diff options
Diffstat (limited to 'src/audio.c')
| -rw-r--r-- | src/audio.c | 647 |
1 files changed, 329 insertions, 318 deletions
diff --git a/src/audio.c b/src/audio.c index d397b064..3ee1fe81 100644 --- a/src/audio.c +++ b/src/audio.c @@ -11,32 +11,40 @@ * - Manage raw audio context * * CONFIGURATION: -* +* * #define AUDIO_STANDALONE * Define to use the module as standalone library (independently of raylib). * Required types and functions are defined in the same module. * +* #define USE_OPENAL_BACKEND +* Use OpenAL Soft audio backend usage +* * #define SUPPORT_FILEFORMAT_WAV * #define SUPPORT_FILEFORMAT_OGG * #define SUPPORT_FILEFORMAT_XM * #define SUPPORT_FILEFORMAT_MOD * #define SUPPORT_FILEFORMAT_FLAC -* Selected desired fileformats to be supported for loading. Some of those formats are +* 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 * -* LIMITATIONS: +* LIMITATIONS (only OpenAL Soft): * Only up to two channels supported: MONO and STEREO (for additional channels, use AL_EXT_MCFORMATS) * Only the following sample sizes supported: 8bit PCM, 16bit PCM, 32-bit float PCM (using AL_EXT_FLOAT32) * * DEPENDENCIES: -* OpenAL Soft - Audio device management (http://kcat.strangesoft.net/openal.html) +* mini_al - Audio device/context management (https://github.com/dr-soft/mini_al) * stb_vorbis - OGG audio files loading (http://www.nothings.org/stb_vorbis/) * jar_xm - XM module file loading * jar_mod - MOD audio file loading * dr_flac - FLAC audio file loading * +* *OpenAL Soft - Audio device management, still used on HTML5 and OSX platforms +* * CONTRIBUTORS: -* Joshua Reisenauer (github: @kd7tck): +* David Reid (github: @mackron) (Nov. 2017): +* - Complete port to mini_al library +* +* Joshua Reisenauer (github: @kd7tck) (2015) * - XM audio module support (jar_xm) * - MOD audio module support (jar_mod) * - Mixing channels support @@ -45,7 +53,7 @@ * * LICENSE: zlib/libpng * -* Copyright (c) 2014-2017 Ramon Santamaria (@raysan5) +* Copyright (c) 2014-2018 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. @@ -64,16 +72,10 @@ * **********************************************************************************************/ -// Default configuration flags (supported features) -//------------------------------------------------- -#define SUPPORT_FILEFORMAT_WAV -#define SUPPORT_FILEFORMAT_OGG -#define SUPPORT_FILEFORMAT_XM -#define SUPPORT_FILEFORMAT_MOD -//------------------------------------------------- +#include "config.h" -#ifndef USE_MINI_AL -#define USE_MINI_AL 1 // Set to 1 to use mini_al; 0 to use OpenAL. +#if !defined(USE_OPENAL_BACKEND) + #define USE_MINI_AL 1 // Set to 1 to use mini_al; 0 to use OpenAL. #endif #if defined(AUDIO_STANDALONE) @@ -86,7 +88,7 @@ #include "external/mini_al.h" // Implemented in mini_al.c. Cannot implement this here because it conflicts with Win32 APIs such as CloseWindow(), etc. -#if !defined(USE_MINI_AL) || USE_MINI_AL == 0 +#if !defined(USE_MINI_AL) || (USE_MINI_AL == 0) #if defined(__APPLE__) #include "OpenAL/al.h" // OpenAL basic header #include "OpenAL/alc.h" // OpenAL context header (like OpenGL, OpenAL requires a context to work) @@ -153,7 +155,12 @@ // Types and Structures Definition //---------------------------------------------------------------------------------- -typedef enum { MUSIC_AUDIO_OGG = 0, MUSIC_AUDIO_FLAC, MUSIC_MODULE_XM, MUSIC_MODULE_MOD } MusicContextType; +typedef enum { + MUSIC_AUDIO_OGG = 0, + MUSIC_AUDIO_FLAC, + MUSIC_MODULE_XM, + MUSIC_MODULE_MOD +} MusicContextType; // Music type (file streaming from memory) typedef struct MusicData { @@ -179,7 +186,13 @@ typedef struct MusicData { } MusicData; #if defined(AUDIO_STANDALONE) -typedef enum { LOG_INFO = 0, LOG_ERROR, LOG_WARNING, LOG_DEBUG, LOG_OTHER } TraceLogType; +typedef enum { + LOG_INFO = 0, + LOG_ERROR, + LOG_WARNING, + LOG_DEBUG, + LOG_OTHER +} TraceLogType; #endif //---------------------------------------------------------------------------------- @@ -206,105 +219,78 @@ void TraceLog(int msgType, const char *text, ...); // Show trace lo #endif //---------------------------------------------------------------------------------- -// Module Functions Definition - Audio Device initialization and Closing +// mini_al AudioBuffer Functionality //---------------------------------------------------------------------------------- #if USE_MINI_AL + #define DEVICE_FORMAT mal_format_f32 #define DEVICE_CHANNELS 2 #define DEVICE_SAMPLE_RATE 44100 typedef enum { AUDIO_BUFFER_USAGE_STATIC = 0, AUDIO_BUFFER_USAGE_STREAM } AudioBufferUsage; +// Audio buffer structure +// NOTE: Slightly different logic is used when feeding data to the playback device depending on whether or not data is streamed typedef struct AudioBuffer AudioBuffer; -struct AudioBuffer -{ - mal_dsp dsp; // For format conversion. +struct AudioBuffer { + mal_dsp dsp; // Required for format conversion float volume; float pitch; bool playing; bool paused; - bool looping; // Always true for AudioStreams. - AudioBufferUsage usage; // Slightly different logic is used when feeding data to the playback device depending on whether or not data is streamed. + bool looping; // Always true for AudioStreams + int usage; // AudioBufferUsage type bool isSubBufferProcessed[2]; unsigned int frameCursorPos; unsigned int bufferSizeInFrames; - AudioBuffer* next; - AudioBuffer* prev; + AudioBuffer *next; + AudioBuffer *prev; unsigned char buffer[1]; }; -void StopAudioBuffer(AudioBuffer* audioBuffer); - - +// mini_al global variables static mal_context context; static mal_device device; -static mal_bool32 isAudioInitialized = MAL_FALSE; -static float masterVolume = 1; static mal_mutex audioLock; -static AudioBuffer* firstAudioBuffer = NULL; // Audio buffers are tracked in a linked list. -static AudioBuffer* lastAudioBuffer = NULL; - -static void TrackAudioBuffer(AudioBuffer* audioBuffer) -{ - mal_mutex_lock(&audioLock); - { - if (firstAudioBuffer == NULL) { - firstAudioBuffer = audioBuffer; - } else { - lastAudioBuffer->next = audioBuffer; - audioBuffer->prev = lastAudioBuffer; - } - - lastAudioBuffer = audioBuffer; - } - mal_mutex_unlock(&audioLock); -} - -static void UntrackAudioBuffer(AudioBuffer* audioBuffer) -{ - mal_mutex_lock(&audioLock); - { - if (audioBuffer->prev == NULL) { - firstAudioBuffer = audioBuffer->next; - } else { - audioBuffer->prev->next = audioBuffer->next; - } - - if (audioBuffer->next == NULL) { - lastAudioBuffer = audioBuffer->prev; - } else { - audioBuffer->next->prev = audioBuffer->prev; - } - - audioBuffer->prev = NULL; - audioBuffer->next = NULL; - } - mal_mutex_unlock(&audioLock); -} - -static void OnLog_MAL(mal_context* pContext, mal_device* pDevice, const char* message) +static bool isAudioInitialized = MAL_FALSE; +static float masterVolume = 1.0f; + +// Audio buffers are tracked in a linked list +static AudioBuffer *firstAudioBuffer = NULL; +static AudioBuffer *lastAudioBuffer = NULL; + +// mini_al functions declaration +static void OnLog(mal_context *pContext, mal_device *pDevice, const char *message); +static mal_uint32 OnSendAudioDataToDevice(mal_device *pDevice, mal_uint32 frameCount, void *pFramesOut); +static mal_uint32 OnAudioBufferDSPRead(mal_dsp *pDSP, mal_uint32 frameCount, void *pFramesOut, void *pUserData); +static void MixAudioFrames(float *framesOut, const float *framesIn, mal_uint32 frameCount, float localVolume); + +// AudioBuffer management functions declaration +// NOTE: Those functions are not exposed by raylib... for the moment +AudioBuffer *CreateAudioBuffer(mal_format format, mal_uint32 channels, mal_uint32 sampleRate, mal_uint32 bufferSizeInFrames, AudioBufferUsage usage); +void DeleteAudioBuffer(AudioBuffer *audioBuffer); +bool IsAudioBufferPlaying(AudioBuffer *audioBuffer); +void PlayAudioBuffer(AudioBuffer *audioBuffer); +void StopAudioBuffer(AudioBuffer *audioBuffer); +void PauseAudioBuffer(AudioBuffer *audioBuffer); +void ResumeAudioBuffer(AudioBuffer *audioBuffer); +void SetAudioBufferVolume(AudioBuffer *audioBuffer, float volume); +void SetAudioBufferPitch(AudioBuffer *audioBuffer, float pitch); +void TrackAudioBuffer(AudioBuffer *audioBuffer); +void UntrackAudioBuffer(AudioBuffer *audioBuffer); + + +// Log callback function +static void OnLog(mal_context *pContext, mal_device *pDevice, const char *message) { (void)pContext; (void)pDevice; - TraceLog(LOG_ERROR, message); // All log messages from mini_al are errors. -} - -// This is the main mixing function. Mixing is pretty simple in this project - it's just an accumulation. -// -// framesOut is both an input and an output. It will be initially filled with zeros outside of this function. -static void MixFrames(float* framesOut, const float* framesIn, mal_uint32 frameCount, float localVolume) -{ - for (mal_uint32 iFrame = 0; iFrame < frameCount; ++iFrame) { - for (mal_uint32 iChannel = 0; iChannel < device.channels; ++iChannel) { - float* frameOut = framesOut + (iFrame * device.channels); - const float* frameIn = framesIn + (iFrame * device.channels); - - frameOut[iChannel] += frameIn[iChannel] * masterVolume * localVolume; - } - } + + TraceLog(LOG_ERROR, message); // All log messages from mini_al are errors } -static mal_uint32 OnSendAudioDataToDevice(mal_device* pDevice, mal_uint32 frameCount, void* pFramesOut) +// Sending audio data to device callback function +static mal_uint32 OnSendAudioDataToDevice(mal_device *pDevice, mal_uint32 frameCount, void *pFramesOut) { // This is where all of the mixing takes place. (void)pDevice; @@ -316,80 +302,194 @@ static mal_uint32 OnSendAudioDataToDevice(mal_device* pDevice, mal_uint32 frameC // want to consider how you might want to avoid this. mal_mutex_lock(&audioLock); { - for (AudioBuffer* audioBuffer = firstAudioBuffer; audioBuffer != NULL; audioBuffer = audioBuffer->next) + for (AudioBuffer *audioBuffer = firstAudioBuffer; audioBuffer != NULL; audioBuffer = audioBuffer->next) { // Ignore stopped or paused sounds. - if (!audioBuffer->playing || audioBuffer->paused) { - continue; - } + if (!audioBuffer->playing || audioBuffer->paused) continue; mal_uint32 framesRead = 0; - for (;;) { - if (framesRead > frameCount) { + for (;;) + { + if (framesRead > frameCount) + { TraceLog(LOG_DEBUG, "Mixed too many frames from audio buffer"); break; } - if (framesRead == frameCount) { - break; - } + + if (framesRead == frameCount) break; // Just read as much data as we can from the stream. mal_uint32 framesToRead = (frameCount - framesRead); - while (framesToRead > 0) { + while (framesToRead > 0) + { float tempBuffer[1024]; // 512 frames for stereo. mal_uint32 framesToReadRightNow = framesToRead; - if (framesToReadRightNow > sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS) { + if (framesToReadRightNow > sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS) + { framesToReadRightNow = sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS; } // If we're not looping, we need to make sure we flush the internal buffers of the DSP pipeline to ensure we get the // last few samples. - mal_bool32 flushDSP = !audioBuffer->looping; + bool flushDSP = !audioBuffer->looping; mal_uint32 framesJustRead = mal_dsp_read_frames_ex(&audioBuffer->dsp, framesToReadRightNow, tempBuffer, flushDSP); - if (framesJustRead > 0) { - float* framesOut = (float*)pFramesOut + (framesRead * device.channels); - float* framesIn = tempBuffer; - MixFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume); + if (framesJustRead > 0) + { + float *framesOut = (float *)pFramesOut + (framesRead*device.channels); + float *framesIn = tempBuffer; + MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume); framesToRead -= framesJustRead; framesRead += framesJustRead; } // If we weren't able to read all the frames we requested, break. - if (framesJustRead < framesToReadRightNow) { - if (!audioBuffer->looping) { + 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. + } + 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; - } + // 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; } } } + mal_mutex_unlock(&audioLock); return frameCount; // We always output the same number of frames that were originally requested. } + +// DSP read from audio buffer callback function +static mal_uint32 OnAudioBufferDSPRead(mal_dsp *pDSP, mal_uint32 frameCount, void *pFramesOut, void *pUserData) +{ + AudioBuffer *audioBuffer = (AudioBuffer *)pUserData; + + mal_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2; + mal_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos/subBufferSizeInFrames; + + if (currentSubBufferIndex > 1) + { + TraceLog(LOG_DEBUG, "Frame cursor position moved too far forward in audio stream"); + 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]; + + mal_uint32 frameSizeInBytes = mal_get_sample_size_in_bytes(audioBuffer->dsp.config.formatIn)*audioBuffer->dsp.config.channelsIn; + + // Fill out every frame until we find a buffer that's marked as processed. Then fill the remainder with 0. + mal_uint32 framesRead = 0; + for (;;) + { + // 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; + } + + mal_uint32 totalFramesRemaining = (frameCount - framesRead); + if (totalFramesRemaining == 0) break; + + mal_uint32 framesRemainingInOutputBuffer; + if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC) + { + framesRemainingInOutputBuffer = audioBuffer->bufferSizeInFrames - audioBuffer->frameCursorPos; + } + else + { + mal_uint32 firstFrameIndexOfThisSubBuffer = subBufferSizeInFrames * currentSubBufferIndex; + framesRemainingInOutputBuffer = subBufferSizeInFrames - (audioBuffer->frameCursorPos - firstFrameIndexOfThisSubBuffer); + } + + mal_uint32 framesToRead = totalFramesRemaining; + if (framesToRead > framesRemainingInOutputBuffer) framesToRead = framesRemainingInOutputBuffer; + + memcpy((unsigned char *)pFramesOut + (framesRead*frameSizeInBytes), audioBuffer->buffer + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes); + audioBuffer->frameCursorPos = (audioBuffer->frameCursorPos + framesToRead) % audioBuffer->bufferSizeInFrames; + 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. + mal_uint32 totalFramesRemaining = (frameCount - framesRead); + if (totalFramesRemaining > 0) + { + memset((unsigned char*)pFramesOut + (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; +} + +// 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, mal_uint32 frameCount, float localVolume) +{ + for (mal_uint32 iFrame = 0; iFrame < frameCount; ++iFrame) + { + for (mal_uint32 iChannel = 0; iChannel < device.channels; ++iChannel) + { + float *frameOut = framesOut + (iFrame*device.channels); + const float *frameIn = framesIn + (iFrame*device.channels); + + frameOut[iChannel] += frameIn[iChannel]*masterVolume*localVolume; + } + } +} #endif +//---------------------------------------------------------------------------------- +// Module Functions Definition - Audio Device initialization and Closing +//---------------------------------------------------------------------------------- // Initialize audio device void InitAudioDevice(void) { #if USE_MINI_AL // Context. - mal_context_config contextConfig = mal_context_config_init(OnLog_MAL); + mal_context_config contextConfig = mal_context_config_init(OnLog); mal_result result = mal_context_init(NULL, 0, &contextConfig, &context); if (result != MAL_SUCCESS) { @@ -400,12 +500,6 @@ void InitAudioDevice(void) // Device. Using the default device. Format is floating point because it simplifies mixing. mal_device_config deviceConfig = mal_device_config_init(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, NULL, OnSendAudioDataToDevice); - // Special case for PLATFORM_RPI. -//#if defined(PLATFORM_RPI) -// deviceConfig.alsa.noMMap = MAL_TRUE; -// deviceConfig.bufferSizeInFrames = 2048; -//#endif - result = mal_device_init(&context, mal_device_type_playback, NULL, &deviceConfig, NULL, &device); if (result != MAL_SUCCESS) { @@ -471,11 +565,8 @@ void InitAudioDevice(void) alListenerf(AL_GAIN, 1.0f); - if (alIsExtensionPresent("AL_EXT_float32")) { - TraceLog(LOG_INFO, "AL_EXT_float32 supported"); - } else { - TraceLog(LOG_INFO, "AL_EXT_float32 not supported"); - } + if (alIsExtensionPresent("AL_EXT_float32")) TraceLog(LOG_INFO, "[EXTENSION] AL_EXT_float32 supported"); + else TraceLog(LOG_INFO, "[EXTENSION] AL_EXT_float32 not supported"); } } #endif @@ -485,7 +576,8 @@ void InitAudioDevice(void) void CloseAudioDevice(void) { #if USE_MINI_AL - if (!isAudioInitialized) { + if (!isAudioInitialized) + { TraceLog(LOG_WARNING, "Could not close audio device because it is not currently initialized"); return; } @@ -535,110 +627,20 @@ void SetMasterVolume(float volume) else if (volume > 1.0f) volume = 1.0f; #if USE_MINI_AL - masterVolume = 1; + masterVolume = volume; #else alListenerf(AL_GAIN, volume); #endif } - //---------------------------------------------------------------------------------- -// Audio Buffer +// Module Functions Definition - Audio Buffer management //---------------------------------------------------------------------------------- #if USE_MINI_AL -static mal_uint32 AudioBuffer_OnDSPRead(mal_dsp* pDSP, mal_uint32 frameCount, void* pFramesOut, void* pUserData) +// Create a new audio buffer. Initially filled with silence +AudioBuffer *CreateAudioBuffer(mal_format format, mal_uint32 channels, mal_uint32 sampleRate, mal_uint32 bufferSizeInFrames, AudioBufferUsage usage) { - AudioBuffer* audioBuffer = (AudioBuffer*)pUserData; - - mal_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames / 2; - mal_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos / subBufferSizeInFrames; - if (currentSubBufferIndex > 1) { - TraceLog(LOG_DEBUG, "Frame cursor position moved too far forward in audio stream"); - 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]; - - mal_uint32 frameSizeInBytes = mal_get_sample_size_in_bytes(audioBuffer->dsp.config.formatIn) * audioBuffer->dsp.config.channelsIn; - - // Fill out every frame until we find a buffer that's marked as processed. Then fill the remainder with 0. - mal_uint32 framesRead = 0; - for (;;) - { - // 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; - } - } - - mal_uint32 totalFramesRemaining = (frameCount - framesRead); - if (totalFramesRemaining == 0) { - break; - } - - mal_uint32 framesRemainingInOutputBuffer; - if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC) { - framesRemainingInOutputBuffer = audioBuffer->bufferSizeInFrames - audioBuffer->frameCursorPos; - } else { - mal_uint32 firstFrameIndexOfThisSubBuffer = subBufferSizeInFrames * currentSubBufferIndex; - framesRemainingInOutputBuffer = subBufferSizeInFrames - (audioBuffer->frameCursorPos - firstFrameIndexOfThisSubBuffer); - } - - - - mal_uint32 framesToRead = totalFramesRemaining; - if (framesToRead > framesRemainingInOutputBuffer) { - framesToRead = framesRemainingInOutputBuffer; - } - - memcpy((unsigned char*)pFramesOut + (framesRead*frameSizeInBytes), audioBuffer->buffer + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes); - audioBuffer->frameCursorPos = (audioBuffer->frameCursorPos + framesToRead) % audioBuffer->bufferSizeInFrames; - 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. - mal_uint32 totalFramesRemaining = (frameCount - framesRead); - if (totalFramesRemaining > 0) { - memset((unsigned char*)pFramesOut + (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; -} - -// Create a new audio buffer. Initially filled with silence. -AudioBuffer* CreateAudioBuffer(mal_format format, mal_uint32 channels, mal_uint32 sampleRate, mal_uint32 bufferSizeInFrames, AudioBufferUsage usage) -{ - AudioBuffer* audioBuffer = (AudioBuffer*)calloc(sizeof(*audioBuffer) + (bufferSizeInFrames*channels*mal_get_sample_size_in_bytes(format)), 1); + AudioBuffer *audioBuffer = (AudioBuffer *)calloc(sizeof(*audioBuffer) + (bufferSizeInFrames*channels*mal_get_sample_size_in_bytes(format)), 1); if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "CreateAudioBuffer() : Failed to allocate memory for audio buffer"); @@ -654,8 +656,9 @@ AudioBuffer* CreateAudioBuffer(mal_format format, mal_uint32 channels, mal_uint3 dspConfig.channelsOut = DEVICE_CHANNELS; dspConfig.sampleRateIn = sampleRate; dspConfig.sampleRateOut = DEVICE_SAMPLE_RATE; - mal_result resultMAL = mal_dsp_init(&dspConfig, AudioBuffer_OnDSPRead, audioBuffer, &audioBuffer->dsp); - if (resultMAL != MAL_SUCCESS) { + mal_result resultMAL = mal_dsp_init(&dspConfig, OnAudioBufferDSPRead, audioBuffer, &audioBuffer->dsp); + if (resultMAL != MAL_SUCCESS) + { TraceLog(LOG_ERROR, "LoadSoundFromWave() : Failed to create data conversion pipeline"); free(audioBuffer); return NULL; @@ -679,8 +682,8 @@ AudioBuffer* CreateAudioBuffer(mal_format format, mal_uint32 channels, mal_uint3 return audioBuffer; } -// Delete an audio buffer. -void DeleteAudioBuffer(AudioBuffer* audioBuffer) +// Delete an audio buffer +void DeleteAudioBuffer(AudioBuffer *audioBuffer) { if (audioBuffer == NULL) { @@ -692,8 +695,8 @@ void DeleteAudioBuffer(AudioBuffer* audioBuffer) free(audioBuffer); } -// Check if an audio buffer is playing. -bool IsAudioBufferPlaying(AudioBuffer* audioBuffer) +// Check if an audio buffer is playing +bool IsAudioBufferPlaying(AudioBuffer *audioBuffer) { if (audioBuffer == NULL) { @@ -704,11 +707,10 @@ bool IsAudioBufferPlaying(AudioBuffer* audioBuffer) return audioBuffer->playing && !audioBuffer->paused; } -// Play an audio buffer. -// -// This will restart the buffer from the start. Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position -// should be maintained. -void PlayAudioBuffer(AudioBuffer* audioBuffer) +// 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 *audioBuffer) { if (audioBuffer == NULL) { @@ -721,8 +723,8 @@ void PlayAudioBuffer(AudioBuffer* audioBuffer) audioBuffer->frameCursorPos = 0; } -// Stop an audio buffer. -void StopAudioBuffer(AudioBuffer* audioBuffer) +// Stop an audio buffer +void StopAudioBuffer(AudioBuffer *audioBuffer) { if (audioBuffer == NULL) { @@ -731,10 +733,7 @@ void StopAudioBuffer(AudioBuffer* audioBuffer) } // Don't do anything if the audio buffer is already stopped. - if (!IsAudioBufferPlaying(audioBuffer)) - { - return; - } + if (!IsAudioBufferPlaying(audioBuffer)) return; audioBuffer->playing = false; audioBuffer->paused = false; @@ -743,8 +742,8 @@ void StopAudioBuffer(AudioBuffer* audioBuffer) audioBuffer->isSubBufferProcessed[1] = true; } -// Pause an audio buffer. -void PauseAudioBuffer(AudioBuffer* audioBuffer) +// Pause an audio buffer +void PauseAudioBuffer(AudioBuffer *audioBuffer) { if (audioBuffer == NULL) { @@ -755,8 +754,8 @@ void PauseAudioBuffer(AudioBuffer* audioBuffer) audioBuffer->paused = true; } -// Resume an audio buffer. -void ResumeAudioBuffer(AudioBuffer* audioBuffer) +// Resume an audio buffer +void ResumeAudioBuffer(AudioBuffer *audioBuffer) { if (audioBuffer == NULL) { @@ -767,8 +766,8 @@ void ResumeAudioBuffer(AudioBuffer* audioBuffer) audioBuffer->paused = false; } -// Set volume for an audio buffer. -void SetAudioBufferVolume(AudioBuffer* audioBuffer, float volume) +// Set volume for an audio buffer +void SetAudioBufferVolume(AudioBuffer *audioBuffer, float volume) { if (audioBuffer == NULL) { @@ -779,8 +778,8 @@ void SetAudioBufferVolume(AudioBuffer* audioBuffer, float volume) audioBuffer->volume = volume; } -// Set pitch for an audio buffer. -void SetAudioBufferPitch(AudioBuffer* audioBuffer, float pitch) +// Set pitch for an audio buffer +void SetAudioBufferPitch(AudioBuffer *audioBuffer, float pitch) { if (audioBuffer == NULL) { @@ -795,6 +794,44 @@ void SetAudioBufferPitch(AudioBuffer* audioBuffer, float pitch) mal_uint32 newOutputSampleRate = (mal_uint32)((((float)audioBuffer->dsp.config.sampleRateOut / (float)audioBuffer->dsp.config.sampleRateIn) / pitch) * audioBuffer->dsp.config.sampleRateIn); mal_dsp_set_output_sample_rate(&audioBuffer->dsp, newOutputSampleRate); } + +// Track audio buffer to linked list next position +void TrackAudioBuffer(AudioBuffer *audioBuffer) +{ + mal_mutex_lock(&audioLock); + + { + if (firstAudioBuffer == NULL) firstAudioBuffer = audioBuffer; + else + { + lastAudioBuffer->next = audioBuffer; + audioBuffer->prev = lastAudioBuffer; + } + + lastAudioBuffer = audioBuffer; + } + + mal_mutex_unlock(&audioLock); +} + +// Untrack audio buffer from linked list +void UntrackAudioBuffer(AudioBuffer *audioBuffer) +{ + mal_mutex_lock(&audioLock); + + { + if (audioBuffer->prev == NULL) firstAudioBuffer = audioBuffer->next; + else audioBuffer->prev->next = audioBuffer->next; + + if (audioBuffer->next == NULL) lastAudioBuffer = audioBuffer->prev; + else audioBuffer->next->prev = audioBuffer->prev; + + audioBuffer->prev = NULL; + audioBuffer->next = NULL; + } + + mal_mutex_unlock(&audioLock); +} #endif //---------------------------------------------------------------------------------- @@ -813,19 +850,6 @@ Wave LoadWave(const char *fileName) #if defined(SUPPORT_FILEFORMAT_FLAC) else if (IsFileExtension(fileName, ".flac")) wave = LoadFLAC(fileName); #endif -#if !defined(AUDIO_STANDALONE) - else if (IsFileExtension(fileName, ".rres")) - { - RRES rres = LoadResource(fileName, 0); - - // NOTE: Parameters for RRES_TYPE_WAVE are: sampleCount, sampleRate, sampleSize, channels - - if (rres[0].type == RRES_TYPE_WAVE) wave = LoadWaveEx(rres[0].data, rres[0].param1, rres[0].param2, rres[0].param3, rres[0].param4); - else TraceLog(LOG_WARNING, "[%s] Resource file does not contain wave data", fileName); - - UnloadResource(rres); - } -#endif else TraceLog(LOG_WARNING, "[%s] Audio fileformat not supported, it can't be loaded", fileName); return wave; @@ -884,23 +908,13 @@ Sound LoadSoundFromWave(Wave wave) mal_uint32 frameCountIn = wave.sampleCount; // Is wave->sampleCount actually the frame count? That terminology needs to change, if so. mal_uint32 frameCount = mal_convert_frames(NULL, DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, NULL, formatIn, wave.channels, wave.sampleRate, frameCountIn); - if (frameCount == 0) { - TraceLog(LOG_ERROR, "LoadSoundFromWave() : Failed to get frame count for format conversion"); - } - + if (frameCount == 0) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Failed to get frame count for format conversion"); AudioBuffer* audioBuffer = CreateAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC); - if (audioBuffer == NULL) - { - TraceLog(LOG_ERROR, "LoadSoundFromWave() : Failed to create audio buffer"); - } - + if (audioBuffer == NULL) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Failed to create audio buffer"); frameCount = mal_convert_frames(audioBuffer->buffer, audioBuffer->dsp.config.formatIn, audioBuffer->dsp.config.channelsIn, audioBuffer->dsp.config.sampleRateIn, wave.data, formatIn, wave.channels, wave.sampleRate, frameCountIn); - if (frameCount == 0) - { - TraceLog(LOG_ERROR, "LoadSoundFromWave() : Format conversion failed"); - } + if (frameCount == 0) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Format conversion failed"); sound.audioBuffer = audioBuffer; #else @@ -975,7 +989,7 @@ void UnloadWave(Wave wave) void UnloadSound(Sound sound) { #if USE_MINI_AL - DeleteAudioBuffer((AudioBuffer*)sound.audioBuffer); + DeleteAudioBuffer((AudioBuffer *)sound.audioBuffer); #else alSourceStop(sound.source); @@ -991,7 +1005,7 @@ void UnloadSound(Sound sound) void UpdateSound(Sound sound, const void *data, int samplesCount) { #if USE_MINI_AL - AudioBuffer* audioBuffer = (AudioBuffer*)sound.audioBuffer; + AudioBuffer *audioBuffer = (AudioBuffer *)sound.audioBuffer; if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "UpdateSound() : Invalid sound - no audio buffer"); @@ -1031,7 +1045,7 @@ void UpdateSound(Sound sound, const void *data, int samplesCount) void PlaySound(Sound sound) { #if USE_MINI_AL - PlayAudioBuffer((AudioBuffer*)sound.audioBuffer); + PlayAudioBuffer((AudioBuffer *)sound.audioBuffer); #else alSourcePlay(sound.source); // Play the sound #endif @@ -1056,7 +1070,7 @@ void PlaySound(Sound sound) void PauseSound(Sound sound) { #if USE_MINI_AL - PauseAudioBuffer((AudioBuffer*)sound.audioBuffer); + PauseAudioBuffer((AudioBuffer *)sound.audioBuffer); #else alSourcePause(sound.source); #endif @@ -1066,7 +1080,7 @@ void PauseSound(Sound sound) void ResumeSound(Sound sound) { #if USE_MINI_AL - ResumeAudioBuffer((AudioBuffer*)sound.audioBuffer); + ResumeAudioBuffer((AudioBuffer *)sound.audioBuffer); #else ALenum state; @@ -1080,7 +1094,7 @@ void ResumeSound(Sound sound) void StopSound(Sound sound) { #if USE_MINI_AL - StopAudioBuffer((AudioBuffer*)sound.audioBuffer); + StopAudioBuffer((AudioBuffer *)sound.audioBuffer); #else alSourceStop(sound.source); #endif @@ -1090,7 +1104,7 @@ void StopSound(Sound sound) bool IsSoundPlaying(Sound sound) { #if USE_MINI_AL - return IsAudioBufferPlaying((AudioBuffer*)sound.audioBuffer); + return IsAudioBufferPlaying((AudioBuffer *)sound.audioBuffer); #else bool playing = false; ALint state; @@ -1106,7 +1120,7 @@ bool IsSoundPlaying(Sound sound) void SetSoundVolume(Sound sound, float volume) { #if USE_MINI_AL - SetAudioBufferVolume((AudioBuffer*)sound.audioBuffer, volume); + SetAudioBufferVolume((AudioBuffer *)sound.audioBuffer, volume); #else alSourcef(sound.source, AL_GAIN, volume); #endif @@ -1116,7 +1130,7 @@ void SetSoundVolume(Sound sound, float volume) void SetSoundPitch(Sound sound, float pitch) { #if USE_MINI_AL - SetAudioBufferPitch((AudioBuffer*)sound.audioBuffer, pitch); + SetAudioBufferPitch((AudioBuffer *)sound.audioBuffer, pitch); #else alSourcef(sound.source, AL_PITCH, pitch); #endif @@ -1125,21 +1139,24 @@ void SetSoundPitch(Sound sound, float pitch) // Convert wave data to desired format void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels) { +#if USE_MINI_AL mal_format formatIn = ((wave->sampleSize == 8) ? mal_format_u8 : ((wave->sampleSize == 16) ? mal_format_s16 : mal_format_f32)); mal_format formatOut = (( sampleSize == 8) ? mal_format_u8 : (( sampleSize == 16) ? mal_format_s16 : mal_format_f32)); mal_uint32 frameCountIn = wave->sampleCount; // Is wave->sampleCount actually the frame count? That terminology needs to change, if so. mal_uint32 frameCount = mal_convert_frames(NULL, formatOut, channels, sampleRate, NULL, formatIn, wave->channels, wave->sampleRate, frameCountIn); - if (frameCount == 0) { + if (frameCount == 0) + { TraceLog(LOG_ERROR, "WaveFormat() : Failed to get frame count for format conversion."); return; } - void* data = malloc(frameCount * channels * (sampleSize/8)); + void *data = malloc(frameCount*channels*(sampleSize/8)); frameCount = mal_convert_frames(data, formatOut, channels, sampleRate, wave->data, formatIn, wave->channels, wave->sampleRate, frameCountIn); - if (frameCount == 0) { + if (frameCount == 0) + { TraceLog(LOG_ERROR, "WaveFormat() : Format conversion failed."); return; } @@ -1151,7 +1168,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels) free(wave->data); wave->data = data; -#if 0 +#else // Format sample rate // NOTE: Only supported 22050 <--> 44100 if (wave->sampleRate != sampleRate) @@ -1414,7 +1431,7 @@ void UnloadMusicStream(Music music) void PlayMusicStream(Music music) { #if USE_MINI_AL - AudioBuffer* audioBuffer = (AudioBuffer*)music->stream.audioBuffer; + AudioBuffer *audioBuffer = (AudioBuffer *)music->stream.audioBuffer; if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "PlayMusicStream() : No audio buffer"); @@ -1426,9 +1443,9 @@ void PlayMusicStream(Music music) // // just make sure to play again on window restore // if (IsMusicPlaying(music)) PlayMusicStream(music); mal_uint32 frameCursorPos = audioBuffer->frameCursorPos; - { - PlayAudioStream(music->stream); // <-- This resets the cursor position. - } + + PlayAudioStream(music->stream); // <-- This resets the cursor position. + audioBuffer->frameCursorPos = frameCursorPos; #else alSourcePlay(music->stream.source); @@ -1512,7 +1529,7 @@ void UpdateMusicStream(Music music) #if USE_MINI_AL bool streamEnding = false; - unsigned int subBufferSizeInFrames = ((AudioBuffer*)music->stream.audioBuffer)->bufferSizeInFrames / 2; + unsigned int subBufferSizeInFrames = ((AudioBuffer *)music->stream.audioBuffer)->bufferSizeInFrames/2; // NOTE: Using dynamic allocation because it could require more than 16KB void *pcm = calloc(subBufferSizeInFrames*music->stream.sampleSize/8*music->stream.channels, 1); @@ -1530,7 +1547,7 @@ void UpdateMusicStream(Music music) case MUSIC_AUDIO_OGG: { // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!) - int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, (short *)pcm, samplesCount*music->stream.channels); + stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, (short *)pcm, samplesCount*music->stream.channels); } break; #if defined(SUPPORT_FILEFORMAT_FLAC) @@ -1576,10 +1593,7 @@ void UpdateMusicStream(Music music) } else { - if (music->loopCount == -1) - { - PlayMusicStream(music); - } + if (music->loopCount == -1) PlayMusicStream(music); } } else @@ -1766,11 +1780,9 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un // The size of a streaming buffer must be at least double the size of a period. unsigned int periodSize = device.bufferSizeInFrames / device.periods; unsigned int subBufferSize = AUDIO_BUFFER_SIZE; - if (subBufferSize < periodSize) { - subBufferSize = periodSize; - } + if (subBufferSize < periodSize) subBufferSize = periodSize; - AudioBuffer* audioBuffer = CreateAudioBuffer(formatIn, stream.channels, stream.sampleRate, subBufferSize*2, AUDIO_BUFFER_USAGE_STREAM); + AudioBuffer *audioBuffer = CreateAudioBuffer(formatIn, stream.channels, stream.sampleRate, subBufferSize*2, AUDIO_BUFFER_USAGE_STREAM); if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "InitAudioStream() : Failed to create audio buffer"); @@ -1835,7 +1847,7 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un void CloseAudioStream(AudioStream stream) { #if USE_MINI_AL - DeleteAudioBuffer((AudioBuffer*)stream.audioBuffer); + DeleteAudioBuffer((AudioBuffer *)stream.audioBuffer); #else // Stop playing channel alSourceStop(stream.source); @@ -1866,7 +1878,7 @@ void CloseAudioStream(AudioStream stream) void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount) { #if USE_MINI_AL - AudioBuffer* audioBuffer = (AudioBuffer*)stream.audioBuffer; + AudioBuffer *audioBuffer = (AudioBuffer *)stream.audioBuffer; if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "UpdateAudioStream() : No audio buffer"); @@ -1889,23 +1901,22 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount) } mal_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2; - unsigned char *subBuffer = audioBuffer->buffer + ((subBufferSizeInFrames * stream.channels * (stream.sampleSize/8)) * subBufferToUpdate); + unsigned char *subBuffer = audioBuffer->buffer + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate); // 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 >= (mal_uint32)samplesCount) { mal_uint32 framesToWrite = subBufferSizeInFrames; - if (framesToWrite > (mal_uint32)samplesCount) { - framesToWrite = (mal_uint32)samplesCount; - } + if (framesToWrite > (mal_uint32)samplesCount) framesToWrite = (mal_uint32)samplesCount; - mal_uint32 bytesToWrite = framesToWrite * stream.channels * (stream.sampleSize/8); + mal_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8); memcpy(subBuffer, data, bytesToWrite); // Any leftover frames should be filled with zeros. mal_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite; - if (leftoverFrameCount > 0) { - memset(subBuffer + bytesToWrite, 0, leftoverFrameCount * stream.channels * (stream.sampleSize/8)); + if (leftoverFrameCount > 0) + { + memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8)); } audioBuffer->isSubBufferProcessed[subBufferToUpdate] = false; @@ -1939,7 +1950,7 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount) bool IsAudioBufferProcessed(AudioStream stream) { #if USE_MINI_AL - AudioBuffer* audioBuffer = (AudioBuffer*)stream.audioBuffer; + AudioBuffer *audioBuffer = (AudioBuffer *)stream.audioBuffer; if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "IsAudioBufferProcessed() : No audio buffer"); @@ -1961,7 +1972,7 @@ bool IsAudioBufferProcessed(AudioStream stream) void PlayAudioStream(AudioStream stream) { #if USE_MINI_AL - PlayAudioBuffer((AudioBuffer*)stream.audioBuffer); + PlayAudioBuffer((AudioBuffer *)stream.audioBuffer); #else alSourcePlay(stream.source); #endif @@ -1971,7 +1982,7 @@ void PlayAudioStream(AudioStream stream) void PauseAudioStream(AudioStream stream) { #if USE_MINI_AL - PauseAudioBuffer((AudioBuffer*)stream.audioBuffer); + PauseAudioBuffer((AudioBuffer *)stream.audioBuffer); #else alSourcePause(stream.source); #endif @@ -1981,7 +1992,7 @@ void PauseAudioStream(AudioStream stream) void ResumeAudioStream(AudioStream stream) { #if USE_MINI_AL - ResumeAudioBuffer((AudioBuffer*)stream.audioBuffer); + ResumeAudioBuffer((AudioBuffer *)stream.audioBuffer); #else ALenum state; alGetSourcei(stream.source, AL_SOURCE_STATE, &state); @@ -1994,7 +2005,7 @@ void ResumeAudioStream(AudioStream stream) bool IsAudioStreamPlaying(AudioStream stream) { #if USE_MINI_AL - return IsAudioBufferPlaying((AudioBuffer*)stream.audioBuffer); + return IsAudioBufferPlaying((AudioBuffer *)stream.audioBuffer); #else bool playing = false; ALint state; @@ -2011,7 +2022,7 @@ bool IsAudioStreamPlaying(AudioStream stream) void StopAudioStream(AudioStream stream) { #if USE_MINI_AL - StopAudioBuffer((AudioBuffer*)stream.audioBuffer); + StopAudioBuffer((AudioBuffer *)stream.audioBuffer); #else alSourceStop(stream.source); #endif @@ -2020,7 +2031,7 @@ void StopAudioStream(AudioStream stream) void SetAudioStreamVolume(AudioStream stream, float volume) { #if USE_MINI_AL - SetAudioBufferVolume((AudioBuffer*)stream.audioBuffer, volume); + SetAudioBufferVolume((AudioBuffer *)stream.audioBuffer, volume); #else alSourcef(stream.source, AL_GAIN, volume); #endif @@ -2029,7 +2040,7 @@ void SetAudioStreamVolume(AudioStream stream, float volume) void SetAudioStreamPitch(AudioStream stream, float pitch) { #if USE_MINI_AL - SetAudioBufferPitch((AudioBuffer*)stream.audioBuffer, pitch); + SetAudioBufferPitch((AudioBuffer *)stream.audioBuffer, pitch); #else alSourcef(stream.source, AL_PITCH, pitch); #endif |