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Remove trailing spaces

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Ray 2019-10-17 17:18:03 +02:00
parent e40c26dea5
commit b75511248d
13 changed files with 304 additions and 304 deletions
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74
src/raudio.c
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@ -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)
{