nsresult
OmxAudioTrackEncoder::GetEncodedTrack(EncodedFrameContainer& aData)
{
PROFILER_LABEL("OmxAACAudioTrackEncoder", "GetEncodedTrack",
js::ProfileEntry::Category::OTHER);
AudioSegment segment;
// Move all the samples from mRawSegment to segment. We only hold
// the monitor in this block.
{
ReentrantMonitorAutoEnter mon(mReentrantMonitor);
// Wait if mEncoder is not initialized nor canceled.
while (!mInitialized && !mCanceled) {
mReentrantMonitor.Wait();
}
if (mCanceled || mEncodingComplete) {
return NS_ERROR_FAILURE;
}
segment.AppendFrom(&mRawSegment);
}
nsresult rv;
if (segment.GetDuration() == 0) {
// Notify EOS at least once, even if segment is empty.
if (mEndOfStream && !mEosSetInEncoder) {
mEosSetInEncoder = true;
rv = mEncoder->Encode(segment, OMXCodecWrapper::BUFFER_EOS);
NS_ENSURE_SUCCESS(rv, rv);
}
// Nothing to encode but encoder could still have encoded data for earlier
// input.
return AppendEncodedFrames(aData);
}
// OMX encoder has limited input buffers only so we have to feed input and get
// output more than once if there are too many samples pending in segment.
while (segment.GetDuration() > 0) {
rv = mEncoder->Encode(segment,
mEndOfStream ? OMXCodecWrapper::BUFFER_EOS : 0);
NS_ENSURE_SUCCESS(rv, rv);
rv = AppendEncodedFrames(aData);
NS_ENSURE_SUCCESS(rv, rv);
}
return NS_OK;
}
// The MediaStreamGraph guarantees that this is actually one block, for
// AudioNodeStreams.
void
AudioNodeStream::ProduceOutput(GraphTime aFrom, GraphTime aTo)
{
if (mMarkAsFinishedAfterThisBlock) {
// This stream was finished the last time that we looked at it, and all
// of the depending streams have finished their output as well, so now
// it's time to mark this stream as finished.
FinishOutput();
}
StreamBuffer::Track* track = EnsureTrack();
AudioSegment* segment = track->Get<AudioSegment>();
mLastChunks.SetLength(1);
mLastChunks[0].SetNull(0);
if (mInCycle) {
// XXX DelayNode not supported yet so just produce silence
mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
} else {
// We need to generate at least one input
uint16_t maxInputs = std::max(uint16_t(1), mEngine->InputCount());
OutputChunks inputChunks;
inputChunks.SetLength(maxInputs);
for (uint16_t i = 0; i < maxInputs; ++i) {
ObtainInputBlock(inputChunks[i], i);
}
bool finished = false;
if (maxInputs <= 1 && mEngine->OutputCount() <= 1) {
mEngine->ProduceAudioBlock(this, inputChunks[0], &mLastChunks[0], &finished);
} else {
mEngine->ProduceAudioBlocksOnPorts(this, inputChunks, mLastChunks, &finished);
}
if (finished) {
mMarkAsFinishedAfterThisBlock = true;
}
}
if (mKind == MediaStreamGraph::EXTERNAL_STREAM) {
segment->AppendAndConsumeChunk(&mLastChunks[0]);
} else {
segment->AppendNullData(mLastChunks[0].GetDuration());
}
for (uint32_t j = 0; j < mListeners.Length(); ++j) {
MediaStreamListener* l = mListeners[j];
AudioChunk copyChunk = mLastChunks[0];
AudioSegment tmpSegment;
tmpSegment.AppendAndConsumeChunk(©Chunk);
l->NotifyQueuedTrackChanges(Graph(), AUDIO_NODE_STREAM_TRACK_ID,
IdealAudioRate(), segment->GetDuration(), 0,
tmpSegment);
}
}
void
DecodedStream::SendAudio(double aVolume, bool aIsSameOrigin,
const PrincipalHandle& aPrincipalHandle)
{
AssertOwnerThread();
if (!mInfo.HasAudio()) {
return;
}
AudioSegment output;
uint32_t rate = mInfo.mAudio.mRate;
AutoTArray<RefPtr<MediaData>,10> audio;
TrackID audioTrackId = mInfo.mAudio.mTrackId;
SourceMediaStream* sourceStream = mData->mStream;
// It's OK to hold references to the AudioData because AudioData
// is ref-counted.
mAudioQueue.GetElementsAfter(mData->mNextAudioTime, &audio);
for (uint32_t i = 0; i < audio.Length(); ++i) {
SendStreamAudio(mData.get(), mStartTime.ref(), audio[i], &output, rate,
aPrincipalHandle);
}
output.ApplyVolume(aVolume);
if (!aIsSameOrigin) {
output.ReplaceWithDisabled();
}
// |mNextAudioTime| is updated as we process each audio sample in
// SendStreamAudio(). This is consistent with how |mNextVideoTime|
// is updated for video samples.
if (output.GetDuration() > 0) {
sourceStream->AppendToTrack(audioTrackId, &output);
}
if (mAudioQueue.IsFinished() && !mData->mHaveSentFinishAudio) {
sourceStream->EndTrack(audioTrackId);
mData->mHaveSentFinishAudio = true;
}
}
void
AudioNodeStream::AdvanceOutputSegment()
{
StreamBuffer::Track* track = EnsureTrack(AUDIO_TRACK);
AudioSegment* segment = track->Get<AudioSegment>();
if (mKind == MediaStreamGraph::EXTERNAL_STREAM) {
segment->AppendAndConsumeChunk(&mLastChunks[0]);
} else {
segment->AppendNullData(mLastChunks[0].GetDuration());
}
for (uint32_t j = 0; j < mListeners.Length(); ++j) {
MediaStreamListener* l = mListeners[j];
AudioChunk copyChunk = mLastChunks[0];
AudioSegment tmpSegment;
tmpSegment.AppendAndConsumeChunk(©Chunk);
l->NotifyQueuedTrackChanges(Graph(), AUDIO_TRACK,
segment->GetDuration(), 0, tmpSegment);
}
}
// The MediaStreamGraph guarantees that this is actually one block, for
// AudioNodeStreams.
void
AudioNodeStream::ProduceOutput(GraphTime aFrom, GraphTime aTo)
{
StreamBuffer::Track* track = EnsureTrack();
AudioChunk outputChunk;
AudioSegment* segment = track->Get<AudioSegment>();
outputChunk.SetNull(0);
if (mInCycle) {
// XXX DelayNode not supported yet so just produce silence
outputChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
} else {
AudioChunk tmpChunk;
AudioChunk* inputChunk = ObtainInputBlock(&tmpChunk);
bool finished = false;
mEngine->ProduceAudioBlock(this, *inputChunk, &outputChunk, &finished);
if (finished) {
FinishOutput();
}
}
mLastChunk = outputChunk;
if (mKind == MediaStreamGraph::EXTERNAL_STREAM) {
segment->AppendAndConsumeChunk(&outputChunk);
} else {
segment->AppendNullData(outputChunk.GetDuration());
}
for (uint32_t j = 0; j < mListeners.Length(); ++j) {
MediaStreamListener* l = mListeners[j];
AudioChunk copyChunk = outputChunk;
AudioSegment tmpSegment;
tmpSegment.AppendAndConsumeChunk(©Chunk);
l->NotifyQueuedTrackChanges(Graph(), AUDIO_NODE_STREAM_TRACK_ID,
IdealAudioRate(), segment->GetDuration(), 0,
tmpSegment);
}
}
void
AudioNodeStream::AdvanceOutputSegment()
{
StreamBuffer::Track* track = EnsureTrack(AUDIO_TRACK);
// No more tracks will be coming
mBuffer.AdvanceKnownTracksTime(STREAM_TIME_MAX);
AudioSegment* segment = track->Get<AudioSegment>();
if (!mLastChunks[0].IsNull()) {
segment->AppendAndConsumeChunk(mLastChunks[0].AsMutableChunk());
} else {
segment->AppendNullData(mLastChunks[0].GetDuration());
}
for (uint32_t j = 0; j < mListeners.Length(); ++j) {
MediaStreamListener* l = mListeners[j];
AudioChunk copyChunk = mLastChunks[0].AsAudioChunk();
AudioSegment tmpSegment;
tmpSegment.AppendAndConsumeChunk(©Chunk);
l->NotifyQueuedTrackChanges(Graph(), AUDIO_TRACK,
segment->GetDuration(), 0, tmpSegment);
}
}
TEST(OpusAudioTrackEncoder, Init)
{
{
// The encoder does not normally recieve enough info from null data to
// init. However, multiple attempts to do so, with sufficiently long
// duration segments, should result in a best effort attempt. The first
// attempt should never do this though, even if the duration is long:
OpusTrackEncoder encoder(48000);
AudioSegment segment;
segment.AppendNullData(48000 * 100);
encoder.TryInit(segment, segment.GetDuration());
EXPECT_FALSE(encoder.IsInitialized());
// Multiple init attempts should result in best effort init:
encoder.TryInit(segment, segment.GetDuration());
EXPECT_TRUE(encoder.IsInitialized());
}
{
// If the duration of the segments given to the encoder is not long then
// we shouldn't try a best effort init:
OpusTrackEncoder encoder(48000);
AudioSegment segment;
segment.AppendNullData(1);
encoder.TryInit(segment, segment.GetDuration());
EXPECT_FALSE(encoder.IsInitialized());
encoder.TryInit(segment, segment.GetDuration());
EXPECT_FALSE(encoder.IsInitialized());
}
{
// For non-null segments we should init immediately
OpusTrackEncoder encoder(48000);
AudioSegment segment;
AudioGenerator generator(2, 48000);
generator.Generate(segment, 1);
encoder.TryInit(segment, segment.GetDuration());
EXPECT_TRUE(encoder.IsInitialized());
}
{
// Test low sample rate bound
OpusTrackEncoder encoder(7999);
AudioSegment segment;
AudioGenerator generator(2, 7999);
generator.Generate(segment, 1);
encoder.TryInit(segment, segment.GetDuration());
EXPECT_FALSE(encoder.IsInitialized());
}
{
// Test low sample rate bound
OpusTrackEncoder encoder(8000);
AudioSegment segment;
AudioGenerator generator(2, 8000);
generator.Generate(segment, 1);
encoder.TryInit(segment, segment.GetDuration());
EXPECT_TRUE(encoder.IsInitialized());
}
{
// Test high sample rate bound
OpusTrackEncoder encoder(192001);
AudioSegment segment;
AudioGenerator generator(2, 192001);
generator.Generate(segment, 1);
encoder.TryInit(segment, segment.GetDuration());
EXPECT_FALSE(encoder.IsInitialized());
}
{
// Test high sample rate bound
OpusTrackEncoder encoder(192000);
AudioSegment segment;
AudioGenerator generator(2, 192000);
generator.Generate(segment, 1);
encoder.TryInit(segment, segment.GetDuration());
EXPECT_TRUE(encoder.IsInitialized());
}
}
void
AudioCaptureStream::ProcessInput(GraphTime aFrom, GraphTime aTo,
uint32_t aFlags)
{
if (!mStarted) {
return;
}
uint32_t inputCount = mInputs.Length();
StreamTracks::Track* track = EnsureTrack(mTrackId);
// Notify the DOM everything is in order.
if (!mTrackCreated) {
for (uint32_t i = 0; i < mListeners.Length(); i++) {
MediaStreamListener* l = mListeners[i];
AudioSegment tmp;
l->NotifyQueuedTrackChanges(
Graph(), mTrackId, 0, TrackEventCommand::TRACK_EVENT_CREATED, tmp);
l->NotifyFinishedTrackCreation(Graph());
}
mTrackCreated = true;
}
if (IsFinishedOnGraphThread()) {
return;
}
// If the captured stream is connected back to a object on the page (be it an
// HTMLMediaElement with a stream as source, or an AudioContext), a cycle
// situation occur. This can work if it's an AudioContext with at least one
// DelayNode, but the MSG will mute the whole cycle otherwise.
if (InMutedCycle() || inputCount == 0) {
track->Get<AudioSegment>()->AppendNullData(aTo - aFrom);
} else {
// We mix down all the tracks of all inputs, to a stereo track. Everything
// is {up,down}-mixed to stereo.
mMixer.StartMixing();
AudioSegment output;
for (uint32_t i = 0; i < inputCount; i++) {
MediaStream* s = mInputs[i]->GetSource();
StreamTracks::TrackIter tracks(s->GetStreamTracks(), MediaSegment::AUDIO);
while (!tracks.IsEnded()) {
AudioSegment* inputSegment = tracks->Get<AudioSegment>();
StreamTime inputStart = s->GraphTimeToStreamTimeWithBlocking(aFrom);
StreamTime inputEnd = s->GraphTimeToStreamTimeWithBlocking(aTo);
if (tracks->IsEnded() && inputSegment->GetDuration() <= inputEnd) {
// If the input track has ended and we have consumed all its data it
// can be ignored.
continue;
}
AudioSegment toMix;
toMix.AppendSlice(*inputSegment, inputStart, inputEnd);
// Care for streams blocked in the [aTo, aFrom] range.
if (inputEnd - inputStart < aTo - aFrom) {
toMix.AppendNullData((aTo - aFrom) - (inputEnd - inputStart));
}
toMix.Mix(mMixer, MONO, Graph()->GraphRate());
tracks.Next();
}
}
// This calls MixerCallback below
mMixer.FinishMixing();
}
// Regardless of the status of the input tracks, we go foward.
mTracks.AdvanceKnownTracksTime(GraphTimeToStreamTimeWithBlocking((aTo)));
}
nsresult
OMXAudioEncoder::Encode(AudioSegment& aSegment, int aInputFlags,
bool* aSendEOS)
{
#ifndef MOZ_SAMPLE_TYPE_S16
#error MediaCodec accepts only 16-bit PCM data.
#endif
MOZ_ASSERT(mStarted, "Configure() should be called before Encode().");
size_t numSamples = aSegment.GetDuration();
// Get input buffer.
InputBufferHelper buffer(mCodec, mInputBufs, *this, aInputFlags);
status_t result = buffer.Dequeue();
if (result == -EAGAIN) {
// All input buffers are full. Caller can try again later after consuming
// some output buffers.
return NS_OK;
}
NS_ENSURE_TRUE(result == OK, NS_ERROR_FAILURE);
size_t sourceSamplesCopied = 0; // Number of copied samples.
if (numSamples > 0) {
// Copy input PCM data to input buffer until queue is empty.
AudioSegment::ChunkIterator iter(const_cast<AudioSegment&>(aSegment));
while (!iter.IsEnded()) {
BufferState result = buffer.ReadChunk(*iter, &sourceSamplesCopied);
if (result == WAIT_FOR_NEW_BUFFER) {
// All input buffers are full. Caller can try again later after
// consuming some output buffers.
aSegment.RemoveLeading(sourceSamplesCopied);
return NS_OK;
} else if (result == BUFFER_FAIL) {
return NS_ERROR_FAILURE;
} else {
iter.Next();
}
}
// Remove the samples already been copied into buffer
if (sourceSamplesCopied > 0) {
aSegment.RemoveLeading(sourceSamplesCopied);
}
} else if (aInputFlags & BUFFER_EOS) {
buffer.SendEOSToBuffer(&sourceSamplesCopied);
}
// Enqueue the remaining data to buffer
MOZ_ASSERT(sourceSamplesCopied > 0, "No data needs to be enqueued!");
int flags = aInputFlags;
if (aSegment.GetDuration() > 0) {
// Don't signal EOS until source segment is empty.
flags &= ~BUFFER_EOS;
}
result = buffer.Enqueue(mTimestamp, flags);
NS_ENSURE_TRUE(result == OK, NS_ERROR_FAILURE);
if (aSendEOS && (aInputFlags & BUFFER_EOS)) {
*aSendEOS = true;
}
return NS_OK;
}
void
AudioNodeExternalInputStream::ProcessInput(GraphTime aFrom, GraphTime aTo,
uint32_t aFlags)
{
// According to spec, number of outputs is always 1.
mLastChunks.SetLength(1);
// GC stuff can result in our input stream being destroyed before this stream.
// Handle that.
if (mInputs.IsEmpty()) {
mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
AdvanceOutputSegment();
return;
}
MOZ_ASSERT(mInputs.Length() == 1);
MediaStream* source = mInputs[0]->GetSource();
nsAutoTArray<AudioSegment,1> audioSegments;
nsAutoTArray<bool,1> trackMapEntriesUsed;
uint32_t inputChannels = 0;
for (StreamBuffer::TrackIter tracks(source->mBuffer, MediaSegment::AUDIO);
!tracks.IsEnded(); tracks.Next()) {
const StreamBuffer::Track& inputTrack = *tracks;
// Create a TrackMapEntry if necessary.
size_t trackMapIndex = GetTrackMapEntry(inputTrack, aFrom);
// Maybe there's nothing in this track yet. If so, ignore it. (While the
// track is only playing silence, we may not be able to determine the
// correct number of channels to start resampling.)
if (trackMapIndex == nsTArray<TrackMapEntry>::NoIndex) {
continue;
}
while (trackMapEntriesUsed.Length() <= trackMapIndex) {
trackMapEntriesUsed.AppendElement(false);
}
trackMapEntriesUsed[trackMapIndex] = true;
TrackMapEntry* trackMap = &mTrackMap[trackMapIndex];
AudioSegment segment;
GraphTime next;
TrackRate inputTrackRate = inputTrack.GetRate();
for (GraphTime t = aFrom; t < aTo; t = next) {
MediaInputPort::InputInterval interval = mInputs[0]->GetNextInputInterval(t);
interval.mEnd = std::min(interval.mEnd, aTo);
if (interval.mStart >= interval.mEnd)
break;
next = interval.mEnd;
// Ticks >= startTicks and < endTicks are in the interval
StreamTime outputEnd = GraphTimeToStreamTime(interval.mEnd);
TrackTicks startTicks = trackMap->mSamplesPassedToResampler + segment.GetDuration();
StreamTime outputStart = GraphTimeToStreamTime(interval.mStart);
NS_ASSERTION(startTicks == TimeToTicksRoundUp(inputTrackRate, outputStart),
"Samples missing");
TrackTicks endTicks = TimeToTicksRoundUp(inputTrackRate, outputEnd);
TrackTicks ticks = endTicks - startTicks;
if (interval.mInputIsBlocked) {
segment.AppendNullData(ticks);
} else {
// See comments in TrackUnionStream::CopyTrackData
StreamTime inputStart = source->GraphTimeToStreamTime(interval.mStart);
StreamTime inputEnd = source->GraphTimeToStreamTime(interval.mEnd);
TrackTicks inputTrackEndPoint =
inputTrack.IsEnded() ? inputTrack.GetEnd() : TRACK_TICKS_MAX;
if (trackMap->mEndOfLastInputIntervalInInputStream != inputStart ||
trackMap->mEndOfLastInputIntervalInOutputStream != outputStart) {
// Start of a new series of intervals where neither stream is blocked.
trackMap->mEndOfConsumedInputTicks = TimeToTicksRoundDown(inputTrackRate, inputStart) - 1;
}
TrackTicks inputStartTicks = trackMap->mEndOfConsumedInputTicks;
TrackTicks inputEndTicks = inputStartTicks + ticks;
trackMap->mEndOfConsumedInputTicks = inputEndTicks;
trackMap->mEndOfLastInputIntervalInInputStream = inputEnd;
trackMap->mEndOfLastInputIntervalInOutputStream = outputEnd;
if (inputStartTicks < 0) {
// Data before the start of the track is just null.
segment.AppendNullData(-inputStartTicks);
inputStartTicks = 0;
}
if (inputEndTicks > inputStartTicks) {
segment.AppendSlice(*inputTrack.GetSegment(),
std::min(inputTrackEndPoint, inputStartTicks),
std::min(inputTrackEndPoint, inputEndTicks));
}
// Pad if we're looking past the end of the track
segment.AppendNullData(ticks - segment.GetDuration());
}
}
trackMap->mSamplesPassedToResampler += segment.GetDuration();
trackMap->ResampleInputData(&segment);
if (trackMap->mResampledData.GetDuration() < mCurrentOutputPosition + WEBAUDIO_BLOCK_SIZE) {
// We don't have enough data. Delay it.
trackMap->mResampledData.InsertNullDataAtStart(
mCurrentOutputPosition + WEBAUDIO_BLOCK_SIZE - trackMap->mResampledData.GetDuration());
}
audioSegments.AppendElement()->AppendSlice(trackMap->mResampledData,
mCurrentOutputPosition, mCurrentOutputPosition + WEBAUDIO_BLOCK_SIZE);
trackMap->mResampledData.ForgetUpTo(mCurrentOutputPosition + WEBAUDIO_BLOCK_SIZE);
inputChannels = GetAudioChannelsSuperset(inputChannels, trackMap->mResamplerChannelCount);
}
for (int32_t i = mTrackMap.Length() - 1; i >= 0; --i) {
if (i >= int32_t(trackMapEntriesUsed.Length()) || !trackMapEntriesUsed[i]) {
mTrackMap.RemoveElementAt(i);
}
}
uint32_t accumulateIndex = 0;
if (inputChannels) {
nsAutoTArray<float,GUESS_AUDIO_CHANNELS*WEBAUDIO_BLOCK_SIZE> downmixBuffer;
for (uint32_t i = 0; i < audioSegments.Length(); ++i) {
AudioChunk tmpChunk;
ConvertSegmentToAudioBlock(&audioSegments[i], &tmpChunk);
if (!tmpChunk.IsNull()) {
if (accumulateIndex == 0) {
AllocateAudioBlock(inputChannels, &mLastChunks[0]);
}
AccumulateInputChunk(accumulateIndex, tmpChunk, &mLastChunks[0], &downmixBuffer);
accumulateIndex++;
}
}
}
if (accumulateIndex == 0) {
mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
}
mCurrentOutputPosition += WEBAUDIO_BLOCK_SIZE;
// Using AudioNodeStream's AdvanceOutputSegment to push the media stream graph along with null data.
AdvanceOutputSegment();
}