Rename variable

This commit is contained in:
Milan Nikolic 2017-02-10 11:43:11 +01:00
parent 2f90318d30
commit aa4f9a657a
4 changed files with 25 additions and 24 deletions

View file

@ -379,7 +379,7 @@ void UnloadSound(Sound sound)
// Update sound buffer with new data
// NOTE: data must match sound.format
void UpdateSound(Sound sound, const void *data, int numSamples)
void UpdateSound(Sound sound, const void *data, int samplesCount)
{
ALint sampleRate, sampleSize, channels;
alGetBufferi(sound.buffer, AL_FREQUENCY, &sampleRate);
@ -390,7 +390,7 @@ void UpdateSound(Sound sound, const void *data, int numSamples)
TraceLog(DEBUG, "UpdateSound() : AL_BITS: %i", sampleSize);
TraceLog(DEBUG, "UpdateSound() : AL_CHANNELS: %i", channels);
unsigned int dataSize = numSamples*channels*sampleSize/8; // Size of data in bytes
unsigned int dataSize = samplesCount*channels*sampleSize/8; // Size of data in bytes
alSourceStop(sound.source); // Stop sound
alSourcei(sound.source, AL_BUFFER, 0); // Unbind buffer from sound to update
@ -757,6 +757,7 @@ void StopMusicStream(Music music)
}
// Update (re-fill) music buffers if data already processed
// TODO: Make sure buffers are ready for update... check music state
void UpdateMusicStream(Music music)
{
ALenum state;
@ -773,13 +774,13 @@ void UpdateMusicStream(Music music)
void *pcm = calloc(AUDIO_BUFFER_SIZE*music->stream.channels*music->stream.sampleSize/8, 1);
int numBuffersToProcess = processed;
int numSamples = 0; // Total size of data steamed in L+R samples for xm floats,
// individual L or R for ogg shorts
int samplesCount = 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++)
{
if (music->samplesLeft >= AUDIO_BUFFER_SIZE) numSamples = AUDIO_BUFFER_SIZE;
else numSamples = music->samplesLeft;
if (music->samplesLeft >= AUDIO_BUFFER_SIZE) samplesCount = AUDIO_BUFFER_SIZE;
else samplesCount = music->samplesLeft;
// TODO: Really don't like ctxType thingy...
switch (music->ctxType)
@ -787,22 +788,22 @@ void UpdateMusicStream(Music music)
case MUSIC_AUDIO_OGG:
{
// NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, (short *)pcm, numSamples*music->stream.channels);
int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, (short *)pcm, samplesCount*music->stream.channels);
} break;
case MUSIC_AUDIO_FLAC:
{
// NOTE: Returns the number of samples to process
unsigned int numSamplesFlac = (unsigned int)drflac_read_s16(music->ctxFlac, numSamples*music->stream.channels, (short *)pcm);
unsigned int numSamplesFlac = (unsigned int)drflac_read_s16(music->ctxFlac, samplesCount*music->stream.channels, (short *)pcm);
} break;
case MUSIC_MODULE_XM: jar_xm_generate_samples_16bit(music->ctxXm, pcm, numSamples); break;
case MUSIC_MODULE_MOD: jar_mod_fillbuffer(&music->ctxMod, pcm, numSamples, 0); break;
case MUSIC_MODULE_XM: jar_xm_generate_samples_16bit(music->ctxXm, pcm, samplesCount); break;
case MUSIC_MODULE_MOD: jar_mod_fillbuffer(&music->ctxMod, pcm, samplesCount, 0); break;
default: break;
}
UpdateAudioStream(music->stream, pcm, numSamples);
music->samplesLeft -= numSamples;
UpdateAudioStream(music->stream, pcm, samplesCount);
music->samplesLeft -= samplesCount;
if (music->samplesLeft <= 0)
{
@ -981,7 +982,7 @@ void CloseAudioStream(AudioStream stream)
// Update audio stream buffers with data
// NOTE: Only updates one buffer per call
void UpdateAudioStream(AudioStream stream, const void *data, int numSamples)
void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
{
ALuint buffer = 0;
alSourceUnqueueBuffers(stream.source, 1, &buffer);
@ -989,7 +990,7 @@ void UpdateAudioStream(AudioStream stream, const void *data, int numSamples)
// Check if any buffer was available for unqueue
if (alGetError() != AL_INVALID_VALUE)
{
alBufferData(buffer, stream.format, data, numSamples*stream.channels*stream.sampleSize/8, stream.sampleRate);
alBufferData(buffer, stream.format, data, samplesCount*stream.channels*stream.sampleSize/8, stream.sampleRate);
alSourceQueueBuffers(stream.source, 1, &buffer);
}
}