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,
nullptr,
mVorbisDsp.vi->channels,
mVorbisDsp.vi->rate));
}
while (frames > 0) {
uint32_t channels = mVorbisDsp.vi->channels;
auto buffer = MakeUnique<AudioDataValue[]>(frames*channels);
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, mVorbisDsp.vi->rate);
if (!duration.isValid()) {
NS_WARNING("Int overflow converting WebM audio duration");
return -1;
}
CheckedInt64 total_duration = FramesToUsecs(mFrames,
mVorbisDsp.vi->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;
};
aTotalFrames += frames;
mCallback->Output(new AudioData(aOffset,
time.value(),
duration.value(),
frames,
Move(buffer),
mVorbisDsp.vi->channels,
mVorbisDsp.vi->rate));
mFrames += frames;
if (vorbis_synthesis_read(&mVorbisDsp, frames) != 0) {
return -1;
}
frames = vorbis_synthesis_pcmout(&mVorbisDsp, &pcm);
}
return aTotalFrames > 0 ? 1 : 0;
}
RefPtr<MediaDataDecoder::DecodePromise>
VorbisDataDecoder::ProcessDecode(MediaRawData* aSample)
{
MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
const unsigned char* aData = aSample->Data();
size_t aLength = aSample->Size();
int64_t aOffset = aSample->mOffset;
MOZ_ASSERT(mPacketCount >= 3);
if (!mLastFrameTime ||
mLastFrameTime.ref() != aSample->mTime.ToMicroseconds()) {
// We are starting a new block.
mFrames = 0;
mLastFrameTime = Some(aSample->mTime.ToMicroseconds());
}
ogg_packet pkt = InitVorbisPacket(
aData, aLength, false, aSample->mEOS,
aSample->mTimecode.ToMicroseconds(), mPacketCount++);
int err = vorbis_synthesis(&mVorbisBlock, &pkt);
if (err) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
RESULT_DETAIL("vorbis_synthesis:%d", err)),
__func__);
}
err = vorbis_synthesis_blockin(&mVorbisDsp, &mVorbisBlock);
if (err) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
RESULT_DETAIL("vorbis_synthesis_blockin:%d", err)),
__func__);
}
VorbisPCMValue** pcm = 0;
int32_t frames = vorbis_synthesis_pcmout(&mVorbisDsp, &pcm);
if (frames == 0) {
return DecodePromise::CreateAndResolve(DecodedData(), __func__);
}
DecodedData results;
while (frames > 0) {
uint32_t channels = mVorbisDsp.vi->channels;
uint32_t rate = mVorbisDsp.vi->rate;
AlignedAudioBuffer buffer(frames*channels);
if (!buffer) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_OUT_OF_MEMORY, __func__), __func__);
}
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]);
}
}
auto duration = FramesToTimeUnit(frames, rate);
if (!duration.IsValid()) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_OVERFLOW_ERR,
RESULT_DETAIL("Overflow converting audio duration")),
__func__);
}
auto total_duration = FramesToTimeUnit(mFrames, rate);
if (!total_duration.IsValid()) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_OVERFLOW_ERR,
RESULT_DETAIL("Overflow converting audio total_duration")),
__func__);
}
auto time = total_duration + aSample->mTime;
if (!time.IsValid()) {
return DecodePromise::CreateAndReject(
MediaResult(
NS_ERROR_DOM_MEDIA_OVERFLOW_ERR,
RESULT_DETAIL("Overflow adding total_duration and aSample->mTime")),
__func__);
};
if (!mAudioConverter) {
AudioConfig in(
AudioConfig::ChannelLayout(channels, VorbisLayout(channels)), rate);
AudioConfig out(channels, rate);
if (!in.IsValid() || !out.IsValid()) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_FATAL_ERR,
RESULT_DETAIL("Invalid channel layout:%u", channels)),
__func__);
}
mAudioConverter = MakeUnique<AudioConverter>(in, out);
}
MOZ_ASSERT(mAudioConverter->CanWorkInPlace());
AudioSampleBuffer data(Move(buffer));
data = mAudioConverter->Process(Move(data));
results.AppendElement(new AudioData(aOffset, time, duration,
frames, data.Forget(), channels, rate));
mFrames += frames;
err = vorbis_synthesis_read(&mVorbisDsp, frames);
if (err) {
return DecodePromise::CreateAndReject(
MediaResult(NS_ERROR_DOM_MEDIA_DECODE_ERR,
RESULT_DETAIL("vorbis_synthesis_read:%d", err)),
__func__);
}
frames = vorbis_synthesis_pcmout(&mVorbisDsp, &pcm);
}
return DecodePromise::CreateAndResolve(Move(results), __func__);
}
