| /* |
| * Copyright (C) 2010 Google Inc. All rights reserved. |
| * Copyright (C) 2016-2019 Apple Inc. All rights reserved. |
| * |
| * Redistribution and use in source and binary forms, with or without |
| * modification, are permitted provided that the following conditions |
| * are met: |
| * 1. Redistributions of source code must retain the above copyright |
| * notice, this list of conditions and the following disclaimer. |
| * 2. Redistributions in binary form must reproduce the above copyright |
| * notice, this list of conditions and the following disclaimer in the |
| * documentation and/or other materials provided with the distribution. |
| * |
| * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY |
| * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED |
| * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE |
| * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY |
| * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES |
| * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; |
| * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON |
| * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT |
| * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS |
| * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
| */ |
| |
| #pragma once |
| |
| #include "ActiveDOMObject.h" |
| #include "AsyncAudioDecoder.h" |
| #include "AudioBus.h" |
| #include "AudioDestinationNode.h" |
| #include "EventTarget.h" |
| #include "MediaCanStartListener.h" |
| #include "MediaProducer.h" |
| #include "PlatformMediaSession.h" |
| #include "ScriptExecutionContext.h" |
| #include "VisibilityChangeClient.h" |
| #include <JavaScriptCore/ConsoleTypes.h> |
| #include <JavaScriptCore/Float32Array.h> |
| #include <atomic> |
| #include <wtf/HashSet.h> |
| #include <wtf/LoggerHelper.h> |
| #include <wtf/MainThread.h> |
| #include <wtf/RefPtr.h> |
| #include <wtf/ThreadSafeRefCounted.h> |
| #include <wtf/Threading.h> |
| #include <wtf/UniqueRef.h> |
| #include <wtf/Vector.h> |
| #include <wtf/text/AtomStringHash.h> |
| |
| namespace WebCore { |
| |
| class AnalyserNode; |
| class AudioBuffer; |
| class AudioBufferCallback; |
| class AudioBufferSourceNode; |
| class AudioListener; |
| class AudioSummingJunction; |
| class BiquadFilterNode; |
| class ChannelMergerNode; |
| class ChannelSplitterNode; |
| class ConvolverNode; |
| class DelayNode; |
| class Document; |
| class DynamicsCompressorNode; |
| class GainNode; |
| class HTMLMediaElement; |
| class MainThreadGenericEventQueue; |
| class MediaElementAudioSourceNode; |
| class MediaStream; |
| class MediaStreamAudioDestinationNode; |
| class MediaStreamAudioSourceNode; |
| class OscillatorNode; |
| class PannerNode; |
| class PeriodicWave; |
| class ScriptProcessorNode; |
| class SecurityOrigin; |
| class WaveShaperNode; |
| |
| template<typename IDLType> class DOMPromiseDeferred; |
| |
| // AudioContext is the cornerstone of the web audio API and all AudioNodes are created from it. |
| // For thread safety between the audio thread and the main thread, it has a rendering graph locking mechanism. |
| |
| class AudioContext |
| : public ActiveDOMObject |
| , public ThreadSafeRefCounted<AudioContext> |
| , public EventTargetWithInlineData |
| , public MediaCanStartListener |
| , public MediaProducer |
| , private PlatformMediaSessionClient |
| , private VisibilityChangeClient |
| #if !RELEASE_LOG_DISABLED |
| , private LoggerHelper |
| #endif |
| { |
| WTF_MAKE_ISO_ALLOCATED(AudioContext); |
| public: |
| // Create an AudioContext for rendering to the audio hardware. |
| static RefPtr<AudioContext> create(Document&); |
| |
| virtual ~AudioContext(); |
| |
| bool isInitialized() const; |
| |
| bool isOfflineContext() const { return m_isOfflineContext; } |
| |
| Document* document() const; // ASSERTs if document no longer exists. |
| |
| Document* hostingDocument() const final; |
| |
| AudioDestinationNode* destination() { return m_destinationNode.get(); } |
| size_t currentSampleFrame() const { return m_destinationNode ? m_destinationNode->currentSampleFrame() : 0; } |
| double currentTime() const { return m_destinationNode ? m_destinationNode->currentTime() : 0.; } |
| float sampleRate() const { return m_destinationNode ? m_destinationNode->sampleRate() : 0.f; } |
| unsigned long activeSourceCount() const { return static_cast<unsigned long>(m_activeSourceCount); } |
| |
| void incrementActiveSourceCount(); |
| void decrementActiveSourceCount(); |
| |
| ExceptionOr<Ref<AudioBuffer>> createBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate); |
| ExceptionOr<Ref<AudioBuffer>> createBuffer(ArrayBuffer&, bool mixToMono); |
| |
| // Asynchronous audio file data decoding. |
| void decodeAudioData(Ref<ArrayBuffer>&&, RefPtr<AudioBufferCallback>&&, RefPtr<AudioBufferCallback>&&); |
| |
| AudioListener* listener() { return m_listener.get(); } |
| |
| void suspendRendering(DOMPromiseDeferred<void>&&); |
| void resumeRendering(DOMPromiseDeferred<void>&&); |
| void close(DOMPromiseDeferred<void>&&); |
| |
| enum class State { Suspended, Running, Interrupted, Closed }; |
| State state() const; |
| bool isClosed() const { return m_state == State::Closed; } |
| |
| bool wouldTaintOrigin(const URL&) const; |
| |
| // The AudioNode create methods are called on the main thread (from JavaScript). |
| ExceptionOr<Ref<AudioBufferSourceNode>> createBufferSource(); |
| #if ENABLE(VIDEO) |
| ExceptionOr<Ref<MediaElementAudioSourceNode>> createMediaElementSource(HTMLMediaElement&); |
| #endif |
| #if ENABLE(MEDIA_STREAM) |
| ExceptionOr<Ref<MediaStreamAudioSourceNode>> createMediaStreamSource(MediaStream&); |
| ExceptionOr<Ref<MediaStreamAudioDestinationNode>> createMediaStreamDestination(); |
| #endif |
| ExceptionOr<Ref<GainNode>> createGain(); |
| ExceptionOr<Ref<BiquadFilterNode>> createBiquadFilter(); |
| ExceptionOr<Ref<WaveShaperNode>> createWaveShaper(); |
| ExceptionOr<Ref<DelayNode>> createDelay(double maxDelayTime); |
| ExceptionOr<Ref<PannerNode>> createPanner(); |
| ExceptionOr<Ref<ConvolverNode>> createConvolver(); |
| ExceptionOr<Ref<DynamicsCompressorNode>> createDynamicsCompressor(); |
| ExceptionOr<Ref<AnalyserNode>> createAnalyser(); |
| ExceptionOr<Ref<ScriptProcessorNode>> createScriptProcessor(size_t bufferSize, size_t numberOfInputChannels, size_t numberOfOutputChannels); |
| ExceptionOr<Ref<ChannelSplitterNode>> createChannelSplitter(size_t numberOfOutputs); |
| ExceptionOr<Ref<ChannelMergerNode>> createChannelMerger(size_t numberOfInputs); |
| ExceptionOr<Ref<OscillatorNode>> createOscillator(); |
| ExceptionOr<Ref<PeriodicWave>> createPeriodicWave(Float32Array& real, Float32Array& imaginary); |
| |
| // When a source node has no more processing to do (has finished playing), then it tells the context to dereference it. |
| void notifyNodeFinishedProcessing(AudioNode*); |
| |
| // Called at the start of each render quantum. |
| void handlePreRenderTasks(); |
| |
| // Called at the end of each render quantum. |
| void handlePostRenderTasks(); |
| |
| // Called periodically at the end of each render quantum to dereference finished source nodes. |
| void derefFinishedSourceNodes(); |
| |
| // We schedule deletion of all marked nodes at the end of each realtime render quantum. |
| void markForDeletion(AudioNode&); |
| void deleteMarkedNodes(); |
| |
| // AudioContext can pull node(s) at the end of each render quantum even when they are not connected to any downstream nodes. |
| // These two methods are called by the nodes who want to add/remove themselves into/from the automatic pull lists. |
| void addAutomaticPullNode(AudioNode&); |
| void removeAutomaticPullNode(AudioNode&); |
| |
| // Called right before handlePostRenderTasks() to handle nodes which need to be pulled even when they are not connected to anything. |
| void processAutomaticPullNodes(size_t framesToProcess); |
| |
| // Keeps track of the number of connections made. |
| void incrementConnectionCount() |
| { |
| ASSERT(isMainThread()); |
| m_connectionCount++; |
| } |
| |
| unsigned connectionCount() const { return m_connectionCount; } |
| |
| // |
| // Thread Safety and Graph Locking: |
| // |
| |
| void setAudioThread(Thread& thread) { m_audioThread = &thread; } // FIXME: check either not initialized or the same |
| Thread* audioThread() const { return m_audioThread; } |
| bool isAudioThread() const; |
| |
| // Returns true only after the audio thread has been started and then shutdown. |
| bool isAudioThreadFinished() { return m_isAudioThreadFinished; } |
| |
| // mustReleaseLock is set to true if we acquired the lock in this method call and caller must unlock(), false if it was previously acquired. |
| void lock(bool& mustReleaseLock); |
| |
| // Returns true if we own the lock. |
| // mustReleaseLock is set to true if we acquired the lock in this method call and caller must unlock(), false if it was previously acquired. |
| bool tryLock(bool& mustReleaseLock); |
| |
| void unlock(); |
| |
| // Returns true if this thread owns the context's lock. |
| bool isGraphOwner() const; |
| |
| // Returns the maximum number of channels we can support. |
| static unsigned maxNumberOfChannels() { return MaxNumberOfChannels; } |
| |
| class AutoLocker { |
| public: |
| explicit AutoLocker(AudioContext& context) |
| : m_context(context) |
| { |
| m_context.lock(m_mustReleaseLock); |
| } |
| |
| ~AutoLocker() |
| { |
| if (m_mustReleaseLock) |
| m_context.unlock(); |
| } |
| |
| private: |
| AudioContext& m_context; |
| bool m_mustReleaseLock; |
| }; |
| |
| // In AudioNode::deref() a tryLock() is used for calling finishDeref(), but if it fails keep track here. |
| void addDeferredFinishDeref(AudioNode*); |
| |
| // In the audio thread at the start of each render cycle, we'll call handleDeferredFinishDerefs(). |
| void handleDeferredFinishDerefs(); |
| |
| // Only accessed when the graph lock is held. |
| void markSummingJunctionDirty(AudioSummingJunction*); |
| void markAudioNodeOutputDirty(AudioNodeOutput*); |
| |
| // Must be called on main thread. |
| void removeMarkedSummingJunction(AudioSummingJunction*); |
| |
| // EventTarget |
| EventTargetInterface eventTargetInterface() const final { return AudioContextEventTargetInterfaceType; } |
| |
| // Reconcile ref/deref which are defined both in ThreadSafeRefCounted and EventTarget. |
| using ThreadSafeRefCounted::ref; |
| using ThreadSafeRefCounted::deref; |
| |
| void startRendering(); |
| void finishedRendering(bool didRendering); |
| |
| static unsigned s_hardwareContextCount; |
| |
| // Restrictions to change default behaviors. |
| enum BehaviorRestrictionFlags { |
| NoRestrictions = 0, |
| RequireUserGestureForAudioStartRestriction = 1 << 0, |
| RequirePageConsentForAudioStartRestriction = 1 << 1, |
| }; |
| typedef unsigned BehaviorRestrictions; |
| |
| BehaviorRestrictions behaviorRestrictions() const { return m_restrictions; } |
| void addBehaviorRestriction(BehaviorRestrictions restriction) { m_restrictions |= restriction; } |
| void removeBehaviorRestriction(BehaviorRestrictions restriction) { m_restrictions &= ~restriction; } |
| |
| void isPlayingAudioDidChange(); |
| |
| void nodeWillBeginPlayback(); |
| |
| #if !RELEASE_LOG_DISABLED |
| const Logger& logger() const final { return m_logger.get(); } |
| const void* logIdentifier() const final { return m_logIdentifier; } |
| WTFLogChannel& logChannel() const final; |
| const void* nextAudioNodeLogIdentifier() { return childLogIdentifier(++m_nextAudioNodeIdentifier); } |
| const void* nextAudioParameterLogIdentifier() { return childLogIdentifier(++m_nextAudioParameterIdentifier); } |
| #endif |
| |
| void postTask(WTF::Function<void()>&&); |
| bool isStopped() const { return m_isStopScheduled; } |
| const SecurityOrigin* origin() const; |
| void addConsoleMessage(MessageSource, MessageLevel, const String& message); |
| |
| // EventTarget |
| ScriptExecutionContext* scriptExecutionContext() const final; |
| |
| protected: |
| explicit AudioContext(Document&); |
| AudioContext(Document&, AudioBuffer* renderTarget); |
| |
| static bool isSampleRateRangeGood(float sampleRate); |
| void clearPendingActivity(); |
| void makePendingActivity(); |
| |
| private: |
| void constructCommon(); |
| |
| void lazyInitialize(); |
| void uninitialize(); |
| |
| bool willBeginPlayback(); |
| bool willPausePlayback(); |
| |
| bool userGestureRequiredForAudioStart() const { return !isOfflineContext() && m_restrictions & RequireUserGestureForAudioStartRestriction; } |
| bool pageConsentRequiredForAudioStart() const { return !isOfflineContext() && m_restrictions & RequirePageConsentForAudioStartRestriction; } |
| |
| void setState(State); |
| |
| void clear(); |
| |
| void scheduleNodeDeletion(); |
| |
| void mediaCanStart(Document&) override; |
| |
| // EventTarget |
| void dispatchEvent(Event&) final; |
| |
| // MediaProducer |
| MediaProducer::MediaStateFlags mediaState() const override; |
| void pageMutedStateDidChange() override; |
| |
| // The context itself keeps a reference to all source nodes. The source nodes, then reference all nodes they're connected to. |
| // In turn, these nodes reference all nodes they're connected to. All nodes are ultimately connected to the AudioDestinationNode. |
| // When the context dereferences a source node, it will be deactivated from the rendering graph along with all other nodes it is |
| // uniquely connected to. See the AudioNode::ref() and AudioNode::deref() methods for more details. |
| void refNode(AudioNode&); |
| void derefNode(AudioNode&); |
| |
| // ActiveDOMObject API. |
| void suspend(ReasonForSuspension) final; |
| void resume() final; |
| void stop() override; |
| const char* activeDOMObjectName() const override; |
| |
| // When the context goes away, there might still be some sources which haven't finished playing. |
| // Make sure to dereference them here. |
| void derefUnfinishedSourceNodes(); |
| |
| // PlatformMediaSessionClient |
| PlatformMediaSession::MediaType mediaType() const override { return PlatformMediaSession::WebAudio; } |
| PlatformMediaSession::MediaType presentationType() const override { return PlatformMediaSession::WebAudio; } |
| PlatformMediaSession::CharacteristicsFlags characteristics() const override { return m_state == State::Running ? PlatformMediaSession::HasAudio : PlatformMediaSession::HasNothing; } |
| void mayResumePlayback(bool shouldResume) override; |
| void suspendPlayback() override; |
| bool canReceiveRemoteControlCommands() const override { return false; } |
| void didReceiveRemoteControlCommand(PlatformMediaSession::RemoteControlCommandType, const PlatformMediaSession::RemoteCommandArgument*) override { } |
| bool supportsSeeking() const override { return false; } |
| bool shouldOverrideBackgroundPlaybackRestriction(PlatformMediaSession::InterruptionType) const override { return false; } |
| String sourceApplicationIdentifier() const override; |
| bool canProduceAudio() const final { return true; } |
| bool isSuspended() const final; |
| bool processingUserGestureForMedia() const final; |
| |
| void visibilityStateChanged() final; |
| |
| // EventTarget |
| void refEventTarget() override { ref(); } |
| void derefEventTarget() override { deref(); } |
| |
| void handleDirtyAudioSummingJunctions(); |
| void handleDirtyAudioNodeOutputs(); |
| |
| void addReaction(State, DOMPromiseDeferred<void>&&); |
| void updateAutomaticPullNodes(); |
| |
| #if !RELEASE_LOG_DISABLED |
| const char* logClassName() const final { return "AudioContext"; } |
| |
| Ref<Logger> m_logger; |
| const void* m_logIdentifier; |
| uint64_t m_nextAudioNodeIdentifier { 0 }; |
| uint64_t m_nextAudioParameterIdentifier { 0 }; |
| #endif |
| |
| // Only accessed in the audio thread. |
| Vector<AudioNode*> m_finishedNodes; |
| |
| // We don't use RefPtr<AudioNode> here because AudioNode has a more complex ref() / deref() implementation |
| // with an optional argument for refType. We need to use the special refType: RefTypeConnection |
| // Either accessed when the graph lock is held, or on the main thread when the audio thread has finished. |
| Vector<AudioNode*> m_referencedNodes; |
| |
| // Accumulate nodes which need to be deleted here. |
| // This is copied to m_nodesToDelete at the end of a render cycle in handlePostRenderTasks(), where we're assured of a stable graph |
| // state which will have no references to any of the nodes in m_nodesToDelete once the context lock is released |
| // (when handlePostRenderTasks() has completed). |
| Vector<AudioNode*> m_nodesMarkedForDeletion; |
| |
| // They will be scheduled for deletion (on the main thread) at the end of a render cycle (in realtime thread). |
| Vector<AudioNode*> m_nodesToDelete; |
| |
| bool m_isDeletionScheduled { false }; |
| bool m_isStopScheduled { false }; |
| bool m_isInitialized { false }; |
| bool m_isAudioThreadFinished { false }; |
| bool m_automaticPullNodesNeedUpdating { false }; |
| bool m_isOfflineContext { false }; |
| |
| // Only accessed when the graph lock is held. |
| HashSet<AudioSummingJunction*> m_dirtySummingJunctions; |
| HashSet<AudioNodeOutput*> m_dirtyAudioNodeOutputs; |
| |
| // For the sake of thread safety, we maintain a seperate Vector of automatic pull nodes for rendering in m_renderingAutomaticPullNodes. |
| // It will be copied from m_automaticPullNodes by updateAutomaticPullNodes() at the very start or end of the rendering quantum. |
| HashSet<AudioNode*> m_automaticPullNodes; |
| Vector<AudioNode*> m_renderingAutomaticPullNodes; |
| // Only accessed in the audio thread. |
| Vector<AudioNode*> m_deferredFinishDerefList; |
| Vector<Vector<DOMPromiseDeferred<void>>> m_stateReactions; |
| |
| std::unique_ptr<PlatformMediaSession> m_mediaSession; |
| UniqueRef<MainThreadGenericEventQueue> m_eventQueue; |
| |
| RefPtr<AudioBuffer> m_renderTarget; |
| RefPtr<AudioDestinationNode> m_destinationNode; |
| RefPtr<AudioListener> m_listener; |
| |
| unsigned m_connectionCount { 0 }; |
| |
| // Graph locking. |
| Lock m_contextGraphMutex; |
| // FIXME: Using volatile seems incorrect. |
| // https://bugs.webkit.org/show_bug.cgi?id=180332 |
| Thread* volatile m_audioThread { nullptr }; |
| Thread* volatile m_graphOwnerThread { nullptr }; // if the lock is held then this is the thread which owns it, otherwise == nullptr. |
| |
| std::unique_ptr<AsyncAudioDecoder> m_audioDecoder; |
| |
| // This is considering 32 is large enough for multiple channels audio. |
| // It is somewhat arbitrary and could be increased if necessary. |
| enum { MaxNumberOfChannels = 32 }; |
| |
| // Number of AudioBufferSourceNodes that are active (playing). |
| std::atomic<int> m_activeSourceCount { 0 }; |
| |
| BehaviorRestrictions m_restrictions { NoRestrictions }; |
| |
| State m_state { State::Suspended }; |
| RefPtr<PendingActivity<AudioContext>> m_pendingActivity; |
| }; |
| |
| // FIXME: Find out why these ==/!= functions are needed and remove them if possible. |
| |
| inline bool operator==(const AudioContext& lhs, const AudioContext& rhs) |
| { |
| return &lhs == &rhs; |
| } |
| |
| inline bool operator!=(const AudioContext& lhs, const AudioContext& rhs) |
| { |
| return &lhs != &rhs; |
| } |
| |
| inline AudioContext::State AudioContext::state() const |
| { |
| return m_state; |
| } |
| |
| } // WebCore |