void BufferedAudioSourceThread::run()
{
std::vector<float> tempBuffer;
while(1)
{
mReadMore.wait();
if(shouldThreadExit())
break;
{ // create a local scope for our locking object
RScopedLock l(&mLock);
for(UInt32 i = 0; i < mSources.size(); ++i)
{
BufferedAudioSource *pSource = mSources[i];
pSource->mLock.lock();
Int64 totalFrames = pSource->getSampleRate() * SECONDS_TO_BUFFER;
Int64 framesThreshold = totalFrames / 2; //
Int64 availableFrames = pSource->mBuffer.size() / pSource->getNumChannels();
pSource->mLock.unlock();
if(availableFrames <= framesThreshold)
{
Int64 numFrames = totalFrames - availableFrames;
Int64 framesToRead = numFrames;
Int64 frames = 0;
tempBuffer.resize(numFrames * pSource->getNumChannels());
float *pBuffer = &tempBuffer[0];
// keep track of last known version when starting to buffer
// to discard any previously buffered data in case source
// changed position. This avoids having to keep a lock
// while buffering data.
int lastVersion = pSource->getPositionVersion();
while(framesToRead > 0 && !pSource->isEOF())
{
int currentVersion = 0;
frames = pSource->decodeData(pBuffer, framesToRead, currentVersion);
// check if version changed since we started buffering
// and discard everything we've buffered before if so
if (currentVersion != lastVersion) {
tempBuffer.erase(tempBuffer.begin(), tempBuffer.begin() + ((numFrames - framesToRead) * pSource->getNumChannels()));
tempBuffer.resize(numFrames * pSource->getNumChannels());
pBuffer = &tempBuffer[0];
framesToRead = numFrames;
}
framesToRead -= frames;
pBuffer += frames * pSource->getNumChannels();
lastVersion = currentVersion;
}
// Now write the data back to the buffer
Int64 framesRead = numFrames - framesToRead;
if(framesRead > 0)
{
pSource->mLock.lock();
pSource->mBuffer.putData(&tempBuffer[0], framesRead * pSource->getNumChannels());
pSource->mLock.unlock();
}
}
}
}
}
}
void BufferedAudioSourceThread::run()
{
std::vector<float> tempBuffer;
while(1)
{
mReadMore.wait();
if(shouldThreadExit())
break;
mLock.lock();
for(UInt32 i = 0; i < mSources.size(); ++i)
{
BufferedAudioSource *pSource = mSources[i];
pSource->mLock.lock();
Int64 totalFrames = pSource->getSampleRate() * SECONDS_TO_BUFFER;
Int64 availableFrames = pSource->mBuffer.size() / pSource->getNumChannels();
pSource->mLock.unlock();
if(availableFrames < totalFrames)
{
Int64 numFrames = totalFrames - availableFrames;
Int64 framesToRead = numFrames;
Int64 frames = 0;
tempBuffer.resize(numFrames * pSource->getNumChannels());
float *pBuffer = &tempBuffer[0];
do
{
frames = pSource->decodeData(pBuffer, framesToRead);
framesToRead -= frames;
pBuffer += frames * pSource->getNumChannels();
}
while(framesToRead > 0 && !pSource->isEOF());
// Now write the data back to the buffer
Int64 framesRead = numFrames - framesToRead;
if(framesRead > 0)
{
pSource->mLock.lock();
int fillPosition = pSource->mBuffer.size();
Int64 totalSamples = pSource->mBuffer.size() + framesRead * pSource->getNumChannels();
pSource->mBuffer.resize(totalSamples);
memcpy(&pSource->mBuffer[fillPosition], &tempBuffer[0], sizeof(float) * framesRead * pSource->getNumChannels());
pSource->mLock.unlock();
}
// Check if we've reached the end
/*
double loopStart;
double loopEnd;
pSource->getLoopPoints(loopStart, loopEnd);
Int64 loopEndFrame = pSource->convertSecondsToSamples(pSource->mLoopEnd);
if(pSource->isLooping() && pSource->mCurrentFrame >= loopEndFrame && loopEndFrame > 0)
pSource->setPosition(loopStart);
if(pSource->isEOF() || (pSource->mCurrentFrame >= loopEndFrame && loopEndFrame > 0))
{
if(pSource->isLooping())
// Reset the buffering position and keep on going...
pSource->setPosition(loopStart);
// else
// // We didn't fill the whole buffer, so resize appropriately
// pSource->mBuffer.resize(availableFrames + numFrames - framesToRead);
}
*/
}
}
mLock.unlock();
}
}
