Пример #1
0
void
AudioCaptureStream::ProcessInput(GraphTime aFrom, GraphTime aTo,
                                 uint32_t aFlags)
{
  uint32_t inputCount = mInputs.Length();
  StreamBuffer::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, MediaStreamListener::TRACK_EVENT_CREATED, tmp);
      l->NotifyFinishedTrackCreation(Graph());
    }
    mTrackCreated = true;
  }

  // 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 (mFinished || 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();
      StreamBuffer::TrackIter tracks(s->GetStreamBuffer(), MediaSegment::AUDIO);
      while (!tracks.IsEnded()) {
        AudioSegment* inputSegment = tracks->Get<AudioSegment>();
        StreamTime inputStart = s->GraphTimeToStreamTimeWithBlocking(aFrom);
        StreamTime inputEnd = s->GraphTimeToStreamTimeWithBlocking(aTo);
        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.
  mBuffer.AdvanceKnownTracksTime(GraphTimeToStreamTimeWithBlocking((aTo)));
}
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();
}