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Share PCM streaming buffer to reduce dynamic allocations (#2532)

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This commit is contained in:
Dan Bechard
2022-06-21 13:26:21 -04:00
committed by GitHub
parent 0afa79067b
commit 8bd3ecaa66
1 changed files with 19 additions and 14 deletions
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33
src/raudio.c
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@@ -351,6 +351,8 @@ typedef struct AudioData {
ma_device device; // miniaudio device
ma_mutex lock; // miniaudio mutex lock
bool isReady; // Check if audio device is ready
size_t pcmCapacity;
void *pcm;
} System;
struct {
AudioBuffer *first; // Pointer to first AudioBuffer in the list
@@ -510,6 +512,7 @@ void CloseAudioDevice(void)
ma_context_uninit(&AUDIO.System.context);
AUDIO.System.isReady = false;
RL_FREE(AUDIO.System.pcm);
TRACELOG(LOG_INFO, "AUDIO: Device closed successfully");
}
@@ -1726,7 +1729,12 @@ void UpdateMusicStream(Music music)
unsigned int subBufferSizeInFrames = music.stream.buffer->sizeInFrames/2;
// NOTE: Using dynamic allocation because it could require more than 16KB
void *pcm = RL_CALLOC(subBufferSizeInFrames*music.stream.channels*music.stream.sampleSize/8, 1);
size_t pcmSize = subBufferSizeInFrames * music.stream.channels * music.stream.sampleSize / 8;
if (AUDIO.System.pcmCapacity < pcmSize) {
RL_FREE(AUDIO.System.pcm);
AUDIO.System.pcm = RL_CALLOC(1, pcmSize);
AUDIO.System.pcmCapacity = pcmSize;
}
int frameCountToStream = 0; // Total size of data in frames to be streamed
@@ -1745,8 +1753,8 @@ void UpdateMusicStream(Music music)
case MUSIC_AUDIO_WAV:
{
// NOTE: Returns the number of samples to process (not required)
if (music.stream.sampleSize == 16) drwav_read_pcm_frames_s16((drwav *)music.ctxData, frameCountToStream, (short *)pcm);
else if (music.stream.sampleSize == 32) drwav_read_pcm_frames_f32((drwav *)music.ctxData, frameCountToStream, (float *)pcm);
if (music.stream.sampleSize == 16) drwav_read_pcm_frames_s16((drwav *)music.ctxData, frameCountToStream, (short *)AUDIO.System.pcm);
else if (music.stream.sampleSize == 32) drwav_read_pcm_frames_f32((drwav *)music.ctxData, frameCountToStream, (float *)AUDIO.System.pcm);
} break;
#endif
@@ -1754,7 +1762,7 @@ void UpdateMusicStream(Music music)
case MUSIC_AUDIO_OGG:
{
// NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)pcm, frameCountToStream*music.stream.channels);
stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)AUDIO.System.pcm, frameCountToStream*music.stream.channels);
} break;
#endif
@@ -1762,14 +1770,14 @@ void UpdateMusicStream(Music music)
case MUSIC_AUDIO_FLAC:
{
// NOTE: Returns the number of samples to process (not required)
drflac_read_pcm_frames_s16((drflac *)music.ctxData, frameCountToStream*music.stream.channels, (short *)pcm);
drflac_read_pcm_frames_s16((drflac *)music.ctxData, frameCountToStream*music.stream.channels, (short *)AUDIO.System.pcm);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3:
{
drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, frameCountToStream, (float *)pcm);
drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, frameCountToStream, (float *)AUDIO.System.pcm);
} break;
#endif
@@ -1777,9 +1785,9 @@ void UpdateMusicStream(Music music)
case MUSIC_MODULE_XM:
{
// NOTE: Internally we consider 2 channels generation, so sampleCount/2
if (AUDIO_DEVICE_FORMAT == ma_format_f32) jar_xm_generate_samples((jar_xm_context_t *)music.ctxData, (float *)pcm, frameCountToStream);
else if (AUDIO_DEVICE_FORMAT == ma_format_s16) jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)pcm, frameCountToStream);
else if (AUDIO_DEVICE_FORMAT == ma_format_u8) jar_xm_generate_samples_8bit((jar_xm_context_t *)music.ctxData, (char *)pcm, frameCountToStream);
if (AUDIO_DEVICE_FORMAT == ma_format_f32) jar_xm_generate_samples((jar_xm_context_t *)music.ctxData, (float *)AUDIO.System.pcm, frameCountToStream);
else if (AUDIO_DEVICE_FORMAT == ma_format_s16) jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)AUDIO.System.pcm, frameCountToStream);
else if (AUDIO_DEVICE_FORMAT == ma_format_u8) jar_xm_generate_samples_8bit((jar_xm_context_t *)music.ctxData, (char *)AUDIO.System.pcm, frameCountToStream);
} break;
#endif
@@ -1787,13 +1795,13 @@ void UpdateMusicStream(Music music)
case MUSIC_MODULE_MOD:
{
// NOTE: 3rd parameter (nbsample) specify the number of stereo 16bits samples you want, so sampleCount/2
jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)pcm, frameCountToStream, 0);
jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)AUDIO.System.pcm, frameCountToStream, 0);
} break;
#endif
default: break;
}
UpdateAudioStream(music.stream, pcm, frameCountToStream);
UpdateAudioStream(music.stream, AUDIO.System.pcm, frameCountToStream);
framesLeft -= frameCountToStream;
@@ -1804,9 +1812,6 @@ void UpdateMusicStream(Music music)
}
}
// Free allocated pcm data
RL_FREE(pcm);
// Reset audio stream for looping
if (streamEnding)
{