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Added Sound parameters data

pull/927/head
Ray 5 anos atrás
pai
commit
543c0ba30d
1 arquivos alterados com 11 adições e 12 exclusões
  1. +11
    -12
      src/raudio.c

+ 11
- 12
src/raudio.c Ver arquivo

@ -471,6 +471,12 @@ static void InitAudioBufferPool()
}
}
// Close the audio buffers pool
static void CloseAudioBufferPool()
{
for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) RL_FREE(audioBufferPool[i]);
}
//----------------------------------------------------------------------------------
// Module Functions Definition - Audio Device initialization and Closing
//----------------------------------------------------------------------------------
@ -544,12 +550,6 @@ void InitAudioDevice(void)
isAudioInitialized = true;
}
// Close the audio buffers pool
static void CloseAudioBufferPool()
{
for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) RL_FREE(audioBufferPool[i]);
}
// Close the audio device for all contexts
void CloseAudioDevice(void)
{
@ -882,6 +882,10 @@ Sound LoadSoundFromWave(Wave wave)
frameCount = (ma_uint32)ma_convert_frames(audioBuffer->buffer, audioBuffer->dsp.formatConverterIn.config.formatIn, audioBuffer->dsp.formatConverterIn.config.channels, audioBuffer->dsp.src.config.sampleRateIn, wave.data, formatIn, wave.channels, wave.sampleRate, frameCountIn);
if (frameCount == 0) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Format conversion failed");
sound.sampleCount = frameCount*DEVICE_CHANNELS;
sound.stream.sampleRate = DEVICE_SAMPLE_RATE;
sound.stream.sampleSize = 32;
sound.stream.channels = DEVICE_CHANNELS;
sound.stream.buffer = audioBuffer;
}
@ -1277,9 +1281,6 @@ Music LoadMusicStream(const char *fileName)
TraceLog(LOG_INFO, "[%s] MP3 channels: %i", fileName, ctxMp3->channels);
music->stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
// TODO: There is not an easy way to compute the total number of samples available
// in an MP3, frames size could be variable... we tried with a 60 seconds music... but crashes...
music->sampleCount = drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels;
music->sampleLeft = music->sampleCount;
music->ctxType = MUSIC_AUDIO_MP3;
@ -1435,7 +1436,6 @@ void ResumeMusicStream(Music music)
}
// Stop music playing (close stream)
// TODO: To clear a buffer, make sure they have been already processed!
void StopMusicStream(Music music)
{
if (music == NULL) return;
@ -1467,7 +1467,6 @@ 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)
{
if (music == NULL) return;
@ -1486,7 +1485,6 @@ void UpdateMusicStream(Music music)
if ((music->sampleLeft/music->stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music->stream.channels;
else samplesCount = music->sampleLeft;
// TODO: Really don't like ctxType thingy...
switch (music->ctxType)
{
#if defined(SUPPORT_FILEFORMAT_OGG)
@ -1531,6 +1529,7 @@ void UpdateMusicStream(Music music)
}
UpdateAudioStream(music->stream, pcm, samplesCount);
if ((music->ctxType == MUSIC_MODULE_XM) || (music->ctxType == MUSIC_MODULE_MOD))
{
if (samplesCount > 1) music->sampleLeft -= samplesCount/2;

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