/* -*- 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 "AudioContext.h" #include "nsContentUtils.h" #include "nsPIDOMWindow.h" #include "mozilla/ErrorResult.h" #include "mozilla/dom/AudioContextBinding.h" #include "mozilla/dom/OfflineAudioContextBinding.h" #include "MediaStreamGraph.h" #include "mozilla/dom/AnalyserNode.h" #include "AudioDestinationNode.h" #include "AudioBufferSourceNode.h" #include "AudioBuffer.h" #include "GainNode.h" #include "DelayNode.h" #include "PannerNode.h" #include "AudioListener.h" #include "DynamicsCompressorNode.h" #include "BiquadFilterNode.h" #include "ScriptProcessorNode.h" #include "ChannelMergerNode.h" #include "ChannelSplitterNode.h" #include "MediaStreamAudioDestinationNode.h" #include "WaveShaperNode.h" #include "PeriodicWave.h" #include "ConvolverNode.h" #include "nsNetUtil.h" namespace mozilla { namespace dom { NS_IMPL_CYCLE_COLLECTION_INHERITED_2(AudioContext, nsDOMEventTargetHelper, mDestination, mListener) NS_IMPL_ADDREF_INHERITED(AudioContext, nsDOMEventTargetHelper) NS_IMPL_RELEASE_INHERITED(AudioContext, nsDOMEventTargetHelper) NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(AudioContext) NS_INTERFACE_MAP_END_INHERITING(nsDOMEventTargetHelper) static uint8_t gWebAudioOutputKey; AudioContext::AudioContext(nsPIDOMWindow* aWindow, bool aIsOffline, uint32_t aNumberOfChannels, uint32_t aLength, float aSampleRate) : mSampleRate(aIsOffline ? aSampleRate : IdealAudioRate()) , mDestination(new AudioDestinationNode(MOZ_THIS_IN_INITIALIZER_LIST(), aIsOffline, aNumberOfChannels, aLength, aSampleRate)) , mNumberOfChannels(aNumberOfChannels) , mIsOffline(aIsOffline) { // Actually play audio mDestination->Stream()->AddAudioOutput(&gWebAudioOutputKey); nsDOMEventTargetHelper::BindToOwner(aWindow); SetIsDOMBinding(); mPannerNodes.Init(); mAudioBufferSourceNodes.Init(); mScriptProcessorNodes.Init(); } AudioContext::~AudioContext() { } JSObject* AudioContext::WrapObject(JSContext* aCx, JS::Handle aScope) { if (mIsOffline) { return OfflineAudioContextBinding::Wrap(aCx, aScope, this); } else { return AudioContextBinding::Wrap(aCx, aScope, this); } } /* static */ already_AddRefed AudioContext::Constructor(const GlobalObject& aGlobal, ErrorResult& aRv) { nsCOMPtr window = do_QueryInterface(aGlobal.Get()); if (!window) { aRv.Throw(NS_ERROR_FAILURE); return nullptr; } nsRefPtr object = new AudioContext(window, false); window->AddAudioContext(object); return object.forget(); } /* static */ already_AddRefed AudioContext::Constructor(const GlobalObject& aGlobal, uint32_t aNumberOfChannels, uint32_t aLength, float aSampleRate, ErrorResult& aRv) { nsCOMPtr window = do_QueryInterface(aGlobal.Get()); if (!window) { aRv.Throw(NS_ERROR_FAILURE); return nullptr; } if (aNumberOfChannels == 0 || aNumberOfChannels > WebAudioUtils::MaxChannelCount || aLength == 0 || aSampleRate <= 1.0f || aSampleRate >= TRACK_RATE_MAX) { // The DOM binding protects us against infinity and NaN aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR); return nullptr; } nsRefPtr object = new AudioContext(window, true, aNumberOfChannels, aLength, aSampleRate); window->AddAudioContext(object); return object.forget(); } already_AddRefed AudioContext::CreateBufferSource() { nsRefPtr bufferNode = new AudioBufferSourceNode(this); mAudioBufferSourceNodes.PutEntry(bufferNode); return bufferNode.forget(); } already_AddRefed AudioContext::CreateBuffer(JSContext* aJSContext, uint32_t aNumberOfChannels, uint32_t aLength, float aSampleRate, ErrorResult& aRv) { if (aSampleRate < 8000 || aSampleRate > 96000 || !aLength) { aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR); return nullptr; } if (aLength > INT32_MAX) { aRv.Throw(NS_ERROR_OUT_OF_MEMORY); return nullptr; } nsRefPtr buffer = new AudioBuffer(this, int32_t(aLength), aSampleRate); if (!buffer->InitializeBuffers(aNumberOfChannels, aJSContext)) { aRv.Throw(NS_ERROR_OUT_OF_MEMORY); return nullptr; } return buffer.forget(); } already_AddRefed AudioContext::CreateBuffer(JSContext* aJSContext, ArrayBuffer& aBuffer, bool aMixToMono, ErrorResult& aRv) { // Sniff the content of the media. // Failed type sniffing will be handled by SyncDecodeMedia. nsAutoCString contentType; NS_SniffContent(NS_DATA_SNIFFER_CATEGORY, nullptr, aBuffer.Data(), aBuffer.Length(), contentType); WebAudioDecodeJob job(contentType, this); if (mDecoder.SyncDecodeMedia(contentType.get(), aBuffer.Data(), aBuffer.Length(), job) && job.mOutput) { nsRefPtr buffer = job.mOutput.forget(); if (aMixToMono) { buffer->MixToMono(aJSContext); } return buffer.forget(); } return nullptr; } namespace { bool IsValidBufferSize(uint32_t aBufferSize) { switch (aBufferSize) { case 0: // let the implementation choose the buffer size case 256: case 512: case 1024: case 2048: case 4096: case 8192: case 16384: return true; default: return false; } } } already_AddRefed AudioContext::CreateMediaStreamDestination() { nsRefPtr node = new MediaStreamAudioDestinationNode(this); return node.forget(); } already_AddRefed AudioContext::CreateScriptProcessor(uint32_t aBufferSize, uint32_t aNumberOfInputChannels, uint32_t aNumberOfOutputChannels, ErrorResult& aRv) { if ((aNumberOfInputChannels == 0 && aNumberOfOutputChannels == 0) || aNumberOfInputChannels > WebAudioUtils::MaxChannelCount || aNumberOfOutputChannels > WebAudioUtils::MaxChannelCount || !IsValidBufferSize(aBufferSize)) { aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR); return nullptr; } nsRefPtr scriptProcessor = new ScriptProcessorNode(this, aBufferSize, aNumberOfInputChannels, aNumberOfOutputChannels); mScriptProcessorNodes.PutEntry(scriptProcessor); return scriptProcessor.forget(); } already_AddRefed AudioContext::CreateAnalyser() { nsRefPtr analyserNode = new AnalyserNode(this); return analyserNode.forget(); } already_AddRefed AudioContext::CreateGain() { nsRefPtr gainNode = new GainNode(this); return gainNode.forget(); } already_AddRefed AudioContext::CreateWaveShaper() { nsRefPtr waveShaperNode = new WaveShaperNode(this); return waveShaperNode.forget(); } already_AddRefed AudioContext::CreateDelay(double aMaxDelayTime, ErrorResult& aRv) { if (aMaxDelayTime > 0. && aMaxDelayTime < 180.) { nsRefPtr delayNode = new DelayNode(this, aMaxDelayTime); return delayNode.forget(); } aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR); return nullptr; } already_AddRefed AudioContext::CreatePanner() { nsRefPtr pannerNode = new PannerNode(this); mPannerNodes.PutEntry(pannerNode); return pannerNode.forget(); } already_AddRefed AudioContext::CreateConvolver() { nsRefPtr convolverNode = new ConvolverNode(this); return convolverNode.forget(); } already_AddRefed AudioContext::CreateChannelSplitter(uint32_t aNumberOfOutputs, ErrorResult& aRv) { if (aNumberOfOutputs == 0 || aNumberOfOutputs > WebAudioUtils::MaxChannelCount) { aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR); return nullptr; } nsRefPtr splitterNode = new ChannelSplitterNode(this, aNumberOfOutputs); return splitterNode.forget(); } already_AddRefed AudioContext::CreateChannelMerger(uint32_t aNumberOfInputs, ErrorResult& aRv) { if (aNumberOfInputs == 0 || aNumberOfInputs > WebAudioUtils::MaxChannelCount) { aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR); return nullptr; } nsRefPtr mergerNode = new ChannelMergerNode(this, aNumberOfInputs); return mergerNode.forget(); } already_AddRefed AudioContext::CreateDynamicsCompressor() { nsRefPtr compressorNode = new DynamicsCompressorNode(this); return compressorNode.forget(); } already_AddRefed AudioContext::CreateBiquadFilter() { nsRefPtr filterNode = new BiquadFilterNode(this); return filterNode.forget(); } already_AddRefed AudioContext::CreatePeriodicWave(const Float32Array& aRealData, const Float32Array& aImagData, ErrorResult& aRv) { if (aRealData.Length() != aImagData.Length() || aRealData.Length() == 0 || aRealData.Length() > 4096) { aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR); return nullptr; } nsRefPtr periodicWave = new PeriodicWave(this, aRealData.Data(), aRealData.Length(), aImagData.Data(), aImagData.Length()); return periodicWave.forget(); } AudioListener* AudioContext::Listener() { if (!mListener) { mListener = new AudioListener(this); } return mListener; } void AudioContext::DecodeAudioData(const ArrayBuffer& aBuffer, DecodeSuccessCallback& aSuccessCallback, const Optional >& aFailureCallback) { // Sniff the content of the media. // Failed type sniffing will be handled by AsyncDecodeMedia. nsAutoCString contentType; NS_SniffContent(NS_DATA_SNIFFER_CATEGORY, nullptr, aBuffer.Data(), aBuffer.Length(), contentType); nsCOMPtr failureCallback; if (aFailureCallback.WasPassed()) { failureCallback = &aFailureCallback.Value(); } nsAutoPtr job( new WebAudioDecodeJob(contentType, this, &aSuccessCallback, failureCallback)); mDecoder.AsyncDecodeMedia(contentType.get(), aBuffer.Data(), aBuffer.Length(), *job); // Transfer the ownership to mDecodeJobs mDecodeJobs.AppendElement(job.forget()); } void AudioContext::RemoveFromDecodeQueue(WebAudioDecodeJob* aDecodeJob) { mDecodeJobs.RemoveElement(aDecodeJob); } void AudioContext::UnregisterAudioBufferSourceNode(AudioBufferSourceNode* aNode) { mAudioBufferSourceNodes.RemoveEntry(aNode); UpdatePannerSource(); } void AudioContext::UnregisterPannerNode(PannerNode* aNode) { mPannerNodes.RemoveEntry(aNode); } void AudioContext::UnregisterScriptProcessorNode(ScriptProcessorNode* aNode) { mScriptProcessorNodes.RemoveEntry(aNode); } static PLDHashOperator FindConnectedSourcesOn(nsPtrHashKey* aEntry, void* aData) { aEntry->GetKey()->FindConnectedSources(); return PL_DHASH_NEXT; } void AudioContext::UpdatePannerSource() { mPannerNodes.EnumerateEntries(FindConnectedSourcesOn, nullptr); } uint32_t AudioContext::MaxChannelCount() const { return mIsOffline ? mNumberOfChannels : AudioStream::MaxNumberOfChannels(); } MediaStreamGraph* AudioContext::Graph() const { return Destination()->Stream()->Graph(); } MediaStream* AudioContext::DestinationStream() const { return Destination()->Stream(); } double AudioContext::CurrentTime() const { return MediaTimeToSeconds(Destination()->Stream()->GetCurrentTime()); } template static PLDHashOperator GetHashtableEntry(nsPtrHashKey* aEntry, void* aData) { nsTArray* array = static_cast*>(aData); array->AppendElement(aEntry->GetKey()); return PL_DHASH_NEXT; } template static void GetHashtableElements(nsTHashtable >& aHashtable, nsTArray& aArray) { aHashtable.EnumerateEntries(&GetHashtableEntry, &aArray); } void AudioContext::Shutdown() { Suspend(); mDecoder.Shutdown(); // Stop all audio buffer source nodes, to make sure that they release // their self-references. // We first gather an array of the nodes and then call Stop on each one, // since Stop may delete the object and therefore trigger a re-entrant // hashtable call to remove the pointer from the hashtable, which is // not safe. nsTArray sourceNodes; GetHashtableElements(mAudioBufferSourceNodes, sourceNodes); for (uint32_t i = 0; i < sourceNodes.Length(); ++i) { ErrorResult rv; sourceNodes[i]->Stop(0.0, rv, true); } // Stop all script processor nodes, to make sure that they release // their self-references. nsTArray spNodes; GetHashtableElements(mScriptProcessorNodes, spNodes); for (uint32_t i = 0; i < spNodes.Length(); ++i) { spNodes[i]->Stop(); } // For offline contexts, we can destroy the MediaStreamGraph at this point. if (mIsOffline) { mDestination->DestroyGraph(); } } void AudioContext::Suspend() { MediaStream* ds = DestinationStream(); if (ds) { ds->ChangeExplicitBlockerCount(1); } } void AudioContext::Resume() { MediaStream* ds = DestinationStream(); if (ds) { ds->ChangeExplicitBlockerCount(-1); } } JSContext* AudioContext::GetJSContext() const { MOZ_ASSERT(NS_IsMainThread()); nsCOMPtr scriptGlobal = do_QueryInterface(GetParentObject()); if (!scriptGlobal) { return nullptr; } nsIScriptContext* scriptContext = scriptGlobal->GetContext(); if (!scriptContext) { return nullptr; } return scriptContext->GetNativeContext(); } void AudioContext::StartRendering() { MOZ_ASSERT(mIsOffline, "This should only be called on OfflineAudioContext"); mDestination->StartRendering(); } } }