C++ AudioChunk::SetNull方法代码示例

void
AudioNodeStream::ObtainInputBlock(AudioChunk& aTmpChunk, uint32_t aPortIndex)
{
  uint32_t inputCount = mInputs.Length();
  uint32_t outputChannelCount = 1;
  nsAutoTArray<AudioChunk*,250> inputChunks;
  for (uint32_t i = 0; i < inputCount; ++i) {
    if (aPortIndex != mInputs[i]->InputNumber()) {
      // This input is connected to a different port
      continue;
    }
    MediaStream* s = mInputs[i]->GetSource();
    AudioNodeStream* a = static_cast<AudioNodeStream*>(s);
    MOZ_ASSERT(a == s->AsAudioNodeStream());
    if (a->IsAudioParamStream()) {
      continue;
    }

    AudioChunk* chunk = &a->mLastChunks[mInputs[i]->OutputNumber()];
    MOZ_ASSERT(chunk);
    if (chunk->IsNull() || chunk->mChannelData.IsEmpty()) {
      continue;
    }

    inputChunks.AppendElement(chunk);
    outputChannelCount =
      GetAudioChannelsSuperset(outputChannelCount, chunk->mChannelData.Length());
  }

  outputChannelCount = ComputedNumberOfChannels(outputChannelCount);

  uint32_t inputChunkCount = inputChunks.Length();
  if (inputChunkCount == 0 ||
      (inputChunkCount == 1 && inputChunks[0]->mChannelData.Length() == 0)) {
    aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
    return;
  }

  if (inputChunkCount == 1 &&
      inputChunks[0]->mChannelData.Length() == outputChannelCount) {
    aTmpChunk = *inputChunks[0];
    return;
  }

  if (outputChannelCount == 0) {
    aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
    return;
  }

  AllocateAudioBlock(outputChannelCount, &aTmpChunk);
  // The static storage here should be 1KB, so it's fine
  nsAutoTArray<float, GUESS_AUDIO_CHANNELS*WEBAUDIO_BLOCK_SIZE> downmixBuffer;

  for (uint32_t i = 0; i < inputChunkCount; ++i) {
    AccumulateInputChunk(i, *inputChunks[i], &aTmpChunk, &downmixBuffer);
  }
}

      NS_IMETHOD Run() override
      {

        auto engine =
          static_cast<ScriptProcessorNodeEngine*>(mStream->Engine());
        AudioChunk output;
        output.SetNull(engine->mBufferSize);
        {
          auto node = static_cast<ScriptProcessorNode*>
            (engine->NodeMainThread());
          if (!node) {
            return NS_OK;
          }

          if (node->HasListenersFor(nsGkAtoms::onaudioprocess)) {
            DispatchAudioProcessEvent(node, &output);
          }
          // The node may have been destroyed during event dispatch.
        }

        // Append it to our output buffer queue
        engine->GetSharedBuffers()->FinishProducingOutputBuffer(output);

        return NS_OK;
      }

// 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);
  }
}

  // graph thread
  AudioChunk GetOutputBuffer()
  {
    MOZ_ASSERT(!NS_IsMainThread());
    AudioChunk buffer;

    {
      MutexAutoLock lock(mOutputQueue.Lock());
      if (mOutputQueue.ReadyToConsume() > 0) {
        if (mDelaySoFar == STREAM_TIME_MAX) {
          mDelaySoFar = 0;
        }
        buffer = mOutputQueue.Consume();
      } else {
        // If we're out of buffers to consume, just output silence
        buffer.SetNull(WEBAUDIO_BLOCK_SIZE);
        if (mDelaySoFar != STREAM_TIME_MAX) {
          // Remember the delay that we just hit
          mDelaySoFar += WEBAUDIO_BLOCK_SIZE;
        }
      }
    }

    return buffer;
  }

void
AudioNodeStream::ObtainInputBlock(AudioChunk& aTmpChunk, uint32_t aPortIndex)
{
  uint32_t inputCount = mInputs.Length();
  uint32_t outputChannelCount = 1;
  nsAutoTArray<AudioChunk*,250> inputChunks;
  for (uint32_t i = 0; i < inputCount; ++i) {
    if (aPortIndex != mInputs[i]->InputNumber()) {
      // This input is connected to a different port
      continue;
    }
    MediaStream* s = mInputs[i]->GetSource();
    AudioNodeStream* a = static_cast<AudioNodeStream*>(s);
    MOZ_ASSERT(a == s->AsAudioNodeStream());
    if (a->IsAudioParamStream()) {
      continue;
    }

    // It is possible for mLastChunks to be empty here, because `a` might be a
    // AudioNodeStream that has not been scheduled yet, because it is further
    // down the graph _but_ as a connection to this node. Because we enforce the
    // presence of at least one DelayNode, with at least one block of delay, and
    // because the output of a DelayNode when it has been fed less that
    // `delayTime` amount of audio is silence, we can simply continue here,
    // because this input would not influence the output of this node. Next
    // iteration, a->mLastChunks.IsEmpty() will be false, and everthing will
    // work as usual.
    if (a->mLastChunks.IsEmpty()) {
      continue;
    }

    AudioChunk* chunk = &a->mLastChunks[mInputs[i]->OutputNumber()];
    MOZ_ASSERT(chunk);
    if (chunk->IsNull() || chunk->mChannelData.IsEmpty()) {
      continue;
    }

    inputChunks.AppendElement(chunk);
    outputChannelCount =
      GetAudioChannelsSuperset(outputChannelCount, chunk->mChannelData.Length());
  }

  outputChannelCount = ComputedNumberOfChannels(outputChannelCount);

  uint32_t inputChunkCount = inputChunks.Length();
  if (inputChunkCount == 0 ||
      (inputChunkCount == 1 && inputChunks[0]->mChannelData.Length() == 0)) {
    aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
    return;
  }

  if (inputChunkCount == 1 &&
      inputChunks[0]->mChannelData.Length() == outputChannelCount) {
    aTmpChunk = *inputChunks[0];
    return;
  }

  if (outputChannelCount == 0) {
    aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
    return;
  }

  AllocateAudioBlock(outputChannelCount, &aTmpChunk);
  // The static storage here should be 1KB, so it's fine
  nsAutoTArray<float, GUESS_AUDIO_CHANNELS*WEBAUDIO_BLOCK_SIZE> downmixBuffer;

  for (uint32_t i = 0; i < inputChunkCount; ++i) {
    AccumulateInputChunk(i, *inputChunks[i], &aTmpChunk, &downmixBuffer);
  }
}

void
AudioNodeStream::ObtainInputBlock(AudioChunk& aTmpChunk, uint32_t aPortIndex)
{
  uint32_t inputCount = mInputs.Length();
  uint32_t outputChannelCount = 1;
  nsAutoTArray<AudioChunk*,250> inputChunks;
  for (uint32_t i = 0; i < inputCount; ++i) {
    if (aPortIndex != mInputs[i]->InputNumber()) {
      // This input is connected to a different port
      continue;
    }
    MediaStream* s = mInputs[i]->GetSource();
    AudioNodeStream* a = static_cast<AudioNodeStream*>(s);
    MOZ_ASSERT(a == s->AsAudioNodeStream());
    if (a->IsFinishedOnGraphThread() ||
        a->IsAudioParamStream()) {
      continue;
    }
    AudioChunk* chunk = &a->mLastChunks[mInputs[i]->OutputNumber()];
    MOZ_ASSERT(chunk);
    if (chunk->IsNull()) {
      continue;
    }

    inputChunks.AppendElement(chunk);
    outputChannelCount =
      GetAudioChannelsSuperset(outputChannelCount, chunk->mChannelData.Length());
  }

  switch (mChannelCountMode) {
  case ChannelCountMode::Explicit:
    // Disregard the output channel count that we've calculated, and just use
    // mNumberOfInputChannels.
    outputChannelCount = mNumberOfInputChannels;
    break;
  case ChannelCountMode::Clamped_max:
    // Clamp the computed output channel count to mNumberOfInputChannels.
    outputChannelCount = std::min(outputChannelCount, mNumberOfInputChannels);
    break;
  case ChannelCountMode::Max:
    // Nothing to do here, just shut up the compiler warning.
    break;
  }

  uint32_t inputChunkCount = inputChunks.Length();
  if (inputChunkCount == 0 ||
      (inputChunkCount == 1 && inputChunks[0]->mChannelData.Length() == 0)) {
    aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
    return;
  }

  if (inputChunkCount == 1 &&
      inputChunks[0]->mChannelData.Length() == outputChannelCount) {
    aTmpChunk = *inputChunks[0];
    return;
  }

  AllocateAudioBlock(outputChannelCount, &aTmpChunk);
  float silenceChannel[WEBAUDIO_BLOCK_SIZE] = {0.f};
  // The static storage here should be 1KB, so it's fine
  nsAutoTArray<float, GUESS_AUDIO_CHANNELS*WEBAUDIO_BLOCK_SIZE> downmixBuffer;

  for (uint32_t i = 0; i < inputChunkCount; ++i) {
    AudioChunk* chunk = inputChunks[i];
    nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> channels;
    channels.AppendElements(chunk->mChannelData);
    if (channels.Length() < outputChannelCount) {
      if (mChannelInterpretation == ChannelInterpretation::Speakers) {
        AudioChannelsUpMix(&channels, outputChannelCount, nullptr);
        NS_ASSERTION(outputChannelCount == channels.Length(),
                     "We called GetAudioChannelsSuperset to avoid this");
      } else {
        // Fill up the remaining channels by zeros
        for (uint32_t j = channels.Length(); j < outputChannelCount; ++j) {
          channels.AppendElement(silenceChannel);
        }
      }
    } else if (channels.Length() > outputChannelCount) {
      if (mChannelInterpretation == ChannelInterpretation::Speakers) {
        nsAutoTArray<float*,GUESS_AUDIO_CHANNELS> outputChannels;
        outputChannels.SetLength(outputChannelCount);
        downmixBuffer.SetLength(outputChannelCount * WEBAUDIO_BLOCK_SIZE);
        for (uint32_t j = 0; j < outputChannelCount; ++j) {
          outputChannels[j] = &downmixBuffer[j * WEBAUDIO_BLOCK_SIZE];
        }

        AudioChannelsDownMix(channels, outputChannels.Elements(),
                             outputChannelCount, WEBAUDIO_BLOCK_SIZE);

        channels.SetLength(outputChannelCount);
        for (uint32_t j = 0; j < channels.Length(); ++j) {
          channels[j] = outputChannels[j];
        }
      } else {
        // Drop the remaining channels
        channels.RemoveElementsAt(outputChannelCount,
                                  channels.Length() - outputChannelCount);
      }
    }

//.........这里部分代码省略.........