Rename variable
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parent
2f90318d30
commit
aa4f9a657a
4 changed files with 25 additions and 24 deletions
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@ -379,7 +379,7 @@ void UnloadSound(Sound sound)
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// Update sound buffer with new data
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// NOTE: data must match sound.format
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void UpdateSound(Sound sound, const void *data, int numSamples)
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void UpdateSound(Sound sound, const void *data, int samplesCount)
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{
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ALint sampleRate, sampleSize, channels;
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alGetBufferi(sound.buffer, AL_FREQUENCY, &sampleRate);
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@ -390,7 +390,7 @@ void UpdateSound(Sound sound, const void *data, int numSamples)
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TraceLog(DEBUG, "UpdateSound() : AL_BITS: %i", sampleSize);
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TraceLog(DEBUG, "UpdateSound() : AL_CHANNELS: %i", channels);
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unsigned int dataSize = numSamples*channels*sampleSize/8; // Size of data in bytes
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unsigned int dataSize = samplesCount*channels*sampleSize/8; // Size of data in bytes
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alSourceStop(sound.source); // Stop sound
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alSourcei(sound.source, AL_BUFFER, 0); // Unbind buffer from sound to update
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@ -757,6 +757,7 @@ void StopMusicStream(Music music)
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}
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// Update (re-fill) music buffers if data already processed
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// TODO: Make sure buffers are ready for update... check music state
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void UpdateMusicStream(Music music)
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{
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ALenum state;
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@ -773,13 +774,13 @@ void UpdateMusicStream(Music music)
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void *pcm = calloc(AUDIO_BUFFER_SIZE*music->stream.channels*music->stream.sampleSize/8, 1);
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int numBuffersToProcess = processed;
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int numSamples = 0; // Total size of data steamed in L+R samples for xm floats,
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// individual L or R for ogg shorts
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int samplesCount = 0; // Total size of data steamed in L+R samples for xm floats,
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//individual L or R for ogg shorts
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for (int i = 0; i < numBuffersToProcess; i++)
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{
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if (music->samplesLeft >= AUDIO_BUFFER_SIZE) numSamples = AUDIO_BUFFER_SIZE;
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else numSamples = music->samplesLeft;
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if (music->samplesLeft >= AUDIO_BUFFER_SIZE) samplesCount = AUDIO_BUFFER_SIZE;
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else samplesCount = music->samplesLeft;
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// TODO: Really don't like ctxType thingy...
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switch (music->ctxType)
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@ -787,22 +788,22 @@ void UpdateMusicStream(Music music)
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case MUSIC_AUDIO_OGG:
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{
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// NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
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int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, (short *)pcm, numSamples*music->stream.channels);
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int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, (short *)pcm, samplesCount*music->stream.channels);
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} break;
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case MUSIC_AUDIO_FLAC:
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{
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// NOTE: Returns the number of samples to process
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unsigned int numSamplesFlac = (unsigned int)drflac_read_s16(music->ctxFlac, numSamples*music->stream.channels, (short *)pcm);
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unsigned int numSamplesFlac = (unsigned int)drflac_read_s16(music->ctxFlac, samplesCount*music->stream.channels, (short *)pcm);
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} break;
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case MUSIC_MODULE_XM: jar_xm_generate_samples_16bit(music->ctxXm, pcm, numSamples); break;
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case MUSIC_MODULE_MOD: jar_mod_fillbuffer(&music->ctxMod, pcm, numSamples, 0); break;
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case MUSIC_MODULE_XM: jar_xm_generate_samples_16bit(music->ctxXm, pcm, samplesCount); break;
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case MUSIC_MODULE_MOD: jar_mod_fillbuffer(&music->ctxMod, pcm, samplesCount, 0); break;
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default: break;
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}
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UpdateAudioStream(music->stream, pcm, numSamples);
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music->samplesLeft -= numSamples;
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UpdateAudioStream(music->stream, pcm, samplesCount);
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music->samplesLeft -= samplesCount;
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if (music->samplesLeft <= 0)
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{
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@ -981,7 +982,7 @@ void CloseAudioStream(AudioStream stream)
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// Update audio stream buffers with data
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// NOTE: Only updates one buffer per call
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void UpdateAudioStream(AudioStream stream, const void *data, int numSamples)
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void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
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{
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ALuint buffer = 0;
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alSourceUnqueueBuffers(stream.source, 1, &buffer);
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@ -989,7 +990,7 @@ void UpdateAudioStream(AudioStream stream, const void *data, int numSamples)
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// Check if any buffer was available for unqueue
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if (alGetError() != AL_INVALID_VALUE)
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{
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alBufferData(buffer, stream.format, data, numSamples*stream.channels*stream.sampleSize/8, stream.sampleRate);
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alBufferData(buffer, stream.format, data, samplesCount*stream.channels*stream.sampleSize/8, stream.sampleRate);
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alSourceQueueBuffers(stream.source, 1, &buffer);
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}
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}
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