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Ray San
2017-12-20 11:37:43 +01:00
parent b63ffcfa0f
commit 1320044e94
1 changed files with 135 additions and 145 deletions
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280
src/audio.c
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@ -227,7 +227,8 @@ void TraceLog(int msgType, const char *text, ...); // Show trace lo
typedef enum { AUDIO_BUFFER_USAGE_STATIC = 0, AUDIO_BUFFER_USAGE_STREAM } AudioBufferUsage; typedef enum { AUDIO_BUFFER_USAGE_STATIC = 0, AUDIO_BUFFER_USAGE_STREAM } AudioBufferUsage;
// Audio buffer structure // Audio buffer structure
typedef struct AudioBuffer { typedef struct AudioBuffer AudioBuffer;
struct AudioBuffer {
mal_dsp dsp; // For format conversion. mal_dsp dsp; // For format conversion.
float volume; float volume;
float pitch; float pitch;
@ -238,10 +239,10 @@ typedef struct AudioBuffer {
bool isSubBufferProcessed[2]; bool isSubBufferProcessed[2];
unsigned int frameCursorPos; unsigned int frameCursorPos;
unsigned int bufferSizeInFrames; unsigned int bufferSizeInFrames;
AudioBuffer* next; AudioBuffer *next;
AudioBuffer* prev; AudioBuffer *prev;
unsigned char buffer[1]; unsigned char buffer[1];
} AudioBuffer; };
void StopAudioBuffer(AudioBuffer *audioBuffer); void StopAudioBuffer(AudioBuffer *audioBuffer);
@ -250,48 +251,46 @@ static mal_device device;
static mal_bool32 isAudioInitialized = MAL_FALSE; static mal_bool32 isAudioInitialized = MAL_FALSE;
static float masterVolume = 1; static float masterVolume = 1;
static mal_mutex audioLock; static mal_mutex audioLock;
static AudioBuffer* firstAudioBuffer = NULL; // Audio buffers are tracked in a linked list. static AudioBuffer *firstAudioBuffer = NULL; // Audio buffers are tracked in a linked list.
static AudioBuffer* lastAudioBuffer = NULL; static AudioBuffer *lastAudioBuffer = NULL;
static void TrackAudioBuffer(AudioBuffer* audioBuffer) static void TrackAudioBuffer(AudioBuffer* audioBuffer)
{ {
mal_mutex_lock(&audioLock); mal_mutex_lock(&audioLock);
{ {
if (firstAudioBuffer == NULL) { if (firstAudioBuffer == NULL) firstAudioBuffer = audioBuffer;
firstAudioBuffer = audioBuffer; else
} else { {
lastAudioBuffer->next = audioBuffer; lastAudioBuffer->next = audioBuffer;
audioBuffer->prev = lastAudioBuffer; audioBuffer->prev = lastAudioBuffer;
} }
lastAudioBuffer = audioBuffer; lastAudioBuffer = audioBuffer;
} }
mal_mutex_unlock(&audioLock); mal_mutex_unlock(&audioLock);
} }
static void UntrackAudioBuffer(AudioBuffer* audioBuffer) static void UntrackAudioBuffer(AudioBuffer* audioBuffer)
{ {
mal_mutex_lock(&audioLock); mal_mutex_lock(&audioLock);
{ {
if (audioBuffer->prev == NULL) { if (audioBuffer->prev == NULL) firstAudioBuffer = audioBuffer->next;
firstAudioBuffer = audioBuffer->next; else audioBuffer->prev->next = audioBuffer->next;
} else {
audioBuffer->prev->next = audioBuffer->next;
}
if (audioBuffer->next == NULL) { if (audioBuffer->next == NULL) lastAudioBuffer = audioBuffer->prev;
lastAudioBuffer = audioBuffer->prev; else audioBuffer->next->prev = audioBuffer->prev;
} else {
audioBuffer->next->prev = audioBuffer->prev;
}
audioBuffer->prev = NULL; audioBuffer->prev = NULL;
audioBuffer->next = NULL; audioBuffer->next = NULL;
} }
mal_mutex_unlock(&audioLock); mal_mutex_unlock(&audioLock);
} }
static void OnLog_MAL(mal_context* pContext, mal_device* pDevice, const char* message) static void OnLog_MAL(mal_context *pContext, mal_device *pDevice, const char *message)
{ {
(void)pContext; (void)pContext;
(void)pDevice; (void)pDevice;
@ -303,17 +302,19 @@ static void OnLog_MAL(mal_context* pContext, mal_device* pDevice, const char* me
// framesOut is both an input and an output. It will be initially filled with zeros outside of this function. // 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) static void MixFrames(float* framesOut, const float* framesIn, mal_uint32 frameCount, float localVolume)
{ {
for (mal_uint32 iFrame = 0; iFrame < frameCount; ++iFrame) { for (mal_uint32 iFrame = 0; iFrame < frameCount; ++iFrame)
for (mal_uint32 iChannel = 0; iChannel < device.channels; ++iChannel) { {
float* frameOut = framesOut + (iFrame * device.channels); for (mal_uint32 iChannel = 0; iChannel < device.channels; ++iChannel)
const float* frameIn = framesIn + (iFrame * device.channels); {
float *frameOut = framesOut + (iFrame*device.channels);
const float *frameIn = framesIn + (iFrame*device.channels);
frameOut[iChannel] += frameIn[iChannel] * masterVolume * localVolume; frameOut[iChannel] += frameIn[iChannel]*masterVolume*localVolume;
} }
} }
} }
static mal_uint32 OnSendAudioDataToDevice(mal_device* pDevice, mal_uint32 frameCount, void* pFramesOut) static mal_uint32 OnSendAudioDataToDevice(mal_device *pDevice, mal_uint32 frameCount, void *pFramesOut)
{ {
// This is where all of the mixing takes place. // This is where all of the mixing takes place.
(void)pDevice; (void)pDevice;
@ -328,27 +329,28 @@ static mal_uint32 OnSendAudioDataToDevice(mal_device* pDevice, mal_uint32 frameC
for (AudioBuffer* audioBuffer = firstAudioBuffer; audioBuffer != NULL; audioBuffer = audioBuffer->next) for (AudioBuffer* audioBuffer = firstAudioBuffer; audioBuffer != NULL; audioBuffer = audioBuffer->next)
{ {
// Ignore stopped or paused sounds. // Ignore stopped or paused sounds.
if (!audioBuffer->playing || audioBuffer->paused) { if (!audioBuffer->playing || audioBuffer->paused) continue;
continue;
}
mal_uint32 framesRead = 0; mal_uint32 framesRead = 0;
for (;;) { for (;;)
if (framesRead > frameCount) { {
if (framesRead > frameCount)
{
TraceLog(LOG_DEBUG, "Mixed too many frames from audio buffer"); TraceLog(LOG_DEBUG, "Mixed too many frames from audio buffer");
break; break;
} }
if (framesRead == frameCount) {
break; if (framesRead == frameCount) break;
}
// Just read as much data as we can from the stream. // Just read as much data as we can from the stream.
mal_uint32 framesToRead = (frameCount - framesRead); mal_uint32 framesToRead = (frameCount - framesRead);
while (framesToRead > 0) { while (framesToRead > 0)
{
float tempBuffer[1024]; // 512 frames for stereo. float tempBuffer[1024]; // 512 frames for stereo.
mal_uint32 framesToReadRightNow = framesToRead; 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; framesToReadRightNow = sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS;
} }
@ -357,9 +359,10 @@ static mal_uint32 OnSendAudioDataToDevice(mal_device* pDevice, mal_uint32 frameC
mal_bool32 flushDSP = !audioBuffer->looping; mal_bool32 flushDSP = !audioBuffer->looping;
mal_uint32 framesJustRead = mal_dsp_read_frames_ex(&audioBuffer->dsp, framesToReadRightNow, tempBuffer, flushDSP); mal_uint32 framesJustRead = mal_dsp_read_frames_ex(&audioBuffer->dsp, framesToReadRightNow, tempBuffer, flushDSP);
if (framesJustRead > 0) { if (framesJustRead > 0)
float* framesOut = (float*)pFramesOut + (framesRead * device.channels); {
float* framesIn = tempBuffer; float *framesOut = (float *)pFramesOut + (framesRead*device.channels);
float *framesIn = tempBuffer;
MixFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume); MixFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume);
framesToRead -= framesJustRead; framesToRead -= framesJustRead;
@ -367,11 +370,15 @@ static mal_uint32 OnSendAudioDataToDevice(mal_device* pDevice, mal_uint32 frameC
} }
// If we weren't able to read all the frames we requested, break. // If we weren't able to read all the frames we requested, break.
if (framesJustRead < framesToReadRightNow) { if (framesJustRead < framesToReadRightNow)
if (!audioBuffer->looping) { {
if (!audioBuffer->looping)
{
StopAudioBuffer(audioBuffer); StopAudioBuffer(audioBuffer);
break; break;
} else { }
else
{
// Should never get here, but just for safety, move the cursor position back to the start and continue the loop. // Should never get here, but just for safety, move the cursor position back to the start and continue the loop.
audioBuffer->frameCursorPos = 0; audioBuffer->frameCursorPos = 0;
continue; continue;
@ -381,12 +388,11 @@ static mal_uint32 OnSendAudioDataToDevice(mal_device* pDevice, mal_uint32 frameC
// If for some reason we weren't able to read every frame we'll need to break from the loop. Not doing this could // 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. // theoretically put us into an infinite loop.
if (framesToRead > 0) { if (framesToRead > 0) break;
break;
}
} }
} }
} }
mal_mutex_unlock(&audioLock); mal_mutex_unlock(&audioLock);
return frameCount; // We always output the same number of frames that were originally requested. return frameCount; // We always output the same number of frames that were originally requested.
@ -488,7 +494,8 @@ void InitAudioDevice(void)
void CloseAudioDevice(void) void CloseAudioDevice(void)
{ {
#if USE_MINI_AL #if USE_MINI_AL
if (!isAudioInitialized) { if (!isAudioInitialized)
{
TraceLog(LOG_WARNING, "Could not close audio device because it is not currently initialized"); TraceLog(LOG_WARNING, "Could not close audio device because it is not currently initialized");
return; return;
} }
@ -551,11 +558,13 @@ void SetMasterVolume(float volume)
#if USE_MINI_AL #if USE_MINI_AL
static mal_uint32 AudioBuffer_OnDSPRead(mal_dsp* pDSP, mal_uint32 frameCount, void* pFramesOut, void* pUserData) static mal_uint32 AudioBuffer_OnDSPRead(mal_dsp* pDSP, mal_uint32 frameCount, void* pFramesOut, void* pUserData)
{ {
AudioBuffer* audioBuffer = (AudioBuffer*)pUserData; AudioBuffer *audioBuffer = (AudioBuffer *)pUserData;
mal_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames / 2; mal_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2;
mal_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos / subBufferSizeInFrames; mal_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos/subBufferSizeInFrames;
if (currentSubBufferIndex > 1) {
if (currentSubBufferIndex > 1)
{
TraceLog(LOG_DEBUG, "Frame cursor position moved too far forward in audio stream"); TraceLog(LOG_DEBUG, "Frame cursor position moved too far forward in audio stream");
return 0; return 0;
} }
@ -565,7 +574,7 @@ static mal_uint32 AudioBuffer_OnDSPRead(mal_dsp* pDSP, mal_uint32 frameCount, vo
isSubBufferProcessed[0] = audioBuffer->isSubBufferProcessed[0]; isSubBufferProcessed[0] = audioBuffer->isSubBufferProcessed[0];
isSubBufferProcessed[1] = audioBuffer->isSubBufferProcessed[1]; isSubBufferProcessed[1] = audioBuffer->isSubBufferProcessed[1];
mal_uint32 frameSizeInBytes = mal_get_sample_size_in_bytes(audioBuffer->dsp.config.formatIn) * audioBuffer->dsp.config.channelsIn; 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. // Fill out every frame until we find a buffer that's marked as processed. Then fill the remainder with 0.
mal_uint32 framesRead = 0; mal_uint32 framesRead = 0;
@ -573,49 +582,47 @@ static mal_uint32 AudioBuffer_OnDSPRead(mal_dsp* pDSP, mal_uint32 frameCount, vo
{ {
// 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 // 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. // 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 (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
if (framesRead >= frameCount) { {
break; if (framesRead >= frameCount) break;
} }
} else { else
if (isSubBufferProcessed[currentSubBufferIndex]) { {
break; if (isSubBufferProcessed[currentSubBufferIndex]) break;
}
} }
mal_uint32 totalFramesRemaining = (frameCount - framesRead); mal_uint32 totalFramesRemaining = (frameCount - framesRead);
if (totalFramesRemaining == 0) { if (totalFramesRemaining == 0) break;
break;
}
mal_uint32 framesRemainingInOutputBuffer; mal_uint32 framesRemainingInOutputBuffer;
if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC) { if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
{
framesRemainingInOutputBuffer = audioBuffer->bufferSizeInFrames - audioBuffer->frameCursorPos; framesRemainingInOutputBuffer = audioBuffer->bufferSizeInFrames - audioBuffer->frameCursorPos;
} else { }
else
{
mal_uint32 firstFrameIndexOfThisSubBuffer = subBufferSizeInFrames * currentSubBufferIndex; mal_uint32 firstFrameIndexOfThisSubBuffer = subBufferSizeInFrames * currentSubBufferIndex;
framesRemainingInOutputBuffer = subBufferSizeInFrames - (audioBuffer->frameCursorPos - firstFrameIndexOfThisSubBuffer); framesRemainingInOutputBuffer = subBufferSizeInFrames - (audioBuffer->frameCursorPos - firstFrameIndexOfThisSubBuffer);
} }
mal_uint32 framesToRead = totalFramesRemaining; mal_uint32 framesToRead = totalFramesRemaining;
if (framesToRead > framesRemainingInOutputBuffer) { if (framesToRead > framesRemainingInOutputBuffer) framesToRead = framesRemainingInOutputBuffer;
framesToRead = framesRemainingInOutputBuffer;
}
memcpy((unsigned char*)pFramesOut + (framesRead*frameSizeInBytes), audioBuffer->buffer + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes); memcpy((unsigned char *)pFramesOut + (framesRead*frameSizeInBytes), audioBuffer->buffer + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes);
audioBuffer->frameCursorPos = (audioBuffer->frameCursorPos + framesToRead) % audioBuffer->bufferSizeInFrames; audioBuffer->frameCursorPos = (audioBuffer->frameCursorPos + framesToRead) % audioBuffer->bufferSizeInFrames;
framesRead += framesToRead; framesRead += framesToRead;
// If we've read to the end of the buffer, mark it as processed. // If we've read to the end of the buffer, mark it as processed.
if (framesToRead == framesRemainingInOutputBuffer) { if (framesToRead == framesRemainingInOutputBuffer)
{
audioBuffer->isSubBufferProcessed[currentSubBufferIndex] = true; audioBuffer->isSubBufferProcessed[currentSubBufferIndex] = true;
isSubBufferProcessed[currentSubBufferIndex] = true; isSubBufferProcessed[currentSubBufferIndex] = true;
currentSubBufferIndex = (currentSubBufferIndex + 1) % 2; currentSubBufferIndex = (currentSubBufferIndex + 1) % 2;
// We need to break from this loop if we're not looping. // We need to break from this loop if we're not looping.
if (!audioBuffer->looping) { if (!audioBuffer->looping)
{
StopAudioBuffer(audioBuffer); StopAudioBuffer(audioBuffer);
break; break;
} }
@ -624,24 +631,23 @@ static mal_uint32 AudioBuffer_OnDSPRead(mal_dsp* pDSP, mal_uint32 frameCount, vo
// Zero-fill excess. // Zero-fill excess.
mal_uint32 totalFramesRemaining = (frameCount - framesRead); mal_uint32 totalFramesRemaining = (frameCount - framesRead);
if (totalFramesRemaining > 0) { if (totalFramesRemaining > 0)
{
memset((unsigned char*)pFramesOut + (framesRead*frameSizeInBytes), 0, totalFramesRemaining*frameSizeInBytes); 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 // 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 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. // to know whether or not a non-looping sound has finished playback.
if (audioBuffer->usage != AUDIO_BUFFER_USAGE_STATIC) { if (audioBuffer->usage != AUDIO_BUFFER_USAGE_STATIC) framesRead += totalFramesRemaining;
framesRead += totalFramesRemaining;
}
} }
return framesRead; return framesRead;
} }
// Create a new audio buffer. Initially filled with silence. // 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 *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) if (audioBuffer == NULL)
{ {
TraceLog(LOG_ERROR, "CreateAudioBuffer() : Failed to allocate memory for audio buffer"); TraceLog(LOG_ERROR, "CreateAudioBuffer() : Failed to allocate memory for audio buffer");
@ -658,7 +664,8 @@ AudioBuffer* CreateAudioBuffer(mal_format format, mal_uint32 channels, mal_uint3
dspConfig.sampleRateIn = sampleRate; dspConfig.sampleRateIn = sampleRate;
dspConfig.sampleRateOut = DEVICE_SAMPLE_RATE; dspConfig.sampleRateOut = DEVICE_SAMPLE_RATE;
mal_result resultMAL = mal_dsp_init(&dspConfig, AudioBuffer_OnDSPRead, audioBuffer, &audioBuffer->dsp); mal_result resultMAL = mal_dsp_init(&dspConfig, AudioBuffer_OnDSPRead, audioBuffer, &audioBuffer->dsp);
if (resultMAL != MAL_SUCCESS) { if (resultMAL != MAL_SUCCESS)
{
TraceLog(LOG_ERROR, "LoadSoundFromWave() : Failed to create data conversion pipeline"); TraceLog(LOG_ERROR, "LoadSoundFromWave() : Failed to create data conversion pipeline");
free(audioBuffer); free(audioBuffer);
return NULL; return NULL;
@ -683,7 +690,7 @@ AudioBuffer* CreateAudioBuffer(mal_format format, mal_uint32 channels, mal_uint3
} }
// Delete an audio buffer. // Delete an audio buffer.
void DeleteAudioBuffer(AudioBuffer* audioBuffer) void DeleteAudioBuffer(AudioBuffer *audioBuffer)
{ {
if (audioBuffer == NULL) if (audioBuffer == NULL)
{ {
@ -696,7 +703,7 @@ void DeleteAudioBuffer(AudioBuffer* audioBuffer)
} }
// Check if an audio buffer is playing. // Check if an audio buffer is playing.
bool IsAudioBufferPlaying(AudioBuffer* audioBuffer) bool IsAudioBufferPlaying(AudioBuffer *audioBuffer)
{ {
if (audioBuffer == NULL) if (audioBuffer == NULL)
{ {
@ -711,7 +718,7 @@ bool IsAudioBufferPlaying(AudioBuffer* audioBuffer)
// //
// This will restart the buffer from the start. Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position // This will restart the buffer from the start. Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position
// should be maintained. // should be maintained.
void PlayAudioBuffer(AudioBuffer* audioBuffer) void PlayAudioBuffer(AudioBuffer *audioBuffer)
{ {
if (audioBuffer == NULL) if (audioBuffer == NULL)
{ {
@ -725,7 +732,7 @@ void PlayAudioBuffer(AudioBuffer* audioBuffer)
} }
// Stop an audio buffer. // Stop an audio buffer.
void StopAudioBuffer(AudioBuffer* audioBuffer) void StopAudioBuffer(AudioBuffer *audioBuffer)
{ {
if (audioBuffer == NULL) if (audioBuffer == NULL)
{ {
@ -734,10 +741,7 @@ void StopAudioBuffer(AudioBuffer* audioBuffer)
} }
// Don't do anything if the audio buffer is already stopped. // Don't do anything if the audio buffer is already stopped.
if (!IsAudioBufferPlaying(audioBuffer)) if (!IsAudioBufferPlaying(audioBuffer)) return;
{
return;
}
audioBuffer->playing = false; audioBuffer->playing = false;
audioBuffer->paused = false; audioBuffer->paused = false;
@ -747,7 +751,7 @@ void StopAudioBuffer(AudioBuffer* audioBuffer)
} }
// Pause an audio buffer. // Pause an audio buffer.
void PauseAudioBuffer(AudioBuffer* audioBuffer) void PauseAudioBuffer(AudioBuffer *audioBuffer)
{ {
if (audioBuffer == NULL) if (audioBuffer == NULL)
{ {
@ -759,7 +763,7 @@ void PauseAudioBuffer(AudioBuffer* audioBuffer)
} }
// Resume an audio buffer. // Resume an audio buffer.
void ResumeAudioBuffer(AudioBuffer* audioBuffer) void ResumeAudioBuffer(AudioBuffer *audioBuffer)
{ {
if (audioBuffer == NULL) if (audioBuffer == NULL)
{ {
@ -771,7 +775,7 @@ void ResumeAudioBuffer(AudioBuffer* audioBuffer)
} }
// Set volume for an audio buffer. // Set volume for an audio buffer.
void SetAudioBufferVolume(AudioBuffer* audioBuffer, float volume) void SetAudioBufferVolume(AudioBuffer *audioBuffer, float volume)
{ {
if (audioBuffer == NULL) if (audioBuffer == NULL)
{ {
@ -783,7 +787,7 @@ void SetAudioBufferVolume(AudioBuffer* audioBuffer, float volume)
} }
// Set pitch for an audio buffer. // Set pitch for an audio buffer.
void SetAudioBufferPitch(AudioBuffer* audioBuffer, float pitch) void SetAudioBufferPitch(AudioBuffer *audioBuffer, float pitch)
{ {
if (audioBuffer == NULL) if (audioBuffer == NULL)
{ {
@ -874,23 +878,13 @@ 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 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); mal_uint32 frameCount = mal_convert_frames(NULL, DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, NULL, formatIn, wave.channels, wave.sampleRate, frameCountIn);
if (frameCount == 0) { if (frameCount == 0) TraceLog(LOG_ERROR, "LoadSoundFromWave() : Failed to get frame count for format conversion");
TraceLog(LOG_ERROR, "LoadSoundFromWave() : Failed to get frame count for format conversion");
}
AudioBuffer* audioBuffer = CreateAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC); AudioBuffer* audioBuffer = CreateAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC);
if (audioBuffer == NULL) if (audioBuffer == NULL) TraceLog(LOG_ERROR, "LoadSoundFromWave() : Failed to create audio buffer");
{
TraceLog(LOG_ERROR, "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); 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) if (frameCount == 0) TraceLog(LOG_ERROR, "LoadSoundFromWave() : Format conversion failed");
{
TraceLog(LOG_ERROR, "LoadSoundFromWave() : Format conversion failed");
}
sound.audioBuffer = audioBuffer; sound.audioBuffer = audioBuffer;
#else #else
@ -965,7 +959,7 @@ void UnloadWave(Wave wave)
void UnloadSound(Sound sound) void UnloadSound(Sound sound)
{ {
#if USE_MINI_AL #if USE_MINI_AL
DeleteAudioBuffer((AudioBuffer*)sound.audioBuffer); DeleteAudioBuffer((AudioBuffer *)sound.audioBuffer);
#else #else
alSourceStop(sound.source); alSourceStop(sound.source);
@ -981,7 +975,7 @@ void UnloadSound(Sound sound)
void UpdateSound(Sound sound, const void *data, int samplesCount) void UpdateSound(Sound sound, const void *data, int samplesCount)
{ {
#if USE_MINI_AL #if USE_MINI_AL
AudioBuffer* audioBuffer = (AudioBuffer*)sound.audioBuffer; AudioBuffer *audioBuffer = (AudioBuffer *)sound.audioBuffer;
if (audioBuffer == NULL) if (audioBuffer == NULL)
{ {
TraceLog(LOG_ERROR, "UpdateSound() : Invalid sound - no audio buffer"); TraceLog(LOG_ERROR, "UpdateSound() : Invalid sound - no audio buffer");
@ -1021,7 +1015,7 @@ void UpdateSound(Sound sound, const void *data, int samplesCount)
void PlaySound(Sound sound) void PlaySound(Sound sound)
{ {
#if USE_MINI_AL #if USE_MINI_AL
PlayAudioBuffer((AudioBuffer*)sound.audioBuffer); PlayAudioBuffer((AudioBuffer *)sound.audioBuffer);
#else #else
alSourcePlay(sound.source); // Play the sound alSourcePlay(sound.source); // Play the sound
#endif #endif
@ -1046,7 +1040,7 @@ void PlaySound(Sound sound)
void PauseSound(Sound sound) void PauseSound(Sound sound)
{ {
#if USE_MINI_AL #if USE_MINI_AL
PauseAudioBuffer((AudioBuffer*)sound.audioBuffer); PauseAudioBuffer((AudioBuffer *)sound.audioBuffer);
#else #else
alSourcePause(sound.source); alSourcePause(sound.source);
#endif #endif
@ -1056,7 +1050,7 @@ void PauseSound(Sound sound)
void ResumeSound(Sound sound) void ResumeSound(Sound sound)
{ {
#if USE_MINI_AL #if USE_MINI_AL
ResumeAudioBuffer((AudioBuffer*)sound.audioBuffer); ResumeAudioBuffer((AudioBuffer *)sound.audioBuffer);
#else #else
ALenum state; ALenum state;
@ -1070,7 +1064,7 @@ void ResumeSound(Sound sound)
void StopSound(Sound sound) void StopSound(Sound sound)
{ {
#if USE_MINI_AL #if USE_MINI_AL
StopAudioBuffer((AudioBuffer*)sound.audioBuffer); StopAudioBuffer((AudioBuffer *)sound.audioBuffer);
#else #else
alSourceStop(sound.source); alSourceStop(sound.source);
#endif #endif
@ -1080,7 +1074,7 @@ void StopSound(Sound sound)
bool IsSoundPlaying(Sound sound) bool IsSoundPlaying(Sound sound)
{ {
#if USE_MINI_AL #if USE_MINI_AL
return IsAudioBufferPlaying((AudioBuffer*)sound.audioBuffer); return IsAudioBufferPlaying((AudioBuffer *)sound.audioBuffer);
#else #else
bool playing = false; bool playing = false;
ALint state; ALint state;
@ -1096,7 +1090,7 @@ bool IsSoundPlaying(Sound sound)
void SetSoundVolume(Sound sound, float volume) void SetSoundVolume(Sound sound, float volume)
{ {
#if USE_MINI_AL #if USE_MINI_AL
SetAudioBufferVolume((AudioBuffer*)sound.audioBuffer, volume); SetAudioBufferVolume((AudioBuffer *)sound.audioBuffer, volume);
#else #else
alSourcef(sound.source, AL_GAIN, volume); alSourcef(sound.source, AL_GAIN, volume);
#endif #endif
@ -1106,7 +1100,7 @@ void SetSoundVolume(Sound sound, float volume)
void SetSoundPitch(Sound sound, float pitch) void SetSoundPitch(Sound sound, float pitch)
{ {
#if USE_MINI_AL #if USE_MINI_AL
SetAudioBufferPitch((AudioBuffer*)sound.audioBuffer, pitch); SetAudioBufferPitch((AudioBuffer *)sound.audioBuffer, pitch);
#else #else
alSourcef(sound.source, AL_PITCH, pitch); alSourcef(sound.source, AL_PITCH, pitch);
#endif #endif
@ -1121,15 +1115,17 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
mal_uint32 frameCountIn = wave->sampleCount; // Is wave->sampleCount actually the frame count? That terminology needs to change, if so. 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); 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."); TraceLog(LOG_ERROR, "WaveFormat() : Failed to get frame count for format conversion.");
return; 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); 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."); TraceLog(LOG_ERROR, "WaveFormat() : Format conversion failed.");
return; return;
} }
@ -1404,7 +1400,7 @@ void UnloadMusicStream(Music music)
void PlayMusicStream(Music music) void PlayMusicStream(Music music)
{ {
#if USE_MINI_AL #if USE_MINI_AL
AudioBuffer* audioBuffer = (AudioBuffer*)music->stream.audioBuffer; AudioBuffer *audioBuffer = (AudioBuffer *)music->stream.audioBuffer;
if (audioBuffer == NULL) if (audioBuffer == NULL)
{ {
TraceLog(LOG_ERROR, "PlayMusicStream() : No audio buffer"); TraceLog(LOG_ERROR, "PlayMusicStream() : No audio buffer");
@ -1416,9 +1412,9 @@ void PlayMusicStream(Music music)
// // just make sure to play again on window restore // // just make sure to play again on window restore
// if (IsMusicPlaying(music)) PlayMusicStream(music); // if (IsMusicPlaying(music)) PlayMusicStream(music);
mal_uint32 frameCursorPos = audioBuffer->frameCursorPos; mal_uint32 frameCursorPos = audioBuffer->frameCursorPos;
{
PlayAudioStream(music->stream); // <-- This resets the cursor position. PlayAudioStream(music->stream); // <-- This resets the cursor position.
}
audioBuffer->frameCursorPos = frameCursorPos; audioBuffer->frameCursorPos = frameCursorPos;
#else #else
alSourcePlay(music->stream.source); alSourcePlay(music->stream.source);
@ -1502,7 +1498,7 @@ void UpdateMusicStream(Music music)
#if USE_MINI_AL #if USE_MINI_AL
bool streamEnding = false; 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 // NOTE: Using dynamic allocation because it could require more than 16KB
void *pcm = calloc(subBufferSizeInFrames*music->stream.sampleSize/8*music->stream.channels, 1); void *pcm = calloc(subBufferSizeInFrames*music->stream.sampleSize/8*music->stream.channels, 1);
@ -1566,10 +1562,7 @@ void UpdateMusicStream(Music music)
} }
else else
{ {
if (music->loopCount == -1) if (music->loopCount == -1) PlayMusicStream(music);
{
PlayMusicStream(music);
}
} }
} }
else else
@ -1756,11 +1749,9 @@ 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. // 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 periodSize = device.bufferSizeInFrames / device.periods;
unsigned int subBufferSize = AUDIO_BUFFER_SIZE; unsigned int subBufferSize = AUDIO_BUFFER_SIZE;
if (subBufferSize < periodSize) { if (subBufferSize < periodSize) 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) if (audioBuffer == NULL)
{ {
TraceLog(LOG_ERROR, "InitAudioStream() : Failed to create audio buffer"); TraceLog(LOG_ERROR, "InitAudioStream() : Failed to create audio buffer");
@ -1825,7 +1816,7 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un
void CloseAudioStream(AudioStream stream) void CloseAudioStream(AudioStream stream)
{ {
#if USE_MINI_AL #if USE_MINI_AL
DeleteAudioBuffer((AudioBuffer*)stream.audioBuffer); DeleteAudioBuffer((AudioBuffer *)stream.audioBuffer);
#else #else
// Stop playing channel // Stop playing channel
alSourceStop(stream.source); alSourceStop(stream.source);
@ -1879,23 +1870,22 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
} }
mal_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2; 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. // 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) if (subBufferSizeInFrames >= (mal_uint32)samplesCount)
{ {
mal_uint32 framesToWrite = subBufferSizeInFrames; mal_uint32 framesToWrite = subBufferSizeInFrames;
if (framesToWrite > (mal_uint32)samplesCount) { if (framesToWrite > (mal_uint32)samplesCount) 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); memcpy(subBuffer, data, bytesToWrite);
// Any leftover frames should be filled with zeros. // Any leftover frames should be filled with zeros.
mal_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite; mal_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite;
if (leftoverFrameCount > 0) { if (leftoverFrameCount > 0)
memset(subBuffer + bytesToWrite, 0, leftoverFrameCount * stream.channels * (stream.sampleSize/8)); {
memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8));
} }
audioBuffer->isSubBufferProcessed[subBufferToUpdate] = false; audioBuffer->isSubBufferProcessed[subBufferToUpdate] = false;
@ -1929,7 +1919,7 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
bool IsAudioBufferProcessed(AudioStream stream) bool IsAudioBufferProcessed(AudioStream stream)
{ {
#if USE_MINI_AL #if USE_MINI_AL
AudioBuffer* audioBuffer = (AudioBuffer*)stream.audioBuffer; AudioBuffer *audioBuffer = (AudioBuffer *)stream.audioBuffer;
if (audioBuffer == NULL) if (audioBuffer == NULL)
{ {
TraceLog(LOG_ERROR, "IsAudioBufferProcessed() : No audio buffer"); TraceLog(LOG_ERROR, "IsAudioBufferProcessed() : No audio buffer");
@ -1951,7 +1941,7 @@ bool IsAudioBufferProcessed(AudioStream stream)
void PlayAudioStream(AudioStream stream) void PlayAudioStream(AudioStream stream)
{ {
#if USE_MINI_AL #if USE_MINI_AL
PlayAudioBuffer((AudioBuffer*)stream.audioBuffer); PlayAudioBuffer((AudioBuffer *)stream.audioBuffer);
#else #else
alSourcePlay(stream.source); alSourcePlay(stream.source);
#endif #endif
@ -1961,7 +1951,7 @@ void PlayAudioStream(AudioStream stream)
void PauseAudioStream(AudioStream stream) void PauseAudioStream(AudioStream stream)
{ {
#if USE_MINI_AL #if USE_MINI_AL
PauseAudioBuffer((AudioBuffer*)stream.audioBuffer); PauseAudioBuffer((AudioBuffer *)stream.audioBuffer);
#else #else
alSourcePause(stream.source); alSourcePause(stream.source);
#endif #endif
@ -1971,7 +1961,7 @@ void PauseAudioStream(AudioStream stream)
void ResumeAudioStream(AudioStream stream) void ResumeAudioStream(AudioStream stream)
{ {
#if USE_MINI_AL #if USE_MINI_AL
ResumeAudioBuffer((AudioBuffer*)stream.audioBuffer); ResumeAudioBuffer((AudioBuffer *)stream.audioBuffer);
#else #else
ALenum state; ALenum state;
alGetSourcei(stream.source, AL_SOURCE_STATE, &state); alGetSourcei(stream.source, AL_SOURCE_STATE, &state);
@ -1984,7 +1974,7 @@ void ResumeAudioStream(AudioStream stream)
bool IsAudioStreamPlaying(AudioStream stream) bool IsAudioStreamPlaying(AudioStream stream)
{ {
#if USE_MINI_AL #if USE_MINI_AL
return IsAudioBufferPlaying((AudioBuffer*)stream.audioBuffer); return IsAudioBufferPlaying((AudioBuffer *)stream.audioBuffer);
#else #else
bool playing = false; bool playing = false;
ALint state; ALint state;
@ -2001,7 +1991,7 @@ bool IsAudioStreamPlaying(AudioStream stream)
void StopAudioStream(AudioStream stream) void StopAudioStream(AudioStream stream)
{ {
#if USE_MINI_AL #if USE_MINI_AL
StopAudioBuffer((AudioBuffer*)stream.audioBuffer); StopAudioBuffer((AudioBuffer *)stream.audioBuffer);
#else #else
alSourceStop(stream.source); alSourceStop(stream.source);
#endif #endif
@ -2010,7 +2000,7 @@ void StopAudioStream(AudioStream stream)
void SetAudioStreamVolume(AudioStream stream, float volume) void SetAudioStreamVolume(AudioStream stream, float volume)
{ {
#if USE_MINI_AL #if USE_MINI_AL
SetAudioBufferVolume((AudioBuffer*)stream.audioBuffer, volume); SetAudioBufferVolume((AudioBuffer *)stream.audioBuffer, volume);
#else #else
alSourcef(stream.source, AL_GAIN, volume); alSourcef(stream.source, AL_GAIN, volume);
#endif #endif
@ -2019,7 +2009,7 @@ void SetAudioStreamVolume(AudioStream stream, float volume)
void SetAudioStreamPitch(AudioStream stream, float pitch) void SetAudioStreamPitch(AudioStream stream, float pitch)
{ {
#if USE_MINI_AL #if USE_MINI_AL
SetAudioBufferPitch((AudioBuffer*)stream.audioBuffer, pitch); SetAudioBufferPitch((AudioBuffer *)stream.audioBuffer, pitch);
#else #else
alSourcef(stream.source, AL_PITCH, pitch); alSourcef(stream.source, AL_PITCH, pitch);
#endif #endif