C++ AudioSegment::GetDuration方法代码示例

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(&copyChunk);
    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(&copyChunk);
    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(&copyChunk);
    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(&copyChunk);
    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());
//.........这里部分代码省略.........