void BufferedAudioSource::close()
{
if(!isLoadedInMemory())
{
BufferedAudioSourceThread::getInstance()->removeSource(this);
}
}
void BufferedAudioSource::prime()
{
if(!isLoadedInMemory())
{
// if this is the first time source is added, trigger to read data immediately
if (BufferedAudioSourceThread::getInstance()->addSource(this)) {
BufferedAudioSourceThread::getInstance()->readMore();
}
}
}
void BufferedAudioSource::flush()
{
if(!isLoadedInMemory())
{
BufferedAudioSourceThread::getInstance()->removeSource(this);
{
RScopedLock l(&mLock);
mBuffer.clear();
}
}
}
void BufferedAudioSource::setPosition(double seconds)
{
seconds = seconds < 0 ? 0.0f : seconds;
seconds = seconds > getLength() ? getLength() : seconds;
RScopedLock l(&mLock);
Int64 frames = (Int64)(seconds * getSampleRate());
if(!isLoadedInMemory())
{
mBuffer.clear();
setDecoderPosition(frames);
BufferedAudioSourceThread::getInstance()->readMore();
}
}
Int64 ExtAudioFileAudioSource::decodeData(float* buffer, UInt32 numFrames)
{
RPRINT("decoding data\n");
UInt32 numChannels = getNumChannels();
OSStatus err = noErr;
mReadBuffer.resize(numChannels * numFrames);
// Set up the buffers
setUpBuffers(&mReadBuffer[0], numChannels, numFrames);
// Read the data out of our file, filling our output buffer
UInt32 framesRead = numFrames;
if(!isLoadedInMemory())
{
RScopedLock l(&mDecodeLock);
err = ExtAudioFileRead (mAudioFile, &framesRead, mpBufferList);
}
else
{
err = ExtAudioFileRead (mAudioFile, &framesRead, mpBufferList);
}
if(err || framesRead == 0)
{
mEOF = true;
RPRINT("done decoding data\n");
return 0;
}
// The data is expected to be interlaced
for(UInt32 j = 0; j < numChannels; ++j)
{
float *pTemp = &mReadBuffer[j * numFrames];
float *pOut = &buffer[j];
for(UInt32 i = j; i < framesRead; i++)
{
*pOut = *pTemp++;
pOut += numChannels;
}
}
RPRINT("done decoding data\n");
return framesRead;
}
Int64 BufferedAudioSource::readFrames(float* buffer, UInt32 numChannels, UInt32 numFrames, AudioSourceState& state)
{
RScopedLock l(&mLock);
Int64 framesRead = numFrames;
int framesAvailable = mBuffer.size() / getNumChannels() - state.mCurrentFrame;
// For disk-streaming sources we calculate available frames using the whole buffer
if(!isLoadedInMemory()) {
framesAvailable = mBuffer.size() / getNumChannels();
}
Int64 loopEndFrame = convertSecondsToSamples(state.mLoopEnd);
Int64 totalFrames = convertSecondsToSamples(getLength());
bool needToLoop = state.mLooping && ((state.mCurrentFrame >= loopEndFrame && loopEndFrame > 0) || (framesAvailable == 0 && mEOF));
if(framesAvailable > 0 && !needToLoop)
{
// Make sure we don't read passed the end loop frame
if(state.mLooping && loopEndFrame > 0 && loopEndFrame > state.mCurrentFrame)
framesRead = (std::min)(framesRead, loopEndFrame-state.mCurrentFrame);
if(framesAvailable < framesRead)
framesRead = framesAvailable;
int sourceChannels = getNumChannels();
int frameOffset = state.mCurrentFrame;
// For disk-streaming sources we always start at the beginning of the buffer
if(!isLoadedInMemory())
frameOffset = 0;
for(UInt32 j = 0; j < numChannels; ++j)
{
float *in = NULL;
if(sourceChannels == 1)
in = mBuffer.getData(0, frameOffset);
else
in = mBuffer.getData(j, frameOffset);
for(UInt32 i = 0; i < framesRead; ++i)
{
*(buffer++) = *in;
in += sourceChannels;
}
}
state.mCurrentFrame += framesRead;
if(!isLoadedInMemory())
{
mBuffer.erase(0, framesRead);
framesAvailable = mBuffer.size() / getNumChannels();
UInt32 minimumFrames = getSampleRate() * SECONDS_TO_BUFFER / 2;
if(framesAvailable <= minimumFrames)
{
BufferedAudioSourceThread::getInstance()->readMore();
}
}
}
else
{
framesRead = ERR_BUFFERING;
if(needToLoop)
{
setPosition(state.mLoopStart);
state.mCurrentFrame = convertSecondsToSamples(state.mLoopStart);
// Now that we rewinded, read new data if preloaded
if(isLoadedInMemory())
{
return readFrames(buffer, numChannels, numFrames, state);
}
}
if(mEOF)
{
return 0; // signal that we are done
}
else if(!isLoadedInMemory())
{
BufferedAudioSourceThread::getInstance()->readMore();
}
}
return framesRead;
}
Int64 BufferedAudioSource::readFrames(float* buffer, UInt32 numChannels, UInt32 numFrames, AudioSourceState& state)
{
mLock.lock();
Int64 framesRead = numFrames;
int framesAvailable = mBuffer.size() / getNumChannels() - state.mCurrentFrame;
// For disk-streaming sources we calculate available frames using the whole buffer
if(!isLoadedInMemory())
framesAvailable = mBuffer.size() / getNumChannels();
mLock.unlock();
Int64 loopEndFrame = convertSecondsToSamples(state.mLoopEnd);
Int64 totalFrames = convertSecondsToSamples(getLength());
bool needToLoop = state.mLooping && ((state.mCurrentFrame >= loopEndFrame && loopEndFrame > 0) || (framesAvailable == 0 && mEOF));
if(framesAvailable > 0 && !needToLoop)
{
RScopedLock l(&mLock);
if(framesAvailable < numFrames)
framesRead = framesAvailable;
int sourceChannels = getNumChannels();
int frameOffset = state.mCurrentFrame;
// For disk-streaming sources we always start at the beginning of the buffer
if(!isLoadedInMemory())
frameOffset = 0;
for(UInt32 j = 0; j < numChannels; ++j)
{
float *in = NULL;
if(sourceChannels == 1)
in = mBuffer.getData(0, frameOffset);
else
in = mBuffer.getData(j, frameOffset);
for(UInt32 i = 0; i < framesRead; ++i)
{
*(buffer++) = *in;
in += sourceChannels;
}
}
state.mCurrentFrame += framesRead;
if(!isLoadedInMemory())
{
mBuffer.erase(0, framesRead);
framesAvailable = mBuffer.size() / getNumChannels();
UInt32 minimumFrames = getSampleRate() / 3; // 1/3 of a second
if(framesAvailable <= minimumFrames)
{
BufferedAudioSourceThread::getInstance()->readMore();
}
}
}
else
{
framesRead = ERR_BUFFERING;
if(needToLoop)
{
setPosition(state.mLoopStart);
state.mCurrentFrame = convertSecondsToSamples(state.mLoopStart);
}
Int64 totalFrames = convertSecondsToSamples(getLength());
if(state.mCurrentFrame >= totalFrames)
{
if(!isLoadedInMemory())
BufferedAudioSourceThread::getInstance()->removeSource(this);
return 0; // signal that we are done
}
else
if(!isLoadedInMemory())
BufferedAudioSourceThread::getInstance()->readMore();
}
return framesRead;
}
