJustRead < framesToReadRightNow) - { - if (!audioBuffer->looping) - { - StopAudioBuffer(audioBuffer); - break; - } - else - { - // Should never get here, but just for safety, - // move the cursor position back to the start and continue the loop - audioBuffer->frameCursorPos = 0; - continue; - } - } - } - - // If for some reason we weren't able to read every frame we'll need to break from the loop - // Not doing this could theoretically put us into an infinite loop - if (framesToRead > 0) break; - } - } - } - - ma_mutex_unlock(&AUDIO.System.lock); -} - -// This is the main mixing function. Mixing is pretty simple in this project - it's just an accumulation. -// NOTE: framesOut is both an input and an output. It will be initially filled with zeros outside of this function. -static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume) -{ - for (ma_uint32 iFrame = 0; iFrame < frameCount; ++iFrame) - { - for (ma_uint32 iChannel = 0; iChannel < AUDIO.System.device.playback.channels; ++iChannel) - { - float *frameOut = framesOut + (iFrame*AUDIO.System.device.playback.channels); - const float *frameIn = framesIn + (iFrame*AUDIO.System.device.playback.channels); - - frameOut[iChannel] += (frameIn[iChannel]*localVolume); - } - } -} - -#if defined(SUPPORT_FILEFORMAT_WAV) -// Load WAV file data into Wave structure -// NOTE: Using dr_wav library -static Wave LoadWAV(const unsigned char *fileData, unsigned int fileSize) -{ - Wave wave = { 0 }; - drwav wav = { 0 }; - - bool success = drwav_init_memory(&wav, fileData, fileSize, NULL); - - if (success) - { - wave.sampleCount = (unsigned int)wav.totalPCMFrameCount*wav.channels; - wave.sampleRate = wav.sampleRate; - wave.sampleSize = 16; // NOTE: We are forcing conversion to 16bit - wave.channels = wav.channels; - wave.data = (short *)RL_MALLOC(wave.sampleCount*sizeof(short)); - drwav_read_pcm_frames_s16(&wav, wav.totalPCMFrameCount, wave.data); - } - else TRACELOG(LOG_WARNING, "WAVE: Failed to load WAV data"); - - drwav_uninit(&wav); - - return wave; -} - -// Save wave data as WAV file -// NOTE: Using dr_wav library -static int SaveWAV(Wave wave, const char *fileName) -{ - int success = false; - - drwav wav = { 0 }; - drwav_data_format format = { 0 }; - format.container = drwav_container_riff; - format.format = DR_WAVE_FORMAT_PCM; - format.channels = wave.channels; - format.sampleRate = wave.sampleRate; - format.bitsPerSample = wave.sampleSize; - - void *fileData = NULL; - size_t fileDataSize = 0; - success = drwav_init_memory_write(&wav, &fileData, &fileDataSize, &format, NULL); - if (success) success = (int)drwav_write_pcm_frames(&wav, wave.sampleCount/wave.channels, wave.data); - drwav_result result = drwav_uninit(&wav); - - if (result == DRWAV_SUCCESS) success = SaveFileData(fileName, (unsigned char *)fileData, (unsigned int)fileDataSize); - - drwav_free(fileData, NULL); - - return success; -} -#endif - -#if defined(SUPPORT_FILEFORMAT_OGG) -// Load OGG file data into Wave structure -// NOTE: Using stb_vorbis library -static Wave LoadOGG(const unsigned char *fileData, unsigned int fileSize) -{ - Wave wave = { 0 }; - - stb_vorbis *oggData = stb_vorbis_open_memory((unsigned char *)fileData, fileSize, NULL, NULL); - - if (oggData != NULL) - { - stb_vorbis_info info = stb_vorbis_get_info(oggData); - - wave.sampleRate = info.sample_rate; - wave.sampleSize = 16; // 16 bit per sample (short) - wave.channels = info.channels; - wave.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggData)*info.channels; // Independent by channel - - float totalSeconds = stb_vorbis_stream_length_in_seconds(oggData); - if (totalSeconds > 10) TRACELOG(LOG_WARNING, "WAVE: OGG audio length larger than 10 seconds (%f sec.), that's a big file in memory, consider music streaming", totalSeconds); - - wave.data = (short *)RL_MALLOC(wave.sampleCount*sizeof(short)); - - // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!) - stb_vorbis_get_samples_short_interleaved(oggData, info.channels, (short *)wave.data, wave.sampleCount); - TRACELOG(LOG_INFO, "WAVE: OGG data loaded successfully (%i Hz, %i bit, %s)", wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo"); - - stb_vorbis_close(oggData); - } - else TRACELOG(LOG_WARNING, "WAVE: Failed to load OGG data"); - - return wave; -} -#endif - -#if defined(SUPPORT_FILEFORMAT_FLAC) -// Load FLAC file data into Wave structure -// NOTE: Using dr_flac library -static Wave LoadFLAC(const unsigned char *fileData, unsigned int fileSize) -{ - Wave wave = { 0 }; - - // Decode the entire FLAC file in one go - unsigned long long int totalFrameCount = 0; - wave.data = drflac_open_memory_and_read_pcm_frames_s16(fileData, fileSize, &wave.channels, &wave.sampleRate, &totalFrameCount, NULL); - - if (wave.data != NULL) - { - wave.sampleCount = (unsigned int)totalFrameCount*wave.channels; - wave.sampleSize = 16; - - TRACELOG(LOG_INFO, "WAVE: FLAC data loaded successfully (%i Hz, %i bit, %s)", wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo"); - } - else TRACELOG(LOG_WARNING, "WAVE: Failed to load FLAC data"); - - return wave; -} -#endif - -#if defined(SUPPORT_FILEFORMAT_MP3) -// Load MP3 file data into Wave structure -// NOTE: Using dr_mp3 library -static Wave LoadMP3(const unsigned char *fileData, unsigned int fileSize) -{ - Wave wave = { 0 }; - drmp3_config config = { 0 }; - - // Decode the entire MP3 file in one go - unsigned long long int totalFrameCount = 0; - wave.data = drmp3_open_memory_and_read_pcm_frames_f32(fileData, fileSize, &config, &totalFrameCount, NULL); - - if (wave.data != NULL) - { - wave.channels = config.channels; - wave.sampleRate = config.sampleRate; - wave.sampleCount = (int)totalFrameCount*wave.channels; - wave.sampleSize = 32; - - // NOTE: Only support up to 2 channels (mono, stereo) - // TODO: Really? - if (wave.channels > 2) TRACELOG(LOG_WARNING, "WAVE: MP3 channels number (%i) not supported", wave.channels); - - TRACELOG(LOG_INFO, "WAVE: MP3 file loaded successfully (%i Hz, %i bit, %s)", wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo"); - } - else TRACELOG(LOG_WARNING, "WAVE: Failed to load MP3 data"); - - return wave; -} -#endif - -// Some required functions for audio standalone module version -#if defined(RAUDIO_STANDALONE) -// Check file extension -static bool IsFileExtension(const char *fileName, const char *ext) -{ - bool result = false; - const char *fileExt; - - if ((fileExt = strrchr(fileName, '.')) != NULL) - { - if (strcmp(fileExt, ext) == 0) result = true; - } - - return result; -} - -// Load data from file into a buffer -static unsigned char *LoadFileData(const char *fileName, unsigned int *bytesRead) -{ - unsigned char *data = NULL; - *bytesRead = 0; - - if (fileName != NULL) - { - FILE *file = fopen(fileName, "rb"); - - if (file != NULL) - { - // WARNING: On binary streams SEEK_END could not be found, - // using fseek() and ftell() could not work in some (rare) cases - fseek(file, 0, SEEK_END); - int size = ftell(file); - fseek(file, 0, SEEK_SET); - - if (size > 0) - { - data = (unsigned char *)RL_MALLOC(size*sizeof(unsigned char)); - - // NOTE: fread() returns number of read elements instead of bytes, so we read [1 byte, size elements] - unsigned int count = (unsigned int)fread(data, sizeof(unsigned char), size, file); - *bytesRead = count; - - if (count != size) TRACELOG(LOG_WARNING, "FILEIO: [%s] File partially loaded", fileName); - else TRACELOG(LOG_INFO, "FILEIO: [%s] File loaded successfully", fileName); - } - else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to read file", fileName); - - fclose(file); - } - else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open file", fileName); - } - else TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid"); - - return data; -} - -// Save data to file from buffer -static bool SaveFileData(const char *fileName, void *data, unsigned int bytesToWrite) -{ - if (fileName != NULL) - { - FILE *file = fopen(fileName, "wb"); - - if (file != NULL) - { - unsigned int count = (unsigned int)fwrite(data, sizeof(unsigned char), bytesToWrite, file); - - if (count == 0) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to write file", fileName); - else if (count != bytesToWrite) TRACELOG(LOG_WARNING, "FILEIO: [%s] File partially written", fileName); - else TRACELOG(LOG_INFO, "FILEIO: [%s] File saved successfully", fileName); - - fclose(file); - } - else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open file", fileName); - } - else TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid"); -} - -// Save text data to file (write), string must be '\0' terminated -static bool SaveFileText(const char *fileName, char *text) -{ - if (fileName != NULL) - { - FILE *file = fopen(fileName, "wt"); - - if (file != NULL) - { - int count = fprintf(file, "%s", text); - - if (count == 0) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to write text file", fileName); - else TRACELOG(LOG_INFO, "FILEIO: [%s] Text file saved successfully", fileName); - - fclose(file); - } - else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open text file", fileName); - } - else TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid"); -} -#endif - -#undef AudioBuffer |