C++ AudioNodeStream类代码示例

void
AudioBufferSourceNode::Stop(double aWhen, ErrorResult& aRv)
{
  if (!WebAudioUtils::IsTimeValid(aWhen)) {
    aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
    return;
  }

  if (!mStartCalled) {
    aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
    return;
  }

  AudioNodeStream* ns = mStream;
  if (!ns || !Context()) {
    // We've already stopped and had our stream shut down
    return;
  }

  ns->SetStreamTimeParameter(STOP, Context(), std::max(0.0, aWhen));
}

void
AudioNodeStream::SetActive()
{
    if (mIsActive || mMarkAsFinishedAfterThisBlock) {
        return;
    }

    mIsActive = true;
    if (IsAudioParamStream()) {
        // Consumers merely influence stream order.
        // They do not read from the stream.
        return;
    }

    for (const auto& consumer : mConsumers) {
        AudioNodeStream* ns = consumer->GetDestination()->AsAudioNodeStream();
        if (ns) {
            ns->IncrementActiveInputCount();
        }
    }
}

void
AudioNode::DestroyMediaStream()
{
  if (mStream) {
    // Remove the node pointer on the engine.
    AudioNodeStream* ns = mStream;
    MOZ_ASSERT(ns, "How come we don't have a stream here?");
    MOZ_ASSERT(ns->Engine()->NodeMainThread() == this,
               "Invalid node reference");
    ns->Engine()->ClearNode();

    mStream->Destroy();
    mStream = nullptr;

    nsCOMPtr<nsIObserverService> obs = services::GetObserverService();
    if (obs) {
      nsAutoString id;
      id.AppendPrintf("%u", mId);
      obs->NotifyObservers(nullptr, "webaudio-node-demise", id.get());
    }
  }
}

void
AudioBufferSourceNode::Stop(double aWhen, ErrorResult& aRv)
{
  if (!mStartCalled) {
    aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
    return;
  }

  if (!mBuffer) {
    // We don't have a buffer, so the stream is never marked as finished.
    // Therefore we need to drop our playing ref right now.
    mPlayingRef.Drop(this);
  }

  AudioNodeStream* ns = static_cast<AudioNodeStream*>(mStream.get());
  if (!ns || !Context()) {
    // We've already stopped and had our stream shut down
    return;
  }

  ns->SetStreamTimeParameter(STOP, Context()->DestinationStream(),
                             std::max(0.0, aWhen));
}

void
AudioBufferSourceNode::SendBufferParameterToStream(JSContext* aCx)
{
    AudioNodeStream* ns = mStream;
    if (!ns) {
        return;
    }

    if (mBuffer) {
        RefPtr<ThreadSharedFloatArrayBufferList> data =
            mBuffer->GetThreadSharedChannelsForRate(aCx);
        ns->SetBuffer(data.forget());

        if (mStartCalled) {
            SendOffsetAndDurationParametersToStream(ns);
        }
    } else {
        ns->SetInt32Parameter(BUFFEREND, 0);
        ns->SetBuffer(nullptr);

        MarkInactive();
    }
}

void
OscillatorNode::Stop(double aWhen, ErrorResult& aRv)
{
  if (!WebAudioUtils::IsTimeValid(aWhen)) {
    aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
    return;
  }

  if (!mStartCalled) {
    aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
    return;
  }

  AudioNodeStream* ns = static_cast<AudioNodeStream*>(mStream.get());
  if (!ns || !Context()) {
    // We've already stopped and had our stream shut down
    return;
  }

  // TODO: Perhaps we need to do more here.
  ns->SetStreamTimeParameter(OscillatorNodeEngine::STOP,
                             Context(), std::max(0.0, aWhen));
}

void
AudioBufferSourceNode::Start(double aWhen, double aOffset,
                             const Optional<double>& aDuration, ErrorResult& aRv)
{
  if (!WebAudioUtils::IsTimeValid(aWhen) ||
      (aDuration.WasPassed() && !WebAudioUtils::IsTimeValid(aDuration.Value()))) {
    aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
    return;
  }

  if (mStartCalled) {
    aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
    return;
  }
  mStartCalled = true;

  AudioNodeStream* ns = mStream;
  if (!ns) {
    // Nothing to play, or we're already dead for some reason
    return;
  }

  // Remember our arguments so that we can use them when we get a new buffer.
  mOffset = aOffset;
  mDuration = aDuration.WasPassed() ? aDuration.Value()
                                    : std::numeric_limits<double>::min();
  // We can't send these parameters without a buffer because we don't know the
  // buffer's sample rate or length.
  if (mBuffer) {
    SendOffsetAndDurationParametersToStream(ns);
  }

  // Don't set parameter unnecessarily
  if (aWhen > 0.0) {
    ns->SetDoubleParameter(START, mContext->DOMTimeToStreamTime(aWhen));
  }
}

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.........