long
nsBufferedAudioStream::DataCallback(void* aBuffer, long aFrames)
{
MonitorAutoLock mon(mMonitor);
uint32_t bytesWanted = aFrames * mBytesPerFrame;
// Adjust bytesWanted to fit what is available in mBuffer.
uint32_t available = NS_MIN(bytesWanted, mBuffer.Length());
NS_ABORT_IF_FALSE(available % mBytesPerFrame == 0, "Must copy complete frames");
if (available > 0) {
// Copy each sample from mBuffer to aBuffer, adjusting the volume during the copy.
float scaled_volume = float(GetVolumeScale() * mVolume);
// Fetch input pointers from the ring buffer.
void* input[2];
uint32_t input_size[2];
mBuffer.PopElements(available, &input[0], &input_size[0], &input[1], &input_size[1]);
uint8_t* output = static_cast<uint8_t*>(aBuffer);
for (int i = 0; i < 2; ++i) {
const AudioDataValue* src = static_cast<const AudioDataValue*>(input[i]);
AudioDataValue* dst = reinterpret_cast<AudioDataValue*>(output);
ConvertAudioSamplesWithScale(src, dst, input_size[i]/sizeof(AudioDataValue),
scaled_volume);
output += input_size[i];
}
NS_ABORT_IF_FALSE(mBuffer.Length() % mBytesPerFrame == 0, "Must copy complete frames");
// Notify any blocked Write() call that more space is available in mBuffer.
mon.NotifyAll();
// Calculate remaining bytes requested by caller. If the stream is not
// draining an underrun has occurred, so fill the remaining buffer with
// silence.
bytesWanted -= available;
}
if (mState != DRAINING) {
memset(static_cast<uint8_t*>(aBuffer) + available, 0, bytesWanted);
mLostFrames += bytesWanted / mBytesPerFrame;
bytesWanted = 0;
}
return aFrames - (bytesWanted / mBytesPerFrame);
}
long
BufferedAudioStream::GetUnprocessed(void* aBuffer, long aFrames)
{
uint8_t* wpos = reinterpret_cast<uint8_t*>(aBuffer);
// Flush the timestretcher pipeline, if we were playing using a playback rate
// other than 1.0.
uint32_t flushedFrames = 0;
if (mTimeStretcher && mTimeStretcher->numSamples()) {
flushedFrames = mTimeStretcher->receiveSamples(reinterpret_cast<AudioDataValue*>(wpos), aFrames);
wpos += FramesToBytes(flushedFrames);
}
uint32_t toPopBytes = FramesToBytes(aFrames - flushedFrames);
uint32_t available = std::min(toPopBytes, mBuffer.Length());
void* input[2];
uint32_t input_size[2];
mBuffer.PopElements(available, &input[0], &input_size[0], &input[1], &input_size[1]);
memcpy(wpos, input[0], input_size[0]);
wpos += input_size[0];
memcpy(wpos, input[1], input_size[1]);
return BytesToFrames(available) + flushedFrames;
}
long
BufferedAudioStream::GetTimeStretched(void* aBuffer, long aFrames)
{
long processedFrames = 0;
// We need to call the non-locking version, because we already have the lock.
if (AudioStream::EnsureTimeStretcherInitialized() != NS_OK) {
return 0;
}
uint8_t* wpos = reinterpret_cast<uint8_t*>(aBuffer);
double playbackRate = static_cast<double>(mInRate) / mOutRate;
uint32_t toPopBytes = FramesToBytes(ceil(aFrames / playbackRate));
uint32_t available = 0;
bool lowOnBufferedData = false;
do {
// Check if we already have enough data in the time stretcher pipeline.
if (mTimeStretcher->numSamples() <= static_cast<uint32_t>(aFrames)) {
void* input[2];
uint32_t input_size[2];
available = std::min(mBuffer.Length(), toPopBytes);
if (available != toPopBytes) {
lowOnBufferedData = true;
}
mBuffer.PopElements(available, &input[0], &input_size[0],
&input[1], &input_size[1]);
for(uint32_t i = 0; i < 2; i++) {
mTimeStretcher->putSamples(reinterpret_cast<AudioDataValue*>(input[i]), BytesToFrames(input_size[i]));
}
}
uint32_t receivedFrames = mTimeStretcher->receiveSamples(reinterpret_cast<AudioDataValue*>(wpos), aFrames - processedFrames);
wpos += FramesToBytes(receivedFrames);
processedFrames += receivedFrames;
} while (processedFrames < aFrames && !lowOnBufferedData);
return processedFrames;
}
