twix/raylib
1
0
Fork 0
mirror of https://github.com/raysan5/raylib.git synced 2025-12-25 10:22:33 -05:00
Code Issues Packages Projects Releases Wiki Activity

Removed useless spacing

Browse Source
This commit is contained in:
raysan5
2016-08-16 11:09:55 +02:00
parent 342b89c5b9
commit 959a228815
8 changed files with 598 additions and 598 deletions
Download Patch File Download Diff File Expand all files Collapse all files

74
src/audio.c
View File

@ -183,7 +183,7 @@ void CloseAudioDevice(void)
alcMakeContextCurrent(NULL);
alcDestroyContext(context);
alcCloseDevice(device);
TraceLog(INFO, "Audio device closed successfully");
}
@ -217,7 +217,7 @@ Sound LoadSound(char *fileName)
else TraceLog(WARNING, "[%s] Sound extension not recognized, it can't be loaded", fileName);
Sound sound = LoadSoundFromWave(wave);
// Sound is loaded, we can unload wave
UnloadWave(wave);
@ -233,7 +233,7 @@ Sound LoadSoundFromWave(Wave wave)
if (wave.data != NULL)
{
ALenum format = 0;
// The OpenAL format is worked out by looking at the number of channels and the sample size (bits per sample)
if (wave.channels == 1)
{
@ -256,7 +256,7 @@ Sound LoadSoundFromWave(Wave wave)
}
}
else TraceLog(WARNING, "Wave number of channels not supported: %i", wave.channels);
// Create an audio source
ALuint source;
alGenSources(1, &source); // Generate pointer to audio source
@ -271,7 +271,7 @@ Sound LoadSoundFromWave(Wave wave)
//----------------------------------------
ALuint buffer;
alGenBuffers(1, &buffer); // Generate pointer to buffer
unsigned int dataSize = wave.sampleCount*wave.sampleSize/8; // Size in bytes
// Upload sound data to buffer
@ -367,7 +367,7 @@ Sound LoadSoundFromRES(const char *rresName, int resId)
free(data);
sound = LoadSoundFromWave(wave);
// Sound is loaded, we can unload wave data
UnloadWave(wave);
}
@ -506,13 +506,13 @@ Music LoadMusicStream(char *fileName)
TraceLog(DEBUG, "[%s] OGG sample rate: %i", fileName, info.sample_rate);
TraceLog(DEBUG, "[%s] OGG channels: %i", fileName, info.channels);
TraceLog(DEBUG, "[%s] OGG memory required: %i", fileName, info.temp_memory_required);
}
}
else if (strcmp(GetExtension(fileName), "xm") == 0)
{
int result = jar_xm_create_context_from_file(&music->ctxXm, 48000, fileName);
if (!result) // XM context created successfully
{
jar_xm_set_max_loop_count(music->ctxXm, 0); // Set infinite number of loops
@ -523,7 +523,7 @@ Music LoadMusicStream(char *fileName)
music->samplesLeft = music->totalSamples;
music->ctxType = MUSIC_MODULE_XM;
music->loop = true;
TraceLog(DEBUG, "[%s] XM number of samples: %i", fileName, music->totalSamples);
TraceLog(DEBUG, "[%s] XM track length: %11.6f sec", fileName, (float)music->totalSamples/48000.0f);
}
@ -555,11 +555,11 @@ Music LoadMusicStream(char *fileName)
void UnloadMusicStream(Music music)
{
CloseAudioStream(music->stream);
if (music->ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close(music->ctxOgg);
else if (music->ctxType == MUSIC_MODULE_XM) jar_xm_free_context(music->ctxXm);
else if (music->ctxType == MUSIC_MODULE_MOD) jar_mod_unload(&music->ctxMod);
free(music);
}
@ -597,58 +597,58 @@ void UpdateMusicStream(Music music)
// Determine if music stream is ready to be written
alGetSourcei(music->stream.source, AL_BUFFERS_PROCESSED, &processed);
int numBuffersToProcess = processed;
if (processed > 0)
{
bool active = true;
short pcm[AUDIO_BUFFER_SIZE];
float pcmf[AUDIO_BUFFER_SIZE];
int numSamples = 0; // Total size of data steamed in L+R samples for xm floats,
int numSamples = 0; // Total size of data steamed in L+R samples for xm floats,
// individual L or R for ogg shorts
for (int i = 0; i < numBuffersToProcess; i++)
{
switch (music->ctxType)
{
case MUSIC_AUDIO_OGG:
case MUSIC_AUDIO_OGG:
{
if (music->samplesLeft >= AUDIO_BUFFER_SIZE) numSamples = AUDIO_BUFFER_SIZE;
else numSamples = music->samplesLeft;
// NOTE: Returns the number of samples to process (should be the same as numSamples -> it is)
int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, pcm, numSamples);
// TODO: Review stereo channels Ogg, not enough samples served!
UpdateAudioStream(music->stream, pcm, numSamplesOgg*music->stream.channels);
music->samplesLeft -= (numSamplesOgg*music->stream.channels);
} break;
case MUSIC_MODULE_XM:
case MUSIC_MODULE_XM:
{
if (music->samplesLeft >= AUDIO_BUFFER_SIZE/2) numSamples = AUDIO_BUFFER_SIZE/2;
else numSamples = music->samplesLeft;
// NOTE: Output buffer is 2*numsamples elements (left and right value for each sample)
jar_xm_generate_samples(music->ctxXm, pcmf, numSamples);
UpdateAudioStream(music->stream, pcmf, numSamples*2); // Using 32bit PCM data
music->samplesLeft -= numSamples;
//TraceLog(INFO, "Samples left: %i", music->samplesLeft);
} break;
case MUSIC_MODULE_MOD:
case MUSIC_MODULE_MOD:
{
if (music->samplesLeft >= AUDIO_BUFFER_SIZE/2) numSamples = AUDIO_BUFFER_SIZE/2;
else numSamples = music->samplesLeft;
// NOTE: Output buffer size is nbsample*channels (default: 48000Hz, 16bit, Stereo)
jar_mod_fillbuffer(&music->ctxMod, pcm, numSamples, 0);
jar_mod_fillbuffer(&music->ctxMod, pcm, numSamples, 0);
UpdateAudioStream(music->stream, pcm, numSamples*2);
music->samplesLeft -= numSamples;
} break;
default: break;
}
@ -659,15 +659,15 @@ void UpdateMusicStream(Music music)
break;
}
}
// Reset audio stream for looping
if (!active && music->loop)
{
// Restart music context (if required)
//if (music->ctxType == MUSIC_MODULE_XM)
//if (music->ctxType == MUSIC_MODULE_XM)
if (music->ctxType == MUSIC_MODULE_MOD) jar_mod_seek_start(&music->ctxMod);
else if (music->ctxType == MUSIC_AUDIO_OGG) stb_vorbis_seek_start(music->ctxOgg);
// Reset samples left to total samples
music->samplesLeft = music->totalSamples;
}
@ -713,7 +713,7 @@ void SetMusicPitch(Music music, float pitch)
float GetMusicTimeLength(Music music)
{
float totalSeconds = (float)music->totalSamples/music->stream.sampleRate;
return totalSeconds;
}
@ -732,7 +732,7 @@ float GetMusicTimePlayed(Music music)
AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels)
{
AudioStream stream = { 0 };
stream.sampleRate = sampleRate;
stream.sampleSize = sampleSize;
stream.channels = channels;
@ -791,7 +791,7 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un
}
alSourceQueueBuffers(stream.source, MAX_STREAM_BUFFERS, stream.buffers);
TraceLog(INFO, "[AUD ID %i] Audio stream loaded successfully", stream.source);
return stream;
@ -806,9 +806,9 @@ void CloseAudioStream(AudioStream stream)
// Flush out all queued buffers
int queued = 0;
alGetSourcei(stream.source, AL_BUFFERS_QUEUED, &queued);
ALuint buffer = 0;
while (queued > 0)
{
alSourceUnqueueBuffers(stream.source, 1, &buffer);
@ -818,7 +818,7 @@ void CloseAudioStream(AudioStream stream)
// Delete source and buffers
alDeleteSources(1, &stream.source);
alDeleteBuffers(MAX_STREAM_BUFFERS, stream.buffers);
TraceLog(INFO, "[AUD ID %i] Unloaded audio stream data", stream.source);
}
@ -828,14 +828,14 @@ void UpdateAudioStream(AudioStream stream, void *data, int numSamples)
{
ALuint buffer = 0;
alSourceUnqueueBuffers(stream.source, 1, &buffer);
// Check if any buffer was available for unqueue
if (alGetError() != AL_INVALID_VALUE)
{
if (stream.sampleSize == 8) alBufferData(buffer, stream.format, (unsigned char *)data, numSamples*sizeof(unsigned char), stream.sampleRate);
else if (stream.sampleSize == 16) alBufferData(buffer, stream.format, (short *)data, numSamples*sizeof(short), stream.sampleRate);
else if (stream.sampleSize == 32) alBufferData(buffer, stream.format, (float *)data, numSamples*sizeof(float), stream.sampleRate);
alSourceQueueBuffers(stream.source, 1, &buffer);
}
}