nsresult AudioStream::SetPreservesPitch(bool aPreservesPitch)
{
// MUST lock since the rate transposer is used from the cubeb callback,
// and rate changes can cause the buffer to be reallocated
MonitorAutoLock mon(mMonitor);
// Avoid instantiating the timestretcher instance if not needed.
if (aPreservesPitch == mAudioClock.GetPreservesPitch()) {
return NS_OK;
}
if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
return NS_ERROR_FAILURE;
}
if (aPreservesPitch == true) {
mTimeStretcher->setTempo(mAudioClock.GetPlaybackRate());
mTimeStretcher->setRate(1.0f);
} else {
mTimeStretcher->setTempo(1.0f);
mTimeStretcher->setRate(mAudioClock.GetPlaybackRate());
}
mAudioClock.SetPreservesPitch(aPreservesPitch);
return NS_OK;
}
nsresult AudioStream::SetPlaybackRate(double aPlaybackRate)
{
// MUST lock since the rate transposer is used from the cubeb callback,
// and rate changes can cause the buffer to be reallocated
MonitorAutoLock mon(mMonitor);
NS_ASSERTION(aPlaybackRate > 0.0,
"Can't handle negative or null playbackrate in the AudioStream.");
// Avoid instantiating the resampler if we are not changing the playback rate.
// GetPreservesPitch/SetPreservesPitch don't need locking before calling
if (aPlaybackRate == mAudioClock.GetPlaybackRate()) {
return NS_OK;
}
if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
return NS_ERROR_FAILURE;
}
mAudioClock.SetPlaybackRateUnlocked(aPlaybackRate);
mOutRate = mInRate / aPlaybackRate;
if (mAudioClock.GetPreservesPitch()) {
mTimeStretcher->setTempo(aPlaybackRate);
mTimeStretcher->setRate(1.0f);
} else {
mTimeStretcher->setTempo(1.0f);
mTimeStretcher->setRate(aPlaybackRate);
}
return NS_OK;
}
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());
}
long
AudioStream::GetTimeStretched(void* aBuffer, long aFrames, int64_t &aTimeMs)
{
mMonitor.AssertCurrentThreadOwns();
long processedFrames = 0;
// We need to call the non-locking version, because we already have the lock.
if (EnsureTimeStretcherInitializedUnlocked() != 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]);
mReadPoint += BytesToFrames(available);
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);
GetBufferInsertTime(aTimeMs);
return processedFrames;
}
nsresult AudioStream::EnsureTimeStretcherInitialized()
{
MonitorAutoLock mon(mMonitor);
return EnsureTimeStretcherInitializedUnlocked();
}
