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buffering of music now uses update audio context

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This commit is contained in:
Joshua Reisenauer 2016-05-12 16:02:23 -07:00
parent f0ada8c40d
commit 83dbc07650
2 changed files with 38 additions and 68 deletions
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96
src/audio.c
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@ -118,12 +118,12 @@ static Music currentMusic[MAX_MUSIC_STREAMS]; // Current
//----------------------------------------------------------------------------------
// Module specific Functions Declaration
//----------------------------------------------------------------------------------
static Wave LoadWAV(const char *fileName); // Load WAV file
static Wave LoadOGG(char *fileName); // Load OGG file
static void UnloadWave(Wave wave); // Unload wave data
static Wave LoadWAV(const char *fileName); // Load WAV file
static Wave LoadOGG(char *fileName); // Load OGG file
static void UnloadWave(Wave wave); // Unload wave data
static bool BufferMusicStream(int index, ALuint buffer); // Fill music buffers with data
static void EmptyMusicStream(int index); // Empty music buffers
static bool BufferMusicStream(int index); // Fill music buffers with data
static void EmptyMusicStream(int index); // Empty music buffers
static unsigned short FillAlBufferWithSilence(AudioContext_t *context, ALuint buffer);// fill buffer with zeros, returns number processed
static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len); // pass two arrays of the same legnth in
@ -970,7 +970,7 @@ float GetMusicTimePlayed(int index)
//----------------------------------------------------------------------------------
// Fill music buffers with new data from music stream
static bool BufferMusicStream(int index, ALuint buffer)
static bool BufferMusicStream(int index)
{
short pcm[MUSIC_BUFFER_SIZE_SHORT];
float pcmf[MUSIC_BUFFER_SIZE_FLOAT];
@ -985,33 +985,17 @@ static bool BufferMusicStream(int index, ALuint buffer)
{
int readlen = MUSIC_BUFFER_SIZE_FLOAT / 2;
jar_xm_generate_samples(currentMusic[index].chipctx, pcmf, readlen); // reads 2*readlen shorts and moves them to buffer+size memory location
UpdateAudioContext(currentMusic[index].ctx, pcmf, MUSIC_BUFFER_SIZE_FLOAT);
size += readlen * currentMusic[index].ctx->channels; // Not sure if this is what it needs
alBufferData(buffer, currentMusic[index].ctx->alFormat, pcmf, size*sizeof(float), 48000);
currentMusic[index].totalSamplesLeft -= size;
if(currentMusic[index].totalSamplesLeft <= 0) active = false;
}
else
{
while (size < MUSIC_BUFFER_SIZE_SHORT)
{
streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic[index].stream, currentMusic[index].ctx->channels, pcm + size, MUSIC_BUFFER_SIZE_SHORT - size);
if (streamedBytes > 0) size += (streamedBytes*currentMusic[index].ctx->channels);
else break;
}
if (size > 0)
{
alBufferData(buffer, currentMusic[index].ctx->alFormat, pcm, size*sizeof(short), currentMusic[index].ctx->sampleRate);
currentMusic[index].totalSamplesLeft -= size;
if(currentMusic[index].totalSamplesLeft <= 0) active = false; // end if no more samples left
}
else
{
active = false;
TraceLog(WARNING, "No more data obtained from stream");
}
streamedBytes = stb_vorbis_get_samples_short_interleaved(currentMusic[index].stream, currentMusic[index].ctx->channels, pcm, MUSIC_BUFFER_SIZE_SHORT);
UpdateAudioContext(currentMusic[index].ctx, pcm, MUSIC_BUFFER_SIZE_SHORT);
currentMusic[index].totalSamplesLeft -= MUSIC_BUFFER_SIZE_SHORT;
if(currentMusic[index].totalSamplesLeft <= 0) active = false;
}
TraceLog(DEBUG, "Streaming music data to buffer. Bytes streamed: %i", size);
}
@ -1038,53 +1022,39 @@ static void EmptyMusicStream(int index)
// Update (re-fill) music buffers if data already processed
void UpdateMusicStream(int index)
{
ALuint buffer = 0;
ALint processed = 0;
ALenum state;
bool active = true;
if (index < MAX_MUSIC_STREAMS && musicEnabled)
{
// Get the number of already processed buffers (if any)
alGetSourcei(currentMusic[index].source, AL_BUFFERS_PROCESSED, &processed);
while (processed > 0)
active = BufferMusicStream(index);
if ((!active) && (currentMusic[index].loop))
{
// Recover processed buffer for refill
alSourceUnqueueBuffers(currentMusic[index].source, 1, &buffer);
// Refill buffer
active = BufferMusicStream(buffer);
// If no more data to stream, restart music (if loop)
if ((!active) && (currentMusic[index].loop))
if(currentMusic[index].chipTune)
{
if(currentMusic[index].chipTune)
{
currentMusic[index].totalSamplesLeft = currentMusic[index].totalLengthSeconds * currentMusic[index].ctx->sampleRate;
}
else
{
stb_vorbis_seek_start(currentMusic[index].stream);
currentMusic[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[index].stream)*currentMusic[index].ctx->channels;
}
active = BufferMusicStream(buffer);
currentMusic[index].totalSamplesLeft = currentMusic[index].totalLengthSeconds * currentMusic[index].ctx->sampleRate;
}
// Add refilled buffer to queue again... don't let the music stop!
alSourceQueueBuffers(currentMusic[index].source, 1, &buffer);
if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data...");
processed--;
else
{
stb_vorbis_seek_start(currentMusic[index].stream);
currentMusic[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[index].stream)*currentMusic[index].ctx->channels;
}
active = BufferMusicStream(index);
}
ALenum state;
alGetSourcei(currentMusic[index].source, AL_SOURCE_STATE, &state);
if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data...");
if ((state != AL_PLAYING) && active) alSourcePlay(currentMusic[index].source);
if (!active) StopMusicStream();
processed--;
}
alGetSourcei(currentMusic[index].source, AL_SOURCE_STATE, &state);
if ((state != AL_PLAYING) && active) alSourcePlay(currentMusic[index].source);
if (!active) StopMusicStream();
}
// Load WAV file into Wave structure

10
src/easings.h
View file

@ -18,11 +18,11 @@
* float speed = 1.f;
* float currentTime = 0.f;
* float currentPos[2] = {0,0};
* float newPos[2] = {1,1};
* float tempPosition[2] = currentPos;//x,y positions
* while(currentPos[0] < newPos[0])
* currentPos[0] = EaseSineIn(currentTime, tempPosition[0], tempPosition[0]-newPos[0], speed);
* currentPos[1] = EaseSineIn(currentTime, tempPosition[1], tempPosition[1]-newPos[0], speed);
* float finalPos[2] = {1,1};
* float startPosition[2] = currentPos;//x,y positions
* while(currentPos[0] < finalPos[0])
* currentPos[0] = EaseSineIn(currentTime, startPosition[0], startPosition[0]-finalPos[0], speed);
* currentPos[1] = EaseSineIn(currentTime, startPosition[1], startPosition[1]-finalPos[0], speed);
* currentTime += diffTime();
*
* A port of Robert Penner's easing equations to C (http://robertpenner.com/easing/)