| /* |
| * Copyright (C) 2010, Google 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. |
| */ |
| |
| #ifndef AudioContext_h |
| #define AudioContext_h |
| |
| #include "ActiveDOMObject.h" |
| #include "AudioBus.h" |
| #include "AudioDestinationNode.h" |
| #include "EventListener.h" |
| #include "EventTarget.h" |
| #include "HRTFDatabaseLoader.h" |
| #include <wtf/HashSet.h> |
| #include <wtf/OwnPtr.h> |
| #include <wtf/PassRefPtr.h> |
| #include <wtf/RefCounted.h> |
| #include <wtf/RefPtr.h> |
| #include <wtf/Threading.h> |
| #include <wtf/Vector.h> |
| #include <wtf/text/AtomicStringHash.h> |
| |
| namespace WebCore { |
| |
| class ArrayBuffer; |
| class AudioBuffer; |
| class AudioBufferSourceNode; |
| class AudioChannelMerger; |
| class AudioChannelSplitter; |
| class AudioGainNode; |
| class AudioPannerNode; |
| class AudioListener; |
| class DelayNode; |
| class Document; |
| class LowPass2FilterNode; |
| class HighPass2FilterNode; |
| class ConvolverNode; |
| class RealtimeAnalyserNode; |
| class JavaScriptAudioNode; |
| |
| // 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 RefCounted<AudioContext>, public EventTarget { |
| public: |
| // Create an AudioContext for rendering to the audio hardware. |
| static PassRefPtr<AudioContext> create(Document*); |
| |
| // Create an AudioContext for offline (non-realtime) rendering. |
| static PassRefPtr<AudioContext> createOfflineContext(Document*, unsigned numberOfChannels, size_t numberOfFrames, double sampleRate); |
| |
| virtual ~AudioContext(); |
| |
| bool isInitialized() const; |
| |
| bool isOfflineContext() { return m_isOfflineContext; } |
| |
| // Returns true when initialize() was called AND all asynchronous initialization has completed. |
| bool isRunnable() const; |
| |
| // Document notification |
| virtual void stop(); |
| |
| Document* document() const; // ASSERTs if document no longer exists. |
| bool hasDocument(); |
| |
| AudioDestinationNode* destination() { return m_destinationNode.get(); } |
| double currentTime() { return m_destinationNode->currentTime(); } |
| double sampleRate() { return m_destinationNode->sampleRate(); } |
| |
| PassRefPtr<AudioBuffer> createBuffer(unsigned numberOfChannels, size_t numberOfFrames, double sampleRate); |
| PassRefPtr<AudioBuffer> createBuffer(ArrayBuffer* arrayBuffer, bool mixToMono); |
| |
| // Keep track of this buffer so we can release memory after the context is shut down... |
| void refBuffer(PassRefPtr<AudioBuffer> buffer); |
| |
| AudioListener* listener() { return m_listener.get(); } |
| |
| // The AudioNode create methods are called on the main thread (from JavaScript). |
| PassRefPtr<AudioBufferSourceNode> createBufferSource(); |
| PassRefPtr<AudioGainNode> createGainNode(); |
| PassRefPtr<DelayNode> createDelayNode(); |
| PassRefPtr<LowPass2FilterNode> createLowPass2Filter(); |
| PassRefPtr<HighPass2FilterNode> createHighPass2Filter(); |
| PassRefPtr<AudioPannerNode> createPanner(); |
| PassRefPtr<ConvolverNode> createConvolver(); |
| PassRefPtr<RealtimeAnalyserNode> createAnalyser(); |
| PassRefPtr<JavaScriptAudioNode> createJavaScriptNode(size_t bufferSize); |
| PassRefPtr<AudioChannelSplitter> createChannelSplitter(); |
| PassRefPtr<AudioChannelMerger> createChannelMerger(); |
| |
| AudioBus* temporaryMonoBus() { return m_temporaryMonoBus.get(); } |
| AudioBus* temporaryStereoBus() { return m_temporaryStereoBus.get(); } |
| |
| // 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 reap all marked nodes at the end of each realtime render quantum in deleteMarkedNodes(). |
| void markForDeletion(AudioNode*); |
| void deleteMarkedNodes(); |
| |
| // 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(ThreadIdentifier thread) { m_audioThread = thread; } // FIXME: check either not initialized or the same |
| ThreadIdentifier 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; |
| |
| class AutoLocker { |
| public: |
| AutoLocker(AudioContext* context) |
| : m_context(context) |
| { |
| ASSERT(context); |
| 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*, AudioNode::RefType); |
| |
| // 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 markAudioNodeInputDirty(AudioNodeInput*); |
| void markAudioNodeOutputDirty(AudioNodeOutput*); |
| |
| // EventTarget |
| virtual ScriptExecutionContext* scriptExecutionContext() const; |
| virtual AudioContext* toAudioContext(); |
| virtual EventTargetData* eventTargetData() { return &m_eventTargetData; } |
| virtual EventTargetData* ensureEventTargetData() { return &m_eventTargetData; } |
| |
| DEFINE_ATTRIBUTE_EVENT_LISTENER(complete); |
| |
| // Reconcile ref/deref which are defined both in AudioNode and EventTarget. |
| using RefCounted<AudioContext>::ref; |
| using RefCounted<AudioContext>::deref; |
| |
| void startRendering(); |
| void fireCompletionEvent(); |
| |
| private: |
| AudioContext(Document*); |
| AudioContext(Document*, unsigned numberOfChannels, size_t numberOfFrames, double sampleRate); |
| void constructCommon(); |
| |
| void lazyInitialize(); |
| void uninitialize(); |
| |
| bool m_isInitialized; |
| bool m_isAudioThreadFinished; |
| bool m_isAudioThreadShutdown; |
| |
| Document* m_document; |
| |
| // 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*); |
| |
| // When the context goes away, there might still be some sources which haven't finished playing. |
| // Make sure to dereference them here. |
| void derefUnfinishedSourceNodes(); |
| |
| RefPtr<AudioDestinationNode> m_destinationNode; |
| RefPtr<AudioListener> m_listener; |
| |
| // Only accessed in the main thread. |
| Vector<RefPtr<AudioBuffer> > m_allocatedBuffers; |
| |
| // 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 at the end of a render cycle (in realtime thread) here. |
| Vector<AudioNode*> m_nodesToDelete; |
| |
| // Only accessed when the graph lock is held. |
| HashSet<AudioNodeInput*> m_dirtyAudioNodeInputs; |
| HashSet<AudioNodeOutput*> m_dirtyAudioNodeOutputs; |
| void handleDirtyAudioNodeInputs(); |
| void handleDirtyAudioNodeOutputs(); |
| |
| OwnPtr<AudioBus> m_temporaryMonoBus; |
| OwnPtr<AudioBus> m_temporaryStereoBus; |
| |
| unsigned m_connectionCount; |
| |
| // Graph locking. |
| Mutex m_contextGraphMutex; |
| volatile ThreadIdentifier m_audioThread; |
| volatile ThreadIdentifier m_graphOwnerThread; // if the lock is held then this is the thread which owns it, otherwise == UndefinedThreadIdentifier |
| |
| // Deferred de-referencing. |
| struct RefInfo { |
| RefInfo(AudioNode* node, AudioNode::RefType refType) |
| : m_node(node) |
| , m_refType(refType) |
| { |
| } |
| AudioNode* m_node; |
| AudioNode::RefType m_refType; |
| }; |
| |
| // Only accessed in the audio thread. |
| Vector<RefInfo> m_deferredFinishDerefList; |
| |
| // HRTF Database loader |
| RefPtr<HRTFDatabaseLoader> m_hrtfDatabaseLoader; |
| |
| // EventTarget |
| virtual void refEventTarget() { ref(); } |
| virtual void derefEventTarget() { deref(); } |
| EventTargetData m_eventTargetData; |
| |
| RefPtr<AudioBuffer> m_renderTarget; |
| |
| bool m_isOfflineContext; |
| }; |
| |
| } // WebCore |
| |
| #endif // AudioContext_h |