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