C++ MediaStream::AsAudioNodeStream方法代码示例

void
AudioNodeStream::ObtainInputBlock(AudioBlock& aTmpChunk,
                                  uint32_t aPortIndex)
{
  uint32_t inputCount = mInputs.Length();
  uint32_t outputChannelCount = 1;
  nsAutoTArray<const AudioBlock*,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;
    }

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

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

  outputChannelCount = ComputedNumberOfChannels(outputChannelCount);

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

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

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

  aTmpChunk.AllocateChannels(outputChannelCount);
  // 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);
  }
}

already_AddRefed<Promise>
AudioContext::Close(ErrorResult& aRv)
{
  nsCOMPtr<nsIGlobalObject> parentObject = do_QueryInterface(GetParentObject());
  RefPtr<Promise> promise;
  promise = Promise::Create(parentObject, aRv);
  if (aRv.Failed()) {
    return nullptr;
  }

  if (mIsOffline) {
    promise->MaybeReject(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
    return promise.forget();
  }

  if (mAudioContextState == AudioContextState::Closed) {
    promise->MaybeResolve(NS_ERROR_DOM_INVALID_STATE_ERR);
    return promise.forget();
  }

  if (Destination()) {
    Destination()->DestroyAudioChannelAgent();
  }

  mPromiseGripArray.AppendElement(promise);

  // This can be called when freeing a document, and the streams are dead at
  // this point, so we need extra null-checks.
  MediaStream* ds = DestinationStream();
  if (ds) {
    nsTArray<MediaStream*> streams;
    // If mSuspendCalled or mCloseCalled are true then we already suspended
    // all our streams, so don't suspend them again. But we still need to do
    // ApplyAudioContextOperation to ensure our new promise is resolved.
    if (!mSuspendCalled && !mCloseCalled) {
      streams = GetAllStreams();
    }
    Graph()->ApplyAudioContextOperation(ds->AsAudioNodeStream(), streams,
                                        AudioContextOperation::Close, promise);
  }
  mCloseCalled = true;

  return promise.forget();
}

already_AddRefed<Promise>
AudioContext::Close(ErrorResult& aRv)
{
  nsCOMPtr<nsIGlobalObject> parentObject = do_QueryInterface(GetParentObject());
  nsRefPtr<Promise> promise;
  promise = Promise::Create(parentObject, aRv);
  if (aRv.Failed()) {
    return nullptr;
  }

  if (mIsOffline) {
    promise->MaybeReject(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
    return promise.forget();
  }

  if (mAudioContextState == AudioContextState::Closed) {
    promise->MaybeResolve(NS_ERROR_DOM_INVALID_STATE_ERR);
    return promise.forget();
  }

  mCloseCalled = true;

  if (Destination()) {
    Destination()->DestroyAudioChannelAgent();
  }

  mPromiseGripArray.AppendElement(promise);

  // This can be called when freeing a document, and the streams are dead at
  // this point, so we need extra null-checks.
  MediaStream* ds = DestinationStream();
  if (ds) {
    Graph()->ApplyAudioContextOperation(ds->AsAudioNodeStream(),
                                        AudioContextOperation::Close, promise);

    if (ds) {
      ds->BlockStreamIfNeeded();
    }
  }
  return promise.forget();
}

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);
  }
}

AudioChunk*
AudioNodeStream::ObtainInputBlock(AudioChunk* aTmpChunk)
{
  uint32_t inputCount = mInputs.Length();
  uint32_t outputChannelCount = 0;
  nsAutoTArray<AudioChunk*,250> inputChunks;
  for (uint32_t i = 0; i < inputCount; ++i) {
    MediaStream* s = mInputs[i]->GetSource();
    AudioNodeStream* a = static_cast<AudioNodeStream*>(s);
    MOZ_ASSERT(a == s->AsAudioNodeStream());
    if (a->IsFinishedOnGraphThread()) {
      continue;
    }
    AudioChunk* chunk = &a->mLastChunk;
    // XXX when we implement DelayNode, this will no longer be true and we'll
    // need to treat a null chunk (when the DelayNode hasn't had a chance
    // to produce data yet) as silence here.
    MOZ_ASSERT(chunk);
    if (chunk->IsNull()) {
      continue;
    }

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

  uint32_t inputChunkCount = inputChunks.Length();
  if (inputChunkCount == 0) {
    aTmpChunk->SetNull(WEBAUDIO_BLOCK_SIZE);
    return aTmpChunk;
  }

  if (inputChunkCount == 1) {
    return inputChunks[0];
  }

  AllocateAudioBlock(outputChannelCount, aTmpChunk);

  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) {
      AudioChannelsUpMix(&channels, outputChannelCount, nullptr);
      NS_ASSERTION(outputChannelCount == channels.Length(),
                   "We called GetAudioChannelsSuperset to avoid this");
    }

    for (uint32_t c = 0; c < channels.Length(); ++c) {
      const float* inputData = static_cast<const float*>(channels[c]);
      float* outputData = static_cast<float*>(const_cast<void*>(aTmpChunk->mChannelData[c]));
      if (inputData) {
        if (i == 0) {
          AudioBlockCopyChannelWithScale(inputData, chunk->mVolume, outputData);
        } else {
          AudioBlockAddChannelWithScale(inputData, chunk->mVolume, outputData);
        }
      } else {
        if (i == 0) {
          memset(outputData, 0, WEBAUDIO_BLOCK_SIZE*sizeof(float));
        }
      }
    }
  }

  return aTmpChunk;
}

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.........