uint32_t
AudioSink::DrainConverter(uint32_t aMaxFrames)
{
MOZ_ASSERT(mOwnerThread->IsCurrentThreadIn());
if (!mConverter || !mLastProcessedPacket || !aMaxFrames) {
// nothing to drain.
return 0;
}
RefPtr<AudioData> lastPacket = mLastProcessedPacket.ref();
mLastProcessedPacket.reset();
// To drain we simply provide an empty packet to the audio converter.
AlignedAudioBuffer convertedData =
mConverter->Process(AudioSampleBuffer(AlignedAudioBuffer())).Forget();
uint32_t frames = convertedData.Length() / mOutputChannels;
if (!convertedData.SetLength(std::min(frames, aMaxFrames) * mOutputChannels)) {
// This can never happen as we were reducing the length of convertData.
mErrored = true;
return 0;
}
RefPtr<AudioData> data =
CreateAudioFromBuffer(Move(convertedData), lastPacket);
if (!data) {
return 0;
}
mProcessedQueue.Push(data);
return data->mFrames;
}
int
VorbisDataDecoder::DoDecode(MediaRawData* aSample)
{
const unsigned char* aData = aSample->Data();
size_t aLength = aSample->Size();
int64_t aOffset = aSample->mOffset;
uint64_t aTstampUsecs = aSample->mTime;
int64_t aTotalFrames = 0;
MOZ_ASSERT(mPacketCount >= 3);
if (!mLastFrameTime || mLastFrameTime.ref() != aSample->mTime) {
// We are starting a new block.
mFrames = 0;
mLastFrameTime = Some(aSample->mTime);
}
ogg_packet pkt = InitVorbisPacket(aData, aLength, false, false, -1, mPacketCount++);
bool first_packet = mPacketCount == 4;
if (vorbis_synthesis(&mVorbisBlock, &pkt) != 0) {
return -1;
}
if (vorbis_synthesis_blockin(&mVorbisDsp,
&mVorbisBlock) != 0) {
return -1;
}
VorbisPCMValue** pcm = 0;
int32_t frames = vorbis_synthesis_pcmout(&mVorbisDsp, &pcm);
// If the first packet of audio in the media produces no data, we
// still need to produce an AudioData for it so that the correct media
// start time is calculated. Otherwise we'd end up with a media start
// time derived from the timecode of the first packet that produced
// data.
if (frames == 0 && first_packet) {
mCallback->Output(new AudioData(aOffset,
aTstampUsecs,
0,
0,
AlignedAudioBuffer(),
mVorbisDsp.vi->channels,
mVorbisDsp.vi->rate));
}
while (frames > 0) {
uint32_t channels = mVorbisDsp.vi->channels;
uint32_t rate = mVorbisDsp.vi->rate;
AlignedAudioBuffer buffer(frames*channels);
if (!buffer) {
return -1;
}
for (uint32_t j = 0; j < channels; ++j) {
VorbisPCMValue* channel = pcm[j];
for (uint32_t i = 0; i < uint32_t(frames); ++i) {
buffer[i*channels + j] = MOZ_CONVERT_VORBIS_SAMPLE(channel[i]);
}
}
CheckedInt64 duration = FramesToUsecs(frames, rate);
if (!duration.isValid()) {
NS_WARNING("Int overflow converting WebM audio duration");
return -1;
}
CheckedInt64 total_duration = FramesToUsecs(mFrames, rate);
if (!total_duration.isValid()) {
NS_WARNING("Int overflow converting WebM audio total_duration");
return -1;
}
CheckedInt64 time = total_duration + aTstampUsecs;
if (!time.isValid()) {
NS_WARNING("Int overflow adding total_duration and aTstampUsecs");
return -1;
};
if (!mAudioConverter) {
AudioConfig in(AudioConfig::ChannelLayout(channels, VorbisLayout(channels)),
rate);
AudioConfig out(channels, rate);
mAudioConverter = MakeUnique<AudioConverter>(in, out);
}
MOZ_ASSERT(mAudioConverter->CanWorkInPlace());
AudioSampleBuffer data(Move(buffer));
data = mAudioConverter->Process(Move(data));
aTotalFrames += frames;
mCallback->Output(new AudioData(aOffset,
time.value(),
duration.value(),
frames,
data.Forget(),
channels,
rate));
mFrames += frames;
if (vorbis_synthesis_read(&mVorbisDsp, frames) != 0) {
return -1;
}
frames = vorbis_synthesis_pcmout(&mVorbisDsp, &pcm);
}
return aTotalFrames > 0 ? 1 : 0;
}
