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(©Chunk); 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); } } 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)); } } } } }