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small fix for streaming

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There is still an issue where audio will cut off after a brief moment
This commit is contained in:
Joshua Reisenauer 2016-04-25 22:18:49 -07:00
parent 04d9deac92
commit 3104d3d6cd
1 changed files with 20 additions and 17 deletions
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37
src/audio.c
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@ -567,10 +567,15 @@ void PlayMusicStream(char *fileName)
// NOTE: Regularly, we must check if a buffer has been processed and refill it: UpdateMusicStream()
currentMusic.totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
currentMusic.totalLengthSeconds = stb_vorbis_stream_length_in_seconds(currentMusic.stream);
}
}
else if (strcmp(GetExtension(fileName),"xm") == 0)
{
// Stop current music, clean buffers, unload current stream
StopMusicStream();
// new song settings for xm chiptune
currentMusic.chipTune = true;
currentMusic.channels = 2;
currentMusic.sampleRate = 48000;
@ -714,39 +719,37 @@ float GetMusicTimePlayed(void)
static bool BufferMusicStream(ALuint buffer)
{
short pcm[MUSIC_BUFFER_SIZE];
int size = 0; // Total size of data steamed (in bytes)
int streamedBytes = 0; // Bytes of data obtained in one samples get
int streamedBytes = 0; // samples of data obtained, channels are not included in calculation
bool active = true; // We can get more data from stream (not finished)
if (musicEnabled)
{
while (size < MUSIC_BUFFER_SIZE)
if (currentMusic.chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
{
if (currentMusic.chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
{
int readlen = (MUSIC_BUFFER_SIZE - size) / 2;
jar_xm_generate_samples_16bit(currentMusic.chipctx, pcm + size, readlen); // reads 2*readlen shorts and moves them to buffer+size memory location
streamedBytes = readlen * 4; // Not sure if this is what it needs
}
else
int readlen = MUSIC_BUFFER_SIZE / 2;
jar_xm_generate_samples_16bit(currentMusic.chipctx, pcm, readlen); // reads 2*readlen shorts and moves them to buffer+size memory location
size += readlen * currentMusic.channels; // Not sure if this is what it needs
}
else
{
while (size < MUSIC_BUFFER_SIZE)
{
streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic.stream, currentMusic.channels, pcm + size, MUSIC_BUFFER_SIZE - size);
if (streamedBytes > 0) size += (streamedBytes*currentMusic.channels);
else break;
}
if (streamedBytes > 0) size += (streamedBytes*currentMusic.channels);
else break;
}
//TraceLog(DEBUG, "Streaming music data to buffer. Bytes streamed: %i", size);
TraceLog(DEBUG, "Streaming music data to buffer. Bytes streamed: %i", size);
}
if (size > 0)
{
alBufferData(buffer, currentMusic.format, pcm, size*sizeof(short), currentMusic.sampleRate);
currentMusic.totalSamplesLeft -= size;
if(currentMusic.totalSamplesLeft <= 0) active = false; // end if no more samples left
}
else
{