/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */ /* vim:set ts=2 sw=2 sts=2 et cindent: */ /* This Source Code Form is subject to the terms of the Mozilla Public * License, v. 2.0. If a copy of the MPL was not distributed with this * file, You can obtain one at http://mozilla.org/MPL/2.0/. */ #include "DelayNode.h" #include "mozilla/dom/DelayNodeBinding.h" #include "AudioNodeEngine.h" #include "AudioNodeStream.h" #include "AudioDestinationNode.h" #include "WebAudioUtils.h" #include "DelayProcessor.h" #include "PlayingRefChangeHandler.h" namespace mozilla { namespace dom { NS_IMPL_CYCLE_COLLECTION_INHERITED_1(DelayNode, AudioNode, mDelay) NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(DelayNode) NS_INTERFACE_MAP_END_INHERITING(AudioNode) NS_IMPL_ADDREF_INHERITED(DelayNode, AudioNode) NS_IMPL_RELEASE_INHERITED(DelayNode, AudioNode) class DelayNodeEngine : public AudioNodeEngine { typedef PlayingRefChangeHandler PlayingRefChanged; public: DelayNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination, int aMaxDelayFrames) : AudioNodeEngine(aNode) , mSource(nullptr) , mDestination(static_cast (aDestination->Stream())) // Keep the default value in sync with the default value in DelayNode::DelayNode. , mDelay(0.f) // Use a smoothing range of 20ms , mProcessor(aMaxDelayFrames, WebAudioUtils::ComputeSmoothingRate(0.02, mDestination->SampleRate())) , mLeftOverData(INT32_MIN) { } virtual DelayNodeEngine* AsDelayNodeEngine() { return this; } void SetSourceStream(AudioNodeStream* aSource) { mSource = aSource; } enum Parameters { DELAY, }; void SetTimelineParameter(uint32_t aIndex, const AudioParamTimeline& aValue, TrackRate aSampleRate) MOZ_OVERRIDE { switch (aIndex) { case DELAY: MOZ_ASSERT(mSource && mDestination); mDelay = aValue; WebAudioUtils::ConvertAudioParamToTicks(mDelay, mSource, mDestination); break; default: NS_ERROR("Bad DelayNodeEngine TimelineParameter"); } } virtual void ProduceAudioBlock(AudioNodeStream* aStream, const AudioChunk& aInput, AudioChunk* aOutput, bool* aFinished) { MOZ_ASSERT(mSource == aStream, "Invalid source stream"); MOZ_ASSERT(aStream->SampleRate() == mDestination->SampleRate()); const uint32_t numChannels = aInput.IsNull() ? mProcessor.BufferChannelCount() : aInput.mChannelData.Length(); if (!aInput.IsNull()) { if (mLeftOverData <= 0) { nsRefPtr refchanged = new PlayingRefChanged(aStream, PlayingRefChanged::ADDREF); aStream->Graph()-> DispatchToMainThreadAfterStreamStateUpdate(refchanged.forget()); } mLeftOverData = mProcessor.MaxDelayFrames(); } else if (mLeftOverData > 0) { mLeftOverData -= WEBAUDIO_BLOCK_SIZE; } else { if (mLeftOverData != INT32_MIN) { mLeftOverData = INT32_MIN; // Delete our buffered data now we no longer need it mProcessor.Reset(); nsRefPtr refchanged = new PlayingRefChanged(aStream, PlayingRefChanged::RELEASE); aStream->Graph()-> DispatchToMainThreadAfterStreamStateUpdate(refchanged.forget()); } *aOutput = aInput; return; } AllocateAudioBlock(numChannels, aOutput); AudioChunk input = aInput; if (!aInput.IsNull() && aInput.mVolume != 1.0f) { // Pre-multiply the input's volume AllocateAudioBlock(numChannels, &input); for (uint32_t i = 0; i < numChannels; ++i) { const float* src = static_cast(aInput.mChannelData[i]); float* dest = static_cast(const_cast(input.mChannelData[i])); AudioBlockCopyChannelWithScale(src, aInput.mVolume, dest); } } const float* const* inputChannels = input.IsNull() ? nullptr : reinterpret_cast(input.mChannelData.Elements()); float* const* outputChannels = reinterpret_cast (const_cast(aOutput->mChannelData.Elements())); bool inCycle = aStream->AsProcessedStream()->InCycle(); double sampleRate = aStream->SampleRate(); if (mDelay.HasSimpleValue()) { // If this DelayNode is in a cycle, make sure the delay value is at least // one block. float delayFrames = mDelay.GetValue() * sampleRate; float delayFramesClamped = inCycle ? std::max(static_cast(WEBAUDIO_BLOCK_SIZE), delayFrames) : delayFrames; mProcessor.Process(delayFramesClamped, inputChannels, outputChannels, numChannels, WEBAUDIO_BLOCK_SIZE); } else { // Compute the delay values for the duration of the input AudioChunk // If this DelayNode is in a cycle, make sure the delay value is at least // one block. double computedDelay[WEBAUDIO_BLOCK_SIZE]; TrackTicks tick = aStream->GetCurrentPosition(); for (size_t counter = 0; counter < WEBAUDIO_BLOCK_SIZE; ++counter) { float delayAtTick = mDelay.GetValueAtTime(tick, counter) * sampleRate; float delayAtTickClamped = inCycle ? std::max(static_cast(WEBAUDIO_BLOCK_SIZE), delayAtTick) : delayAtTick; computedDelay[counter] = delayAtTickClamped; } mProcessor.Process(computedDelay, inputChannels, outputChannels, numChannels, WEBAUDIO_BLOCK_SIZE); } } AudioNodeStream* mSource; AudioNodeStream* mDestination; AudioParamTimeline mDelay; DelayProcessor mProcessor; // How much data we have in our buffer which needs to be flushed out when our inputs // finish. int32_t mLeftOverData; }; DelayNode::DelayNode(AudioContext* aContext, double aMaxDelay) : AudioNode(aContext, 2, ChannelCountMode::Max, ChannelInterpretation::Speakers) , mDelay(new AudioParam(MOZ_THIS_IN_INITIALIZER_LIST(), SendDelayToStream, 0.0f)) { DelayNodeEngine* engine = new DelayNodeEngine(this, aContext->Destination(), ceil(aContext->SampleRate() * aMaxDelay)); mStream = aContext->Graph()->CreateAudioNodeStream(engine, MediaStreamGraph::INTERNAL_STREAM); engine->SetSourceStream(static_cast (mStream.get())); } JSObject* DelayNode::WrapObject(JSContext* aCx, JS::Handle aScope) { return DelayNodeBinding::Wrap(aCx, aScope, this); } void DelayNode::SendDelayToStream(AudioNode* aNode) { DelayNode* This = static_cast(aNode); SendTimelineParameterToStream(This, DelayNodeEngine::DELAY, *This->mDelay); } } }