void
AudioStream::GetUnprocessed(AudioBufferWriter& aWriter)
{
mMonitor.AssertCurrentThreadOwns();
// Flush the timestretcher pipeline, if we were playing using a playback rate
// other than 1.0.
if (mTimeStretcher && mTimeStretcher->numSamples()) {
auto timeStretcher = mTimeStretcher;
aWriter.Write([timeStretcher] (AudioDataValue* aPtr, uint32_t aFrames) {
return timeStretcher->receiveSamples(aPtr, aFrames);
}, aWriter.Available());
// TODO: There might be still unprocessed samples in the stretcher.
// We should either remove or flush them so they won't be in the output
// next time we switch a playback rate other than 1.0.
NS_WARN_IF(mTimeStretcher->numUnprocessedSamples() > 0);
}
while (aWriter.Available() > 0) {
UniquePtr<Chunk> c = mDataSource.PopFrames(aWriter.Available());
if (c->Frames() == 0) {
break;
}
MOZ_ASSERT(c->Frames() <= aWriter.Available());
if (Downmix(c.get())) {
aWriter.Write(c->Data(), c->Frames());
} else {
// Write silence if downmixing fails.
aWriter.WriteZeros(c->Frames());
}
}
}
void
AudioStream::GetTimeStretched(AudioBufferWriter& aWriter)
{
mMonitor.AssertCurrentThreadOwns();
// We need to call the non-locking version, because we already have the lock.
if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
return;
}
double playbackRate = static_cast<double>(mInRate) / mOutRate;
uint32_t toPopFrames = ceil(aWriter.Available() * playbackRate);
while (mTimeStretcher->numSamples() < aWriter.Available()) {
UniquePtr<Chunk> c = mDataSource.PopFrames(toPopFrames);
if (c->Frames() == 0) {
break;
}
MOZ_ASSERT(c->Frames() <= toPopFrames);
if (Downmix(c.get())) {
mTimeStretcher->putSamples(c->Data(), c->Frames());
} else {
// Write silence if downmixing fails.
nsAutoTArray<AudioDataValue, 1000> buf;
buf.SetLength(mOutChannels * c->Frames());
memset(buf.Elements(), 0, buf.Length() * sizeof(AudioDataValue));
mTimeStretcher->putSamples(buf.Elements(), c->Frames());
}
}
auto timeStretcher = mTimeStretcher;
aWriter.Write([timeStretcher] (AudioDataValue* aPtr, uint32_t aFrames) {
return timeStretcher->receiveSamples(aPtr, aFrames);
}, aWriter.Available());
}
void
AudioStream::GetTimeStretched(AudioBufferWriter& aWriter)
{
mMonitor.AssertCurrentThreadOwns();
// We need to call the non-locking version, because we already have the lock.
if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
return;
}
uint32_t toPopFrames =
ceil(aWriter.Available() * mAudioClock.GetPlaybackRate());
while (mTimeStretcher->numSamples() < aWriter.Available()) {
UniquePtr<Chunk> c = mDataSource.PopFrames(toPopFrames);
if (c->Frames() == 0) {
break;
}
MOZ_ASSERT(c->Frames() <= toPopFrames);
if (IsValidAudioFormat(c.get())) {
mTimeStretcher->putSamples(c->Data(), c->Frames());
} else {
// Write silence if invalid format.
AutoTArray<AudioDataValue, 1000> buf;
auto size = CheckedUint32(mOutChannels) * c->Frames();
if (!size.isValid()) {
// The overflow should not happen in normal case.
LOGW("Invalid member data: %d channels, %d frames", mOutChannels, c->Frames());
return;
}
buf.SetLength(size.value());
size = size * sizeof(AudioDataValue);
if (!size.isValid()) {
LOGW("The required memory size is too large.");
return;
}
memset(buf.Elements(), 0, size.value());
mTimeStretcher->putSamples(buf.Elements(), c->Frames());
}
}
auto timeStretcher = mTimeStretcher;
aWriter.Write([timeStretcher] (AudioDataValue* aPtr, uint32_t aFrames) {
return timeStretcher->receiveSamples(aPtr, aFrames);
}, aWriter.Available());
}