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REVIEWED: UpdateMusicStream() #2228

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Trying to implement proper looping, independently of frame rate.
This commit is contained in:
Ray 2022-07-11 21:19:21 +02:00
parent 0379b94b7a
commit 3ebfee5dbc
1 changed files with 64 additions and 30 deletions
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94
src/raudio.c
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@ -1724,28 +1724,34 @@ void UpdateMusicStream(Music music)
bool streamEnding = false;
unsigned int subBufferSizeInFrames = music.stream.buffer->sizeInFrames/2;
// NOTE: Using dynamic allocation because it could require more than 16KB
// On first call of this function we lazily pre-allocated a temp buffer to read audio files/memory data in
unsigned int pcmSize = subBufferSizeInFrames*music.stream.channels*music.stream.sampleSize/8;
if (AUDIO.System.pcmCapacity < pcmSize)
if (AUDIO.System.pcmBufferSize < pcmSize)
{
RL_FREE(AUDIO.System.pcm);
AUDIO.System.pcm = RL_CALLOC(1, pcmSize);
AUDIO.System.pcmCapacity = pcmSize;
RL_FREE(AUDIO.System.pcmBuffer);
AUDIO.System.pcmBuffer = RL_CALLOC(1, pcmSize);
AUDIO.System.pcmBufferSize = pcmSize;
}
int frameCountToStream = 0; // Total size of data in frames to be streamed
int framesLeft = music.frameCount - music.stream.buffer->framesProcessed; // Frames left to be processed
int framesToStream = 0; // Total frames to be streamed
unsigned int framesLoopingExtra = 0; // In case music requires to loop, we could need to add more frames from beginning to fill buffer
// TODO: Get the framesLeft using framesProcessed... but first, get total frames processed correctly...
//ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
unsigned int framesLeft = music.frameCount - music.stream.buffer->framesProcessed;
while (IsAudioStreamProcessed(music.stream))
// Check both sub-buffers to check if they require refilling
for (int i = 0; i < 2; i++)
{
// WARNING: If audio needs to loop but the frames left are less than the actual size of buffer to fill,
// the buffer is only partially filled and no refill is done until next frame call, generating a silence
// TODO: Possible solution: In case of music loop, fill frames left + frames from start to fill the buffer to process
if (framesLeft >= subBufferSizeInFrames) frameCountToStream = subBufferSizeInFrames;
else frameCountToStream = framesLeft;
if ((music.stream.buffer != NULL) && !music.stream.buffer->isSubBufferProcessed[i]) continue; // No refilling required, move to next sub-buffer
if (framesLeft >= subBufferSizeInFrames) framesToStream = subBufferSizeInFrames;
else
{
framesToStream = framesLeft;
// WARNING: If audio needs to loop but the frames left are less than the actual size of buffer to fill,
// the buffer is only partially filled and no refill is done until next frame call, generating a silence
// SOLUTION: In case of music loop, fill frames left + frames from start to fill the buffer to process
if (music.looping) framesLoopingExtra = subBufferSizeInFrames - framesLeft;
}
switch (music.ctxType)
{
@ -1753,8 +1759,18 @@ 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 *)AUDIO.System.pcm);
else if (music.stream.sampleSize == 32) drwav_read_pcm_frames_f32((drwav *)music.ctxData, frameCountToStream, (float *)AUDIO.System.pcm);
if (music.stream.sampleSize == 16) drwav_read_pcm_frames_s16((drwav *)music.ctxData, framesToStream, (short *)AUDIO.System.pcmBuffer);
else if (music.stream.sampleSize == 32) drwav_read_pcm_frames_f32((drwav *)music.ctxData, framesToStream, (float *)AUDIO.System.pcmBuffer);
if (framesLoopingExtra > 0)
{
drwav_seek_to_pcm_frame((drwav *)music.ctxData, 0);
if (music.stream.sampleSize == 16) drwav_read_pcm_frames_s16((drwav *)music.ctxData, framesLoopingExtra, (short *)AUDIO.System.pcmBuffer + framesToStream*music.stream.channels);
else if (music.stream.sampleSize == 32) drwav_read_pcm_frames_f32((drwav *)music.ctxData, framesLoopingExtra, (float *)AUDIO.System.pcmBuffer + framesToStream*music.stream.channels);
framesToStream += framesLoopingExtra;
}
} break;
#endif
@ -1762,7 +1778,9 @@ 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 *)AUDIO.System.pcm, frameCountToStream*music.stream.channels);
stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)AUDIO.System.pcmBuffer, framesToStream*music.stream.channels);
// stb_vorbis_seek_start((stb_vorbis *)music.ctxData);
} break;
#endif
@ -1772,12 +1790,16 @@ void UpdateMusicStream(Music music)
// NOTE: Returns the number of samples to process (not required)
drflac_read_pcm_frames_s16((drflac *)music.ctxData, frameCountToStream*music.stream.channels, (short *)AUDIO.System.pcm);
// drflac_seek_to_pcm_frame((drflac *)music.ctxData, 0);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3:
{
drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, frameCountToStream, (float *)AUDIO.System.pcm);
drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, framesToStream, (float *)AUDIO.System.pcmBuffer);
//drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, 0);
} break;
#endif
@ -1785,9 +1807,11 @@ 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 *)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);
if (AUDIO_DEVICE_FORMAT == ma_format_f32) jar_xm_generate_samples((jar_xm_context_t *)music.ctxData, (float *)AUDIO.System.pcmBuffer, framesToStream);
else if (AUDIO_DEVICE_FORMAT == ma_format_s16) jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)AUDIO.System.pcmBuffer, framesToStream);
else if (AUDIO_DEVICE_FORMAT == ma_format_u8) jar_xm_generate_samples_8bit((jar_xm_context_t *)music.ctxData, (char *)AUDIO.System.pcmBuffer, framesToStream);
//jar_xm_reset((jar_xm_context_t *)music.ctxData);
} break;
#endif
@ -1795,18 +1819,20 @@ 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 *)AUDIO.System.pcm, frameCountToStream, 0);
jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)AUDIO.System.pcmBuffer, framesToStream, 0);
//jar_mod_seek_start((jar_mod_context_t *)music.ctxData);
} break;
#endif
default: break;
}
UpdateAudioStream(music.stream, AUDIO.System.pcm, frameCountToStream);
UpdateAudioStream(music.stream, AUDIO.System.pcmBuffer, framesToStream);
framesLeft -= frameCountToStream;
if (framesLeft <= 0)
if (framesLeft <= subBufferSizeInFrames)
{
// Streaming is ending, we filled latest frames from input
streamEnding = true;
break;
}
@ -1815,8 +1841,16 @@ void UpdateMusicStream(Music music)
// Reset audio stream for looping
if (streamEnding)
{
StopMusicStream(music); // Stop music (and reset)
if (music.looping) PlayMusicStream(music); // Play again
if (music.looping)
{
PlayMusicStream(music); // Play again
// Set cursor offset to extra frames filled previously
music.stream.buffer->frameCursorPos = framesLoopingExtra;
// TODO: It's not working properly... :(
}
else StopMusicStream(music); // Stop music (and reset)
}
else
{