diff options
Diffstat (limited to 'src/raudio.c')
| -rw-r--r-- | src/raudio.c | 74 |
1 files changed, 37 insertions, 37 deletions
diff --git a/src/raudio.c b/src/raudio.c index 447e5080..68fa95c8 100644 --- a/src/raudio.c +++ b/src/raudio.c @@ -198,10 +198,10 @@ typedef enum { AUDIO_BUFFER_USAGE_STATIC = 0, AUDIO_BUFFER_USAGE_STREAM } AudioB // playback device depending on whether or not data is streamed struct rAudioBuffer { ma_pcm_converter dsp; // PCM data converter - + float volume; // Audio buffer volume float pitch; // Audio buffer pitch - + bool playing; // Audio buffer state: AUDIO_PLAYING bool paused; // Audio buffer state: AUDIO_PAUSED bool looping; // Audio buffer looping, always true for AudioStreams @@ -209,11 +209,11 @@ struct rAudioBuffer { bool isSubBufferProcessed[2]; // SubBuffer processed (virtual double buffer) unsigned int frameCursorPos; // Frame cursor position - unsigned int bufferSizeInFrames; // Total buffer size in frames + unsigned int bufferSizeInFrames; // Total buffer size in frames unsigned int totalFramesProcessed; // Total frames processed in this buffer (required for play timming) - + unsigned char *buffer; // Data buffer, on music stream keeps filling - + rAudioBuffer *next; // Next audio buffer on the list rAudioBuffer *prev; // Previous audio buffer on the list }; @@ -289,7 +289,7 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const if (!audioBuffer->playing || audioBuffer->paused) continue; ma_uint32 framesRead = 0; - + while (1) { if (framesRead > frameCount) @@ -302,7 +302,7 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const // Just read as much data as we can from the stream ma_uint32 framesToRead = (frameCount - framesRead); - + while (framesToRead > 0) { float tempBuffer[1024]; // 512 frames for stereo @@ -387,7 +387,7 @@ static ma_uint32 OnAudioBufferDSPRead(ma_pcm_converter *pDSP, void *pFramesOut, { if (framesRead >= frameCount) break; } - else + else { if (isSubBufferProcessed[currentSubBufferIndex]) break; } @@ -465,7 +465,7 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 fr static void InitAudioBufferPool() { // Dummy buffers - for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) + for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) { audioBufferPool[i] = InitAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, 0, AUDIO_BUFFER_USAGE_STATIC); } @@ -486,7 +486,7 @@ void InitAudioDevice(void) // Init audio context ma_context_config contextConfig = ma_context_config_init(); contextConfig.logCallback = OnLog; - + ma_result result = ma_context_init(NULL, 0, &contextConfig, &context); if (result != MA_SUCCESS) { @@ -589,7 +589,7 @@ AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam { AudioBuffer *audioBuffer = (AudioBuffer *)RL_CALLOC(1, sizeof(AudioBuffer)); audioBuffer->buffer = RL_CALLOC(bufferSizeInFrames*channels*ma_get_bytes_per_sample(format), 1); - + if (audioBuffer == NULL) { TraceLog(LOG_ERROR, "InitAudioBuffer() : Failed to allocate memory for audio buffer"); @@ -608,7 +608,7 @@ AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam dspConfig.onRead = OnAudioBufferDSPRead; // Callback on data reading dspConfig.pUserData = audioBuffer; // Audio data pointer dspConfig.allowDynamicSampleRate = true; // Required for pitch shifting - + ma_result result = ma_pcm_converter_init(&dspConfig, &audioBuffer->dsp); if (result != MA_SUCCESS) @@ -655,7 +655,7 @@ void CloseAudioBuffer(AudioBuffer *buffer) bool IsAudioBufferPlaying(AudioBuffer *buffer) { bool result = false; - + if (buffer != NULL) result = (buffer->playing && !buffer->paused); else TraceLog(LOG_ERROR, "IsAudioBufferPlaying() : No audio buffer"); @@ -698,7 +698,7 @@ void StopAudioBuffer(AudioBuffer *buffer) void PauseAudioBuffer(AudioBuffer *buffer) { if (buffer != NULL) buffer->paused = true; - else TraceLog(LOG_ERROR, "PauseAudioBuffer() : No audio buffer"); + else TraceLog(LOG_ERROR, "PauseAudioBuffer() : No audio buffer"); } // Resume an audio buffer @@ -722,8 +722,8 @@ void SetAudioBufferPitch(AudioBuffer *buffer, float pitch) { float pitchMul = pitch/buffer->pitch; - // Pitching is just an adjustment of the sample rate. - // Note that this changes the duration of the sound: + // Pitching is just an adjustment of the sample rate. + // Note that this changes the duration of the sound: // - higher pitches will make the sound faster // - lower pitches make it slower ma_uint32 newOutputSampleRate = (ma_uint32)((float)buffer->dsp.src.config.sampleRateOut/pitchMul); @@ -816,7 +816,7 @@ Sound LoadSoundFromWave(Wave wave) if (wave.data != NULL) { - // When using miniaudio we need to do our own mixing. + // When using miniaudio we need to do our own mixing. // To simplify this we need convert the format of each sound to be consistent with // the format used to open the playback device. We can do this two ways: // @@ -909,7 +909,7 @@ void ExportWaveAsCode(Wave wave, const char *fileName) int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8; FILE *txtFile = fopen(fileName, "wt"); - + if (txtFile != NULL) { fprintf(txtFile, "\n//////////////////////////////////////////////////////////////////////////////////\n"); @@ -967,7 +967,7 @@ void PlaySoundMulti(Sound sound) oldAge = audioBufferPoolChannels[i]; oldIndex = i; } - + if (!IsAudioBufferPlaying(audioBufferPool[i])) { index = i; @@ -979,17 +979,17 @@ void PlaySoundMulti(Sound sound) if (index == -1) { TraceLog(LOG_WARNING,"pool age %i ended a sound early no room in buffer pool", audioBufferPoolCounter); - + if (oldIndex == -1) { // Shouldn't be able to get here... but just in case something odd happens! TraceLog(LOG_ERROR,"sound buffer pool couldn't determine oldest buffer not playing sound"); - + return; } - + index = oldIndex; - + // Just in case... StopAudioBuffer(audioBufferPool[index]); } @@ -1000,7 +1000,7 @@ void PlaySoundMulti(Sound sound) audioBufferPoolChannels[index] = audioBufferPoolCounter; audioBufferPoolCounter++; - + audioBufferPool[index]->volume = sound.stream.buffer->volume; audioBufferPool[index]->pitch = sound.stream.buffer->pitch; audioBufferPool[index]->looping = sound.stream.buffer->looping; @@ -1023,12 +1023,12 @@ void StopSoundMulti(void) int GetSoundsPlaying(void) { int counter = 0; - + for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) { if (IsAudioBufferPlaying(audioBufferPool[i])) counter++; } - + return counter; } @@ -1211,7 +1211,7 @@ Music LoadMusicStream(const char *fileName) { drmp3 *ctxMp3 = RL_MALLOC(sizeof(drmp3)); music.ctxData = ctxMp3; - + int result = drmp3_init_file(ctxMp3, fileName, NULL); if (result > 0) @@ -1242,7 +1242,7 @@ Music LoadMusicStream(const char *fileName) music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm); music.loopCount = 0; // Infinite loop by default musicLoaded = true; - + music.ctxData = ctxXm; } } @@ -1252,7 +1252,7 @@ Music LoadMusicStream(const char *fileName) { jar_mod_context_t *ctxMod = RL_MALLOC(sizeof(jar_mod_context_t)); music.ctxData = ctxMod; - + jar_mod_init(ctxMod); int result = jar_mod_load_file(ctxMod, fileName); @@ -1335,7 +1335,7 @@ void PlayMusicStream(Music music) { // For music streams, we need to make sure we maintain the frame cursor position // This is a hack for this section of code in UpdateMusicStream() - // NOTE: In case window is minimized, music stream is stopped, just make sure to + // NOTE: In case window is minimized, music stream is stopped, just make sure to // play again on window restore: if (IsMusicPlaying(music)) PlayMusicStream(music); ma_uint32 frameCursorPos = audioBuffer->frameCursorPos; PlayAudioStream(music.stream); // WARNING: This resets the cursor position. @@ -1395,7 +1395,7 @@ void UpdateMusicStream(Music music) void *pcm = RL_CALLOC(subBufferSizeInFrames*music.stream.channels*music.stream.sampleSize/8, 1); int samplesCount = 0; // Total size of data streamed in L+R samples for xm floats, individual L or R for ogg shorts - + // TODO: Get the sampleLeft using totalFramesProcessed... but first, get total frames processed correctly... //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels; int sampleLeft = music.sampleCount - (music.stream.buffer->totalFramesProcessed*music.stream.channels); @@ -1449,7 +1449,7 @@ void UpdateMusicStream(Music music) } UpdateAudioStream(music.stream, pcm, samplesCount); - + if ((music.ctxType == MUSIC_MODULE_XM) || (music.ctxType == MUSIC_MODULE_MOD)) { if (samplesCount > 1) sampleLeft -= samplesCount/2; @@ -1549,11 +1549,11 @@ 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.playback.internalBufferSizeInFrames/device.playback.internalPeriods; unsigned int subBufferSize = AUDIO_BUFFER_SIZE; - + if (subBufferSize < periodSize) subBufferSize = periodSize; stream.buffer = InitAudioBuffer(formatIn, stream.channels, stream.sampleRate, subBufferSize*2, AUDIO_BUFFER_USAGE_STREAM); - + if (stream.buffer != NULL) { stream.buffer->looping = true; // Always loop for streaming buffers @@ -1578,7 +1578,7 @@ void CloseAudioStream(AudioStream stream) void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount) { AudioBuffer *audioBuffer = stream.buffer; - + if (audioBuffer != NULL) { if (audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1]) @@ -1587,7 +1587,7 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount) if (audioBuffer->isSubBufferProcessed[0] && audioBuffer->isSubBufferProcessed[1]) { - // Both buffers are available for updating. + // Both buffers are available for updating. // Update the first one and make sure the cursor is moved back to the front. subBufferToUpdate = 0; audioBuffer->frameCursorPos = 0; @@ -1604,7 +1604,7 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount) // TODO: Get total frames processed on this buffer... DOES NOT WORK. audioBuffer->totalFramesProcessed += subBufferSizeInFrames; - // Does this API expect a whole buffer to be updated in one go? + // Does this API expect a whole buffer to be updated in one go? // Assuming so, but if not will need to change this logic. if (subBufferSizeInFrames >= (ma_uint32)samplesCount/stream.channels) { |