void RestartVorbisPlayer(VorbisPlayer *self)
{
stb_vorbis_seek_start(self->vorbis);
self->currentframe=NULL;
self->framepos=0;
self->framelength=0;
}
bool AEAudioStreamUpdate(AEAudioStream* self){
ALint processed=0;
alGetSourcei(self->source, AL_BUFFERS_PROCESSED, &processed);
while(processed--){
ALuint buffer=0;
alSourceUnqueueBuffers(self->source, 1, &buffer);
if(not AEAudioStreamStream(self, buffer)){
bool shouldExit=true;
if(self->shouldLoop){
stb_vorbis_seek_start(self->stream);
self->totalSamplesLeft=stb_vorbis_stream_length_in_samples(self->stream) * self->info.channels;
shouldExit=not AEAudioStreamStream(self, buffer);
}
if(shouldExit) return false;
}
alSourceQueueBuffers(self->source, 1, &buffer);
}
return true;
}
void snd_PlayStreaming( int streamID, float volume, float pan ) // todo: fade in?
{
assert( ( streamID >= 0 ) && ( streamID < MAX_STREAMING_SOUNDS ) );
SDL_LockAudioDevice( devID ); {
if( ( streamingSounds[streamID].access != NULL ) && !streamingSounds[streamID].playing ) {
streamingSounds[streamID].sdlStream = SDL_NewAudioStream( AUDIO_S16,
(Uint8)( streamingSounds[streamID].access->channels ), streamingSounds[streamID].access->sample_rate,
WORKING_FORMAT, streamingSounds[streamID].channels, WORKING_RATE );
if( streamingSounds[streamID].sdlStream == NULL ) {
llog( LOG_ERROR, "Unable to create SDL_AudioStream for streaming sound." );
return;
}
streamingSounds[streamID].playing = true;
streamingSounds[streamID].volume = volume;
streamingSounds[streamID].pan = pan;
stb_vorbis_seek_start( streamingSounds[streamID].access );
streamingSounds[streamID].readDone = false;
}
} SDL_UnlockAudioDevice( devID );
}
virtual void pull(AudioChunk &chunk)
{
if (!chunk.length()) return;
if (!valid || finished) {chunk.silence(); return;}
int samples, have = 0, need = chunk.length();
//Create pointers to 16-bit data
short *d16[PG_MAX_CHANNELS];
for (Uint32 i = 0; i < chunk.format().channels; ++i)
d16[i] = (short*) chunk.start(i);
while (true)
{
samples = stb_vorbis_get_samples_short(ogg,
chunk.format().channels, d16, (need-have));
if (samples < 0)
{
finished = true;
//cout << " VORBIS ERROR" << endl;
break;
}
if (samples == 0)
{
//File's end
if (loop)
{
stb_vorbis_seek_start(ogg);
continue;
}
else
{
finished = true;
break;
}
}
for (Uint32 i=0; i < chunk.format().channels; ++i)
d16[i] += samples;
have += samples;
//if (have > need) cout << "VORBIS OVERDRAW" << endl;
//std::cout << "OGG pull: " << have << "/" << need << std::endl;
if (have >= need) break;
}
//Cutoff marker if necessary
if (have < need) chunk.cutoff(have);
//Upsample data to 24-bit Sint32s
for (Uint32 i=0; i < chunk.format().channels; ++i)
{
Sint32 *start = chunk.start(i), *op = start + have;
short *ip = d16[i];
while (op!=start) {*(--op) = 256 * Sint32(*(--ip));}
}
}
void Sound::RewindDecoder(void* decoder)
{
if (!decoder)
return;
stb_vorbis* vorbis = static_cast<stb_vorbis*>(decoder);
stb_vorbis_seek_start(vorbis);
}
static inline void LoadNextFrameIfNeeded(VorbisPlayer *self)
{
if(self->framepos>=self->framelength)
{
self->framepos=0;
self->framelength=stb_vorbis_get_frame_float(self->vorbis,NULL,&self->currentframe);
if(!self->framelength)
{
stb_vorbis_seek_start(self->vorbis);
self->framelength=stb_vorbis_get_frame_float(self->vorbis,NULL,&self->currentframe);
}
}
}
result WavStreamInstance::rewind()
{
if (mOgg)
{
stb_vorbis_seek_start(mOgg);
}
else
if (mFile)
{
mFile->seek(mParent->mDataOffset);
}
mOffset = 0;
mStreamTime = 0;
return 0;
}
int WavStreamInstance::rewind()
{
if (mOgg)
{
stb_vorbis_seek_start(mOgg);
}
else
if (mFile)
{
fseek(mFile, mParent->mDataOffset, SEEK_SET);
}
mOffset = 0;
mStreamTime = 0;
return 1;
}
// Update (re-fill) music buffers if data already processed
extern void UpdateMusicStream()
{
ALuint buffer = 0;
ALint processed = 0;
bool active = true;
if (musicEnabled)
{
// Get the number of already processed buffers (if any)
alGetSourcei(currentMusic.source, AL_BUFFERS_PROCESSED, &processed);
while (processed > 0)
{
// Recover processed buffer for refill
alSourceUnqueueBuffers(currentMusic.source, 1, &buffer);
// Refill buffer
active = BufferMusicStream(buffer);
// If no more data to stream, restart music (if loop)
if ((!active) && (currentMusic.loop))
{
if (currentMusic.loop)
{
stb_vorbis_seek_start(currentMusic.stream);
currentMusic.totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels;
active = BufferMusicStream(buffer);
}
}
// Add refilled buffer to queue again... don't let the music stop!
alSourceQueueBuffers(currentMusic.source, 1, &buffer);
if(alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Ogg playing, error buffering data...");
processed--;
}
ALenum state;
alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state);
if ((state != AL_PLAYING) && active) alSourcePlay(currentMusic.source);
if (!active) StopMusicStream();
}
}
void AudioStream::Update()
{ //Hacer con variable auxiliar
ALint buffersProcessed;
alGetSourcei(source->GetID(), AL_BUFFERS_PROCESSED, &buffersProcessed);
while (buffersProcessed--) {
uint32 buffer;
alSourceUnqueueBuffers(source->GetID(), 1, &buffer);
if (Stream(buffer)) {
alSourceQueueBuffers(source->GetID(), 1, &buffer);
}
else if (shouldLoop) {
stb_vorbis_seek_start(stream);
samplesLeft = stb_vorbis_stream_length_in_samples(stream) * info.channels;
if (Stream(buffer)) alSourceQueueBuffers(source->GetID(), 1, &buffer);
}
}
}
unsigned OggVorbisSoundStream::GetData(signed char* dest, unsigned numBytes)
{
if (!decoder_)
return 0;
stb_vorbis* vorbis = static_cast<stb_vorbis*>(decoder_);
unsigned channels = stereo_ ? 2 : 1;
unsigned outSamples = stb_vorbis_get_samples_short_interleaved(vorbis, channels, (short*)dest, numBytes >> 1);
unsigned outBytes = (outSamples * channels) << 1;
// Rewind and retry if is looping and produced less output than should have
if (outBytes < numBytes && !stopAtEnd_)
{
numBytes -= outBytes;
stb_vorbis_seek_start(vorbis);
outSamples = stb_vorbis_get_samples_short_interleaved(vorbis, channels, (short*)(dest + outBytes), numBytes >> 1);
outBytes += (outSamples * channels) << 1;
}
void MusicOGG::run()
{
while (m_isPlaying)
{
if (m_bufferCount < MUSIC_BUFFER_COUNT)
{
m_mutex.lock();
auto pBuffer = m_buffers.back();
m_buffers.pop_back();
m_mutex.unlock();
pBuffer->count = stb_vorbis_get_samples_float_interleaved(m_pStream, m_engineChannelCount, pBuffer->data.data(), m_bufferMax) * m_engineChannelCount;
pBuffer->offset = 0;
if (pBuffer->count)
{
m_mutex.lock();
m_buffers.insert(m_buffers.begin() + m_bufferCount, pBuffer);
++m_bufferCount;
m_mutex.unlock();
}
else
{
if (m_loop)
{
stb_vorbis_seek_start(m_pStream);
m_mutex.lock();
m_buffers.push_back(pBuffer);
m_mutex.unlock();
continue;
}
else
{
// Done playing!
stb_vorbis_close(m_pStream);
m_pStream = nullptr;
m_done = true;
break;
}
}
}
std::this_thread::sleep_for(std::chrono::milliseconds(10));
}
}
int MusicPlayer::pullAudio(AUDIO_SAMPLE_TYPE *target, int sampleCount)
{
memset(target, 0, sampleCount * sizeof(AUDIO_SAMPLE_TYPE));
#ifdef USE_OGG
if(v && error==0)
{
int samples = stb_vorbis_get_samples_short_interleaved(v, channels, target, sampleCount);
if(samples==0)
{
stb_vorbis_seek_start(v);
stb_vorbis_get_samples_short_interleaved(v, channels, target, sampleCount);
// error=-1;
}
}
#endif
#ifdef USE_MP3
if (!error)
{
size_t done;
error = mpg123_read(mh, (unsigned char *) target, sampleCount
* sizeof(AUDIO_SAMPLE_TYPE), &done);
// read will return MPG123_DONE eventually...
if (error == MPG123_DONE)
{
mpg123_seek(mh, 0, SEEK_SET);
error = mpg123_read(mh, (unsigned char *) target, sampleCount
* sizeof(AUDIO_SAMPLE_TYPE), &done);
}
// ignore format change
if (error == MPG123_NEW_FORMAT)
error = MPG123_OK;
}
#endif
return sampleCount;
}
const i32 audio_updateAndPlay(MemoryArena_ *arena, AudioManager *audioManager,
AudioRenderer *audioRenderer)
{
AudioVorbis *audio = audioRenderer->audio;
if (!audio) return 0;
u32 alSourceId = getSourceId(audioManager, audioRenderer);
if (alIsSource(alSourceId) == AL_FALSE)
{
DEBUG_LOG("audio_updateAndPlay(): Update failed on invalid source id");
return -1;
}
ALint audioState;
alGetSourcei(alSourceId, AL_SOURCE_STATE, &audioState);
AL_CHECK_ERROR();
if (audioState == AL_STOPPED || audioState == AL_INITIAL)
{
if (audioState == AL_STOPPED)
{
if (audioRenderer->numPlays != AUDIO_REPEAT_INFINITE)
audioRenderer->numPlays--;
if (audioRenderer->numPlays == AUDIO_REPEAT_INFINITE ||
audioRenderer->numPlays > 0)
{
// TODO(doyle): Delete and recreate fixes clicking when reusing
// buffers
alDeleteBuffers(ARRAY_COUNT(audioRenderer->bufferId),
audioRenderer->bufferId);
AL_CHECK_ERROR();
alGenBuffers(ARRAY_COUNT(audioRenderer->bufferId),
audioRenderer->bufferId);
AL_CHECK_ERROR();
}
else
{
i32 result =
rendererRelease(arena, audioManager, audioRenderer);
return result;
}
}
if (audioRenderer->isStreaming)
{
stb_vorbis_seek_start(audio->file);
for (i32 i = 0; i < ARRAY_COUNT(audioRenderer->bufferId); i++)
{
i16 audioChunk[AUDIO_CHUNK_SIZE_] = {0};
stb_vorbis_get_samples_short_interleaved(
audio->file, audio->info.channels, audioChunk,
AUDIO_CHUNK_SIZE_);
alBufferData(audioRenderer->bufferId[i], audioRenderer->format,
audioChunk, AUDIO_CHUNK_SIZE_ * sizeof(i16),
audio->info.sample_rate);
AL_CHECK_ERROR();
}
alSourceQueueBuffers(alSourceId,
ARRAY_COUNT(audioRenderer->bufferId),
audioRenderer->bufferId);
}
else
{
alSourceQueueBuffers(alSourceId, 1, audioRenderer->bufferId);
}
AL_CHECK_ERROR();
alSourcePlay(alSourceId);
AL_CHECK_ERROR();
}
else if (audioState == AL_PLAYING)
{
ALint numProcessedBuffers;
alGetSourcei(alSourceId, AL_BUFFERS_PROCESSED,
&numProcessedBuffers);
AL_CHECK_ERROR();
if (numProcessedBuffers > 0)
{
// TODO(doyle): Possibly wrong, we should pass in all processed buffers?
ALint numBuffersToUnqueue = 1;
ALuint emptyBufferId;
alSourceUnqueueBuffers(alSourceId, numBuffersToUnqueue,
&emptyBufferId);
AL_CHECK_ERROR();
i16 audioChunk[AUDIO_CHUNK_SIZE_] = {0};
i32 sampleCount = stb_vorbis_get_samples_short_interleaved(
audio->file, audio->info.channels, audioChunk,
AUDIO_CHUNK_SIZE_);
/* There are still samples to play */
if (sampleCount > 0)
{
alBufferData(emptyBufferId, audioRenderer->format, audioChunk,
sampleCount * audio->info.channels * sizeof(i16),
audio->info.sample_rate);
AL_CHECK_ERROR();
alSourceQueueBuffers(alSourceId, 1, &emptyBufferId);
AL_CHECK_ERROR();
}
}
}
return 0;
}
void WavStreamInstance::getAudio(float *aBuffer, unsigned int aSamples)
{
unsigned int channels = mChannels;
if (mFile == NULL)
return;
if (mOgg)
{
unsigned int offset = 0;
if (mOggFrameOffset < mOggFrameSize)
{
int b = getOggData(mOggOutputs, aBuffer, aSamples, aSamples, mOggFrameSize, mOggFrameOffset, channels);
mOffset += b;
offset += b;
mOggFrameOffset += b;
}
while (offset < aSamples)
{
mOggFrameSize = stb_vorbis_get_frame_float(mOgg, NULL, &mOggOutputs);
mOggFrameOffset = 0;
int b;
b = getOggData(mOggOutputs, aBuffer + offset, aSamples - offset, aSamples, mOggFrameSize, mOggFrameOffset, channels);
mOffset += b;
offset += b;
mOggFrameOffset += b;
if (mOffset >= mParent->mSampleCount)
{
if (mFlags & AudioSourceInstance::LOOPING)
{
stb_vorbis_seek_start(mOgg);
mOffset = aSamples - offset;
mLoopCount++;
}
else
{
unsigned int i;
for (i = 0; i < channels; i++)
memset(aBuffer + offset + i * aSamples, 0, sizeof(float) * (aSamples - offset));
mOffset += aSamples - offset;
offset = aSamples;
}
}
}
}
else
{
unsigned int copysize = aSamples;
if (copysize + mOffset > mParent->mSampleCount)
{
copysize = mParent->mSampleCount - mOffset;
}
getWavData(mFile, aBuffer, copysize, aSamples, channels, mParent->mChannels, mParent->mBits);
if (copysize != aSamples)
{
if (mFlags & AudioSourceInstance::LOOPING)
{
mFile->seek(mParent->mDataOffset);
getWavData(mFile, aBuffer + copysize, aSamples - copysize, aSamples, channels, mParent->mChannels, mParent->mBits);
mOffset = aSamples - copysize;
mLoopCount++;
}
else
{
unsigned int i;
for (i = 0; i < channels; i++)
memset(aBuffer + copysize + i * aSamples, 0, sizeof(float) * (aSamples - copysize));
mOffset += aSamples;
}
}
else
{
mOffset += aSamples;
}
}
}
// stereo LRLRLR order
void mixerCallback( void* userdata, Uint8* streamData, int len )
{
memset( streamData, len, workingSilence );
memset( workingBuffer, 0, workingBufferSize );
if( workingBuffer == NULL ) {
return;
}
int numSamples = ( len / WORKING_CHANNELS ) / ( ( SDL_AUDIO_MASK_BITSIZE & WORKING_FORMAT ) / 8 );
#if 0
int soundFreq = 440; // wave length
float step = 1.0f / WORKING_RATE;
// for testing audio output
for( int s = 0; s < numSamples; ++s ) {
int streamIdx = ( s * WORKING_CHANNELS );
testTimePassed += 1.0f / WORKING_RATE;
float v = sinf( testTimePassed * soundFreq * M_TWO_PI_F );
workingBuffer[streamIdx] = v * 0.1f;
workingBuffer[streamIdx+1] = v * 0.1f;
}
#else
// advance each playing sound
for( EntityID id = idSet_GetFirstValidID( &playingIDSet ); id != INVALID_ENTITY_ID; id = idSet_GetNextValidID( &playingIDSet, id ) ) {
int i = idSet_GetIndex( id );
Sound* snd = &( playingSounds[i] );
Sample* sample = &( samples[snd->sample] );
// for right now lets assume the pitch will stay the same, when we get it working it'll just involve
// changing the speed at which we move through the array
bool soundDone = false;
float volume = snd->volume * sbSoundGroups[snd->group].volume * masterVolume;
for( int s = 0; ( s < numSamples ) && !soundDone; ++s ) {
int streamIdx = ( s * WORKING_CHANNELS );
// we're assuming stereo output here
if( sample->numChannels == 1 ) {
float data = sample->data[(int)snd->pos] * volume;
/* left */ workingBuffer[streamIdx] += data * inverseLerp( 1.0f, 0.0f, snd->pan );
/* right */ workingBuffer[streamIdx+1] += data * inverseLerp( -1.0f, 0.0f, snd->pan );
snd->pos += snd->pitch;
} else {
// if the sample is stereo then we ignore panning
// NOTE: Pitch change doesn't work with stereo samples yet
workingBuffer[streamIdx] += sample->data[(int)snd->pos] * volume;
workingBuffer[streamIdx+1] += sample->data[(int)snd->pos+1] * volume;
snd->pos += 2.0f;
}
if( snd->pos >= sample->numSamples ) {
if( sample->loops ) {
snd->pos -= (float)sample->numSamples;
} else {
soundDone = true;
}
}
}
if( soundDone ) {
idSet_ReleaseID( &playingIDSet, id ); // this doesn't invalidate the id for the loop
}
}
for( int i = 0; i < MAX_STREAMING_SOUNDS; ++i ) {
if( !streamingSounds[i].playing ) continue;
StreamingSound* stream = &( streamingSounds[i] );
bool soundDone = false;
float volume = stream->volume * sbSoundGroups[stream->group].volume * masterVolume;
// if the next buffer fill would be past what we have loaded then load some more
// note: the SDL_AudioStreamAvailable return value is in bytes
while( ( SDL_AudioStreamAvailable( stream->sdlStream ) < len ) && !( stream->readDone ) ) {
int read = 0;
int request = STREAM_READ_BUFFER_SIZE / sizeof( short );
size_t test = ARRAY_SIZE( streamReadBuffer );
// returns the number of samples stored per channel
int samplesPerChannel = stb_vorbis_get_samples_short_interleaved(
stream->access, stream->access->channels, streamReadBuffer, request );
read = samplesPerChannel * sizeof( short );
SDL_AudioStreamPut( stream->sdlStream, streamReadBuffer, samplesPerChannel * stream->channels * sizeof( short ) );
// reached the end of the file, are we looping?
if( read != request ) {
if( stream->loops ) {
stb_vorbis_seek_start( stream->access );
} else {
stream->readDone = true;
SDL_AudioStreamFlush( stream->sdlStream );
}
}
}
// read data from the audio stream until there is no more left or the end of the buffer to fill
int bytesToStream = numSamples * stream->channels * sizeof( sbStreamWorkingBuffer[0] );
sb_Reserve( sbStreamWorkingBuffer, (size_t)bytesToStream );
int gotten = SDL_AudioStreamGet( stream->sdlStream, sbStreamWorkingBuffer, bytesToStream );
if( gotten < 0 ) {
llog( LOG_ERROR, "Error reading from sdlStream: %s", SDL_GetError( ) );
stream->playing = false;
continue;
} else if( gotten == 0 ) {
// end of stream
stream->playing = false;
}
int samplesGotten = gotten / ( stream->channels * sizeof( sbStreamWorkingBuffer[0] ) );
for( int s = 0; s < samplesGotten; ++s ) {
// then just mix those samples
int streamIdx = ( s * WORKING_CHANNELS );
int workingIdx = ( s * stream->channels );
// we're assuming stereo output here
if( stream->access->channels == 1 ) {
float data = sbStreamWorkingBuffer[workingIdx] * volume;
workingBuffer[streamIdx] += data * inverseLerp( 1.0f, 0.0f, stream->pan ); // left
workingBuffer[streamIdx+1] += data * inverseLerp( -1.0f, 0.0f, stream->pan ); // right
} else {
// if the sample is stereo then we ignore panning
workingBuffer[streamIdx] += sbStreamWorkingBuffer[workingIdx] * volume;
workingBuffer[streamIdx+1] += sbStreamWorkingBuffer[workingIdx + 1] * volume;
}
}
}
#endif
memcpy( streamData, workingBuffer, len );
}
void nVorbisStream::rewind()
{
stb_vorbis_seek_start(_vorbis);
}
