Move webaudio to oilpan transitional types

We don't allow on-heap allocations in the audio thread. We change AsyncAudioDecoder::decode so we won't allocate an AudioBuffer in the audio thread.

BUG=340522

Review URL: https://codereview.chromium.org/205173002

git-svn-id: svn://svn.chromium.org/blink/trunk@173725 bbb929c8-8fbe-4397-9dbb-9b2b20218538

third_party/WebKit/Source/modules/webaudio/AnalyserNode.h

@@ -35,9 +35,9 @@ class ExceptionState;
 
 class AnalyserNode FINAL : public AudioBasicInspectorNode {
 public:
-    static PassRefPtr<AnalyserNode> create(AudioContext* context, float sampleRate)
+    static PassRefPtrWillBeRawPtr<AnalyserNode> create(AudioContext* context, float sampleRate)
     {
-        return adoptRef(new AnalyserNode(context, sampleRate));
+        return adoptRefWillBeNoop(new AnalyserNode(context, sampleRate));
     }
 
     virtual ~AnalyserNode();

third_party/WebKit/Source/modules/webaudio/AsyncAudioDecoder.cpp

@@ -31,6 +31,8 @@
 #include "modules/webaudio/AudioBuffer.h"
 #include "modules/webaudio/AudioBufferCallback.h"
 #include "platform/Task.h"
+#include "platform/audio/AudioBus.h"
+#include "platform/audio/AudioFileReader.h"
 #include "public/platform/Platform.h"
 #include "wtf/ArrayBuffer.h"
 #include "wtf/MainThread.h"
@@ -63,26 +65,26 @@ void AsyncAudioDecoder::decodeAsync(ArrayBuffer* audioData, float sampleRate, Pa
 
 void AsyncAudioDecoder::decode(ArrayBuffer* audioData, float sampleRate, AudioBufferCallback* successCallback, AudioBufferCallback* errorCallback)
 {
-    // Do the actual decoding and invoke the callback.
-    RefPtr<AudioBuffer> audioBuffer = AudioBuffer::createFromAudioFileData(audioData->data(), audioData->byteLength(), false, sampleRate);
+    RefPtr<AudioBus> bus = createBusFromInMemoryAudioFile(audioData->data(), audioData->byteLength(), false, sampleRate);
 
     // Decoding is finished, but we need to do the callbacks on the main thread.
     // The leaked reference to audioBuffer is picked up in notifyComplete.
-    callOnMainThread(WTF::bind(&AsyncAudioDecoder::notifyComplete, audioData, successCallback, errorCallback, audioBuffer.release().leakRef()));
+    callOnMainThread(WTF::bind(&AsyncAudioDecoder::notifyComplete, audioData, successCallback, errorCallback, bus.release().leakRef()));
 }
 
-void AsyncAudioDecoder::notifyComplete(ArrayBuffer* audioData, AudioBufferCallback* successCallback, AudioBufferCallback* errorCallback, AudioBuffer* audioBuffer)
+void AsyncAudioDecoder::notifyComplete(ArrayBuffer* audioData, AudioBufferCallback* successCallback, AudioBufferCallback* errorCallback, AudioBus* audioBus)
 {
     // Adopt references, so everything gets correctly dereffed.
     RefPtr<ArrayBuffer> audioDataRef = adoptRef(audioData);
     OwnPtr<AudioBufferCallback> successCallbackPtr = adoptPtr(successCallback);
     OwnPtr<AudioBufferCallback> errorCallbackPtr = adoptPtr(errorCallback);
-    RefPtr<AudioBuffer> audioBufferRef = adoptRef(audioBuffer);
+    RefPtr<AudioBus> audioBusRef = adoptRef(audioBus);
 
-    if (audioBuffer && successCallback)
-        successCallback->handleEvent(audioBuffer);
+    RefPtrWillBeRawPtr<AudioBuffer> audioBuffer = AudioBuffer::createFromAudioBus(audioBus);
+    if (audioBuffer.get() && successCallback)
+        successCallback->handleEvent(audioBuffer.get());
     else if (errorCallback)
-        errorCallback->handleEvent(audioBuffer);
+        errorCallback->handleEvent(audioBuffer.get());
 }
 
 } // namespace WebCore

third_party/WebKit/Source/modules/webaudio/AsyncAudioDecoder.h

@@ -25,6 +25,7 @@
 #ifndef AsyncAudioDecoder_h
 #define AsyncAudioDecoder_h
 
+#include "platform/heap/Handle.h"
 #include "public/platform/WebThread.h"
 #include "wtf/Forward.h"
 #include "wtf/OwnPtr.h"
@@ -33,6 +34,7 @@ namespace WebCore {
 
 class AudioBuffer;
 class AudioBufferCallback;
+class AudioBus;
 
 // AsyncAudioDecoder asynchronously decodes audio file data from an ArrayBuffer in a worker thread.
 // Upon successful decoding, a completion callback will be invoked with the decoded PCM data in an AudioBuffer.
@@ -47,8 +49,9 @@ public:
     void decodeAsync(ArrayBuffer* audioData, float sampleRate, PassOwnPtr<AudioBufferCallback> successCallback, PassOwnPtr<AudioBufferCallback> errorCallback);
 
 private:
+    PassRefPtrWillBeRawPtr<AudioBuffer> createAudioBufferFromAudioBus(AudioBus*);
     static void decode(ArrayBuffer* audioData, float sampleRate, AudioBufferCallback* successCallback, AudioBufferCallback* errorCallback);
-    static void notifyComplete(ArrayBuffer* audioData, AudioBufferCallback* successCallback, AudioBufferCallback* errorCallback, AudioBuffer*);
+    static void notifyComplete(ArrayBuffer* audioData, AudioBufferCallback* successCallback, AudioBufferCallback* errorCallback, AudioBus*);
 
     OwnPtr<blink::WebThread> m_thread;
 };

third_party/WebKit/Source/modules/webaudio/AudioBuffer.cpp

@@ -53,19 +53,19 @@ float AudioBuffer::maxAllowedSampleRate()
     return 192000;
 }
 
-PassRefPtr<AudioBuffer> AudioBuffer::create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate)
+PassRefPtrWillBeRawPtr<AudioBuffer> AudioBuffer::create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate)
 {
     if (sampleRate < minAllowedSampleRate() || sampleRate > maxAllowedSampleRate() || numberOfChannels > AudioContext::maxNumberOfChannels() || !numberOfChannels || !numberOfFrames)
         return nullptr;
 
-    RefPtr<AudioBuffer> buffer = adoptRef(new AudioBuffer(numberOfChannels, numberOfFrames, sampleRate));
+    RefPtrWillBeRawPtr<AudioBuffer> buffer = adoptRefWillBeNoop(new AudioBuffer(numberOfChannels, numberOfFrames, sampleRate));
 
     if (!buffer->createdSuccessfully(numberOfChannels))
         return nullptr;
     return buffer;
 }
 
-PassRefPtr<AudioBuffer> AudioBuffer::create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<AudioBuffer> AudioBuffer::create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState)
 {
     if (!numberOfChannels || numberOfChannels > AudioContext::maxNumberOfChannels()) {
         exceptionState.throwDOMException(
@@ -103,7 +103,7 @@ PassRefPtr<AudioBuffer> AudioBuffer::create(unsigned numberOfChannels, size_t nu
         return nullptr;
     }
 
-    RefPtr<AudioBuffer> audioBuffer = create(numberOfChannels, numberOfFrames, sampleRate);
+    RefPtrWillBeRawPtr<AudioBuffer> audioBuffer = create(numberOfChannels, numberOfFrames, sampleRate);
 
     if (!audioBuffer.get()) {
         exceptionState.throwDOMException(
@@ -118,11 +118,11 @@ PassRefPtr<AudioBuffer> AudioBuffer::create(unsigned numberOfChannels, size_t nu
     return audioBuffer;
 }
 
-PassRefPtr<AudioBuffer> AudioBuffer::createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, float sampleRate)
+PassRefPtrWillBeRawPtr<AudioBuffer> AudioBuffer::createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, float sampleRate)
 {
     RefPtr<AudioBus> bus = createBusFromInMemoryAudioFile(data, dataSize, mixToMono, sampleRate);
     if (bus.get()) {
-        RefPtr<AudioBuffer> buffer = adoptRef(new AudioBuffer(bus.get()));
+        RefPtrWillBeRawPtr<AudioBuffer> buffer = adoptRefWillBeNoop(new AudioBuffer(bus.get()));
         if (buffer->createdSuccessfully(bus->numberOfChannels()))
             return buffer;
     }
@@ -130,6 +130,16 @@ PassRefPtr<AudioBuffer> AudioBuffer::createFromAudioFileData(const void* data, s
     return nullptr;
 }
 
+PassRefPtrWillBeRawPtr<AudioBuffer> AudioBuffer::createFromAudioBus(AudioBus* bus)
+{
+    if (!bus)
+        return nullptr;
+    RefPtrWillBeRawPtr<AudioBuffer> buffer = adoptRefWillBeNoop(new AudioBuffer(bus));
+    if (buffer->createdSuccessfully(bus->numberOfChannels()))
+        return buffer;
+    return nullptr;
+}
+
 bool AudioBuffer::createdSuccessfully(unsigned desiredNumberOfChannels) const
 {
     return numberOfChannels() == desiredNumberOfChannels;

third_party/WebKit/Source/modules/webaudio/AudioBuffer.h

@@ -41,13 +41,15 @@ namespace WebCore {
 class AudioBus;
 class ExceptionState;
 
-class AudioBuffer : public ScriptWrappable, public RefCounted<AudioBuffer> {
+class AudioBuffer : public RefCountedWillBeGarbageCollectedFinalized<AudioBuffer>, public ScriptWrappable {
 public:
-    static PassRefPtr<AudioBuffer> create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
-    static PassRefPtr<AudioBuffer> create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState&);
+    static PassRefPtrWillBeRawPtr<AudioBuffer> create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
+    static PassRefPtrWillBeRawPtr<AudioBuffer> create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState&);
 
     // Returns 0 if data is not a valid audio file.
-    static PassRefPtr<AudioBuffer> createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, float sampleRate);
+    static PassRefPtrWillBeRawPtr<AudioBuffer> createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, float sampleRate);
+
+    static PassRefPtrWillBeRawPtr<AudioBuffer> createFromAudioBus(AudioBus*);
 
     // Format
     size_t length() const { return m_length; }
@@ -62,6 +64,9 @@ public:
 
     static float minAllowedSampleRate();
     static float maxAllowedSampleRate();
+
+    void trace(Visitor*) { }
+
 protected:
     AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
     explicit AudioBuffer(AudioBus*);

third_party/WebKit/Source/modules/webaudio/AudioBuffer.idl

@@ -27,6 +27,7 @@
  */
 
 [
+    WillBeGarbageCollected,
     Conditional=WEB_AUDIO
 ] interface AudioBuffer {
     readonly attribute long length; // in sample-frames

third_party/WebKit/Source/modules/webaudio/AudioBufferSourceNode.cpp

@@ -49,9 +49,9 @@ const double DefaultGrainDuration = 0.020; // 20ms
 // to minimize linear interpolation aliasing.
 const double MaxRate = 1024;
 
-PassRefPtr<AudioBufferSourceNode> AudioBufferSourceNode::create(AudioContext* context, float sampleRate)
+PassRefPtrWillBeRawPtr<AudioBufferSourceNode> AudioBufferSourceNode::create(AudioContext* context, float sampleRate)
 {
-    return adoptRef(new AudioBufferSourceNode(context, sampleRate));
+    return adoptRefWillBeNoop(new AudioBufferSourceNode(context, sampleRate));
 }
 
 AudioBufferSourceNode::AudioBufferSourceNode(AudioContext* context, float sampleRate)
@@ -475,6 +475,13 @@ void AudioBufferSourceNode::finish()
     AudioScheduledSourceNode::finish();
 }
 
+void AudioBufferSourceNode::trace(Visitor* visitor)
+{
+    visitor->trace(m_buffer);
+    visitor->trace(m_playbackRate);
+    AudioScheduledSourceNode::trace(visitor);
+}
+
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)

third_party/WebKit/Source/modules/webaudio/AudioBufferSourceNode.h

@@ -44,7 +44,7 @@ class AudioContext;
 
 class AudioBufferSourceNode FINAL : public AudioScheduledSourceNode {
 public:
-    static PassRefPtr<AudioBufferSourceNode> create(AudioContext*, float sampleRate);
+    static PassRefPtrWillBeRawPtr<AudioBufferSourceNode> create(AudioContext*, float sampleRate);
 
     virtual ~AudioBufferSourceNode();
 
@@ -89,6 +89,8 @@ public:
     // AudioScheduledSourceNode
     virtual void finish() OVERRIDE;
 
+    virtual void trace(Visitor*) OVERRIDE;
+
 private:
     AudioBufferSourceNode(AudioContext*, float sampleRate);
 
@@ -99,14 +101,14 @@ private:
     inline bool renderSilenceAndFinishIfNotLooping(AudioBus*, unsigned index, size_t framesToProcess);
 
     // m_buffer holds the sample data which this node outputs.
-    RefPtr<AudioBuffer> m_buffer;
+    RefPtrWillBeMember<AudioBuffer> m_buffer;
 
     // Pointers for the buffer and destination.
     OwnPtr<const float*[]> m_sourceChannels;
     OwnPtr<float*[]> m_destinationChannels;
 
     // Used for the "playbackRate" attributes.
-    RefPtr<AudioParam> m_playbackRate;
+    RefPtrWillBeMember<AudioParam> m_playbackRate;
 
     // If m_isLooping is false, then this node will be done playing and become inactive after it reaches the end of the sample data in the buffer.
     // If true, it will wrap around to the start of the buffer each time it reaches the end.

third_party/WebKit/Source/modules/webaudio/AudioContext.h

@@ -29,9 +29,10 @@
 #include "core/dom/ActiveDOMObject.h"
 #include "core/events/EventListener.h"
 #include "core/events/EventTarget.h"
-#include "platform/audio/AudioBus.h"
 #include "modules/webaudio/AsyncAudioDecoder.h"
 #include "modules/webaudio/AudioDestinationNode.h"
+#include "platform/audio/AudioBus.h"
+#include "platform/heap/Handle.h"
 #include "wtf/HashSet.h"
 #include "wtf/MainThread.h"
 #include "wtf/OwnPtr.h"
@@ -73,14 +74,16 @@ class WaveShaperNode;
 // 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 ScriptWrappable, public ThreadSafeRefCounted<AudioContext>, public EventTargetWithInlineData {
-    DEFINE_EVENT_TARGET_REFCOUNTING(ThreadSafeRefCounted<AudioContext>);
+class AudioContext : public ThreadSafeRefCountedWillBeThreadSafeRefCountedGarbageCollected<AudioContext>, public ActiveDOMObject, public ScriptWrappable, public EventTargetWithInlineData {
+    DEFINE_EVENT_TARGET_REFCOUNTING(ThreadSafeRefCountedWillBeThreadSafeRefCountedGarbageCollected<AudioContext>);
 public:
     // Create an AudioContext for rendering to the audio hardware.
-    static PassRefPtr<AudioContext> create(Document&, ExceptionState&);
+    static PassRefPtrWillBeRawPtr<AudioContext> create(Document&, ExceptionState&);
 
     virtual ~AudioContext();
 
+    virtual void trace(Visitor*);
+
     bool isInitialized() const;
     // The constructor of an AudioNode must call this to initialize the context.
     void lazyInitialize();
@@ -96,7 +99,7 @@ public:
     double currentTime() const { return m_destinationNode->currentTime(); }
     float sampleRate() const { return m_destinationNode->sampleRate(); }
 
-    PassRefPtr<AudioBuffer> createBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState&);
+    PassRefPtrWillBeRawPtr<AudioBuffer> createBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState&);
 
     // Asynchronous audio file data decoding.
     void decodeAudioData(ArrayBuffer*, PassOwnPtr<AudioBufferCallback>, PassOwnPtr<AudioBufferCallback>, ExceptionState&);
@@ -104,29 +107,29 @@ public:
     AudioListener* listener() { return m_listener.get(); }
 
     // The AudioNode create methods are called on the main thread (from JavaScript).
-    PassRefPtr<AudioBufferSourceNode> createBufferSource();
-    PassRefPtr<MediaElementAudioSourceNode> createMediaElementSource(HTMLMediaElement*, ExceptionState&);
-    PassRefPtr<MediaStreamAudioSourceNode> createMediaStreamSource(MediaStream*, ExceptionState&);
-    PassRefPtr<MediaStreamAudioDestinationNode> createMediaStreamDestination();
-    PassRefPtr<GainNode> createGain();
-    PassRefPtr<BiquadFilterNode> createBiquadFilter();
-    PassRefPtr<WaveShaperNode> createWaveShaper();
-    PassRefPtr<DelayNode> createDelay(ExceptionState&);
-    PassRefPtr<DelayNode> createDelay(double maxDelayTime, ExceptionState&);
-    PassRefPtr<PannerNode> createPanner();
-    PassRefPtr<ConvolverNode> createConvolver();
-    PassRefPtr<DynamicsCompressorNode> createDynamicsCompressor();
-    PassRefPtr<AnalyserNode> createAnalyser();
-    PassRefPtr<ScriptProcessorNode> createScriptProcessor(ExceptionState&);
-    PassRefPtr<ScriptProcessorNode> createScriptProcessor(size_t bufferSize, ExceptionState&);
-    PassRefPtr<ScriptProcessorNode> createScriptProcessor(size_t bufferSize, size_t numberOfInputChannels, ExceptionState&);
-    PassRefPtr<ScriptProcessorNode> createScriptProcessor(size_t bufferSize, size_t numberOfInputChannels, size_t numberOfOutputChannels, ExceptionState&);
-    PassRefPtr<ChannelSplitterNode> createChannelSplitter(ExceptionState&);
-    PassRefPtr<ChannelSplitterNode> createChannelSplitter(size_t numberOfOutputs, ExceptionState&);
-    PassRefPtr<ChannelMergerNode> createChannelMerger(ExceptionState&);
-    PassRefPtr<ChannelMergerNode> createChannelMerger(size_t numberOfInputs, ExceptionState&);
-    PassRefPtr<OscillatorNode> createOscillator();
-    PassRefPtr<PeriodicWave> createPeriodicWave(Float32Array* real, Float32Array* imag, ExceptionState&);
+    PassRefPtrWillBeRawPtr<AudioBufferSourceNode> createBufferSource();
+    PassRefPtrWillBeRawPtr<MediaElementAudioSourceNode> createMediaElementSource(HTMLMediaElement*, ExceptionState&);
+    PassRefPtrWillBeRawPtr<MediaStreamAudioSourceNode> createMediaStreamSource(MediaStream*, ExceptionState&);
+    PassRefPtrWillBeRawPtr<MediaStreamAudioDestinationNode> createMediaStreamDestination();
+    PassRefPtrWillBeRawPtr<GainNode> createGain();
+    PassRefPtrWillBeRawPtr<BiquadFilterNode> createBiquadFilter();
+    PassRefPtrWillBeRawPtr<WaveShaperNode> createWaveShaper();
+    PassRefPtrWillBeRawPtr<DelayNode> createDelay(ExceptionState&);
+    PassRefPtrWillBeRawPtr<DelayNode> createDelay(double maxDelayTime, ExceptionState&);
+    PassRefPtrWillBeRawPtr<PannerNode> createPanner();
+    PassRefPtrWillBeRawPtr<ConvolverNode> createConvolver();
+    PassRefPtrWillBeRawPtr<DynamicsCompressorNode> createDynamicsCompressor();
+    PassRefPtrWillBeRawPtr<AnalyserNode> createAnalyser();
+    PassRefPtrWillBeRawPtr<ScriptProcessorNode> createScriptProcessor(ExceptionState&);
+    PassRefPtrWillBeRawPtr<ScriptProcessorNode> createScriptProcessor(size_t bufferSize, ExceptionState&);
+    PassRefPtrWillBeRawPtr<ScriptProcessorNode> createScriptProcessor(size_t bufferSize, size_t numberOfInputChannels, ExceptionState&);
+    PassRefPtrWillBeRawPtr<ScriptProcessorNode> createScriptProcessor(size_t bufferSize, size_t numberOfInputChannels, size_t numberOfOutputChannels, ExceptionState&);
+    PassRefPtrWillBeRawPtr<ChannelSplitterNode> createChannelSplitter(ExceptionState&);
+    PassRefPtrWillBeRawPtr<ChannelSplitterNode> createChannelSplitter(size_t numberOfOutputs, ExceptionState&);
+    PassRefPtrWillBeRawPtr<ChannelMergerNode> createChannelMerger(ExceptionState&);
+    PassRefPtrWillBeRawPtr<ChannelMergerNode> createChannelMerger(size_t numberOfInputs, ExceptionState&);
+    PassRefPtrWillBeRawPtr<OscillatorNode> createOscillator();
+    PassRefPtrWillBeRawPtr<PeriodicWave> createPeriodicWave(Float32Array* real, Float32Array* imag, ExceptionState&);
 
     // When a source node has no more processing to do (has finished playing), then it tells the context to dereference it.
     void notifyNodeFinishedProcessing(AudioNode*);
@@ -266,8 +269,8 @@ private:
     // Make sure to dereference them here.
     void derefUnfinishedSourceNodes();
 
-    RefPtr<AudioDestinationNode> m_destinationNode;
-    RefPtr<AudioListener> m_listener;
+    RefPtrWillBeMember<AudioDestinationNode> m_destinationNode;
+    RefPtrWillBeMember<AudioListener> m_listener;
 
     // Only accessed in the audio thread.
     Vector<AudioNode*> m_finishedNodes;
@@ -288,7 +291,7 @@ private:
     bool m_isDeletionScheduled;
 
     // Only accessed when the graph lock is held.
-    HashSet<AudioSummingJunction*> m_dirtySummingJunctions;
+    HashSet<AudioSummingJunction* > m_dirtySummingJunctions;
     HashSet<AudioNodeOutput*> m_dirtyAudioNodeOutputs;
     void handleDirtyAudioSummingJunctions();
     void handleDirtyAudioNodeOutputs();
@@ -311,7 +314,7 @@ private:
     // Only accessed in the audio thread.
     Vector<AudioNode*> m_deferredFinishDerefList;
 
-    RefPtr<AudioBuffer> m_renderTarget;
+    RefPtrWillBeMember<AudioBuffer> m_renderTarget;
 
     bool m_isOfflineContext;
 

third_party/WebKit/Source/modules/webaudio/AudioContext.idl

@@ -24,6 +24,7 @@
  */
 
 [
+    WillBeGarbageCollected,
     ActiveDOMObject,
     Conditional=WEB_AUDIO,
     Constructor,

third_party/WebKit/Source/modules/webaudio/AudioListener.h

@@ -31,6 +31,7 @@
 
 #include "bindings/v8/ScriptWrappable.h"
 #include "platform/geometry/FloatPoint3D.h"
+#include "platform/heap/Handle.h"
 #include "wtf/PassRefPtr.h"
 #include "wtf/RefCounted.h"
 #include "wtf/Vector.h"
@@ -41,11 +42,11 @@ class PannerNode;
 
 // AudioListener maintains the state of the listener in the audio scene as defined in the OpenAL specification.
 
-class AudioListener : public ScriptWrappable, public RefCounted<AudioListener> {
+class AudioListener : public RefCountedWillBeGarbageCollectedFinalized<AudioListener>, public ScriptWrappable {
 public:
-    static PassRefPtr<AudioListener> create()
+    static PassRefPtrWillBeRawPtr<AudioListener> create()
     {
-        return adoptRef(new AudioListener());
+        return adoptRefWillBeNoop(new AudioListener());
     }
     virtual ~AudioListener();
 
@@ -78,6 +79,8 @@ public:
     void addPanner(PannerNode*);
     void removePanner(PannerNode*);
 
+    void trace(Visitor*) { }
+
 private:
     AudioListener();
 

third_party/WebKit/Source/modules/webaudio/AudioListener.idl

@@ -27,6 +27,7 @@
  */
 
 [
+    WillBeGarbageCollected,
     Conditional=WEB_AUDIO
 ] interface AudioListener {
     attribute float dopplerFactor;  // same as OpenAL (default 1.0)

third_party/WebKit/Source/modules/webaudio/AudioNode.cpp

@@ -48,6 +48,9 @@ AudioNode::AudioNode(AudioContext* context, float sampleRate)
     , m_nodeType(NodeTypeUnknown)
     , m_context(context)
     , m_sampleRate(sampleRate)
+#if ENABLE(OILPAN)
+    , m_keepAlive(adoptPtr(new Persistent<AudioNode>(this)))
+#endif
     , m_lastProcessingTime(-1)
     , m_lastNonSilentTime(-1)
     , m_normalRefCount(1) // start out with normal refCount == 1 (like WTF::RefCounted class)
@@ -473,6 +476,9 @@ void AudioNode::disableOutputsIfNecessary()
 
 void AudioNode::ref(RefType refType)
 {
+#if ENABLE(OILPAN)
+    ASSERT(m_keepAlive);
+#endif
     switch (refType) {
     case RefTypeNormal:
         atomicIncrement(&m_normalRefCount);
@@ -587,6 +593,24 @@ void AudioNode::printNodeCounts()
 
 #endif // DEBUG_AUDIONODE_REFERENCES
 
+void AudioNode::trace(Visitor* visitor)
+{
+    visitor->trace(m_context);
+}
+
+#if ENABLE(OILPAN)
+void AudioNode::clearKeepAlive()
+{
+    // It is safe to drop the self-persistent when the ref count
+    // of a AudioNode reaches zero. At that point, the
+    // AudioNode node is removed from the AudioContext and
+    // it cannot be reattached. Therefore, the reference count
+    // will not go above zero again.
+    ASSERT(m_keepAlive);
+    m_keepAlive = nullptr;
+}
+#endif
+
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)

third_party/WebKit/Source/modules/webaudio/AudioNode.h

@@ -50,7 +50,8 @@ class ExceptionState;
 // An AudioDestinationNode has one input and no outputs and represents the final destination to the audio hardware.
 // Most processing nodes such as filters will have one input and one output, although multiple inputs and outputs are possible.
 
-class AudioNode : public ScriptWrappable, public EventTargetWithInlineData {
+// AudioNode has its own ref-counting mechanism that use RefTypes so we cannot use RefCountedGarbageCollected.
+class AudioNode : public NoBaseWillBeGarbageCollectedFinalized<AudioNode>, public ScriptWrappable, public EventTargetWithInlineData {
 public:
     enum { ProcessingSizeInFrames = 128 };
 
@@ -179,6 +180,12 @@ public:
     virtual const AtomicString& interfaceName() const OVERRIDE FINAL;
     virtual ExecutionContext* executionContext() const OVERRIDE FINAL;
 
+    virtual void trace(Visitor*);
+
+#if ENABLE(OILPAN)
+    void clearKeepAlive();
+#endif
+
 protected:
     // Inputs and outputs must be created before the AudioNode is initialized.
     void addInput(PassOwnPtr<AudioNodeInput>);
@@ -195,11 +202,23 @@ protected:
 private:
     volatile bool m_isInitialized;
     NodeType m_nodeType;
-    RefPtr<AudioContext> m_context;
+    RefPtrWillBeMember<AudioContext> m_context;
     float m_sampleRate;
     Vector<OwnPtr<AudioNodeInput> > m_inputs;
     Vector<OwnPtr<AudioNodeOutput> > m_outputs;
 
+#if ENABLE(OILPAN)
+    // AudioNodes are in the oilpan heap but they are still reference counted at
+    // the same time. This is because we are not allowed to stop the audio
+    // thread and thus the audio thread cannot allocate objects in the oilpan
+    // heap.
+    // The m_keepAlive handle is used to keep a persistent reference to this
+    // AudioNode while someone has a reference to this AudioNode through a
+    // RefPtr.
+    GC_PLUGIN_IGNORE("http://crbug.com/353083")
+    OwnPtr<Persistent<AudioNode> > m_keepAlive;
+#endif
+
     double m_lastProcessingTime;
     double m_lastNonSilentTime;
 

third_party/WebKit/Source/modules/webaudio/AudioNode.idl

@@ -23,6 +23,7 @@
  */
 
 [
+    WillBeGarbageCollected,
     Conditional=WEB_AUDIO,
     Custom=Wrap,
 ] interface AudioNode : EventTarget {

third_party/WebKit/Source/modules/webaudio/AudioNodeOutput.h

@@ -140,8 +140,7 @@ private:
     unsigned m_renderingFanOutCount;
     unsigned m_renderingParamFanOutCount;
 
-    HashSet<RefPtr<AudioParam> > m_params;
-    typedef HashSet<RefPtr<AudioParam> >::iterator ParamsIterator;
+    WillBePersistentHeapHashSet<RefPtrWillBeMember<AudioParam> > m_params;
 };
 
 } // namespace WebCore

third_party/WebKit/Source/modules/webaudio/AudioParam.h

@@ -43,14 +43,14 @@ namespace WebCore {
 
 class AudioNodeOutput;
 
-class AudioParam FINAL : public RefCounted<AudioParam>, public ScriptWrappable, public AudioSummingJunction {
+class AudioParam FINAL : public RefCountedWillBeGarbageCollectedFinalized<AudioParam>, public ScriptWrappable, public AudioSummingJunction {
 public:
     static const double DefaultSmoothingConstant;
     static const double SnapThreshold;
 
-    static PassRefPtr<AudioParam> create(AudioContext* context, const String& name, double defaultValue, double minValue, double maxValue, unsigned units = 0)
+    static PassRefPtrWillBeRawPtr<AudioParam> create(AudioContext* context, const String& name, double defaultValue, double minValue, double maxValue, unsigned units = 0)
     {
-        return adoptRef(new AudioParam(context, name, defaultValue, minValue, maxValue, units));
+        return adoptRefWillBeNoop(new AudioParam(context, name, defaultValue, minValue, maxValue, units));
     }
 
     // AudioSummingJunction
@@ -102,6 +102,8 @@ public:
     void connect(AudioNodeOutput*);
     void disconnect(AudioNodeOutput*);
 
+    void trace(Visitor*) { }
+
 protected:
     AudioParam(AudioContext* context, const String& name, double defaultValue, double minValue, double maxValue, unsigned units = 0)
         : AudioSummingJunction(context)

third_party/WebKit/Source/modules/webaudio/AudioParam.idl

@@ -27,6 +27,7 @@
  */
 
 [
+    WillBeGarbageCollected,
     Conditional=WEB_AUDIO
 ] interface AudioParam {
     attribute float value;

third_party/WebKit/Source/modules/webaudio/AudioProcessingEvent.cpp

@@ -35,7 +35,7 @@ PassRefPtrWillBeRawPtr<AudioProcessingEvent> AudioProcessingEvent::create()
     return adoptRefWillBeNoop(new AudioProcessingEvent);
 }
 
-PassRefPtrWillBeRawPtr<AudioProcessingEvent> AudioProcessingEvent::create(PassRefPtr<AudioBuffer> inputBuffer, PassRefPtr<AudioBuffer> outputBuffer, double playbackTime)
+PassRefPtrWillBeRawPtr<AudioProcessingEvent> AudioProcessingEvent::create(PassRefPtrWillBeRawPtr<AudioBuffer> inputBuffer, PassRefPtrWillBeRawPtr<AudioBuffer> outputBuffer, double playbackTime)
 {
     return adoptRefWillBeNoop(new AudioProcessingEvent(inputBuffer, outputBuffer, playbackTime));
 }
@@ -45,7 +45,7 @@ AudioProcessingEvent::AudioProcessingEvent()
     ScriptWrappable::init(this);
 }
 
-AudioProcessingEvent::AudioProcessingEvent(PassRefPtr<AudioBuffer> inputBuffer, PassRefPtr<AudioBuffer> outputBuffer, double playbackTime)
+AudioProcessingEvent::AudioProcessingEvent(PassRefPtrWillBeRawPtr<AudioBuffer> inputBuffer, PassRefPtrWillBeRawPtr<AudioBuffer> outputBuffer, double playbackTime)
     : Event(EventTypeNames::audioprocess, true, false)
     , m_inputBuffer(inputBuffer)
     , m_outputBuffer(outputBuffer)
@@ -65,6 +65,8 @@ const AtomicString& AudioProcessingEvent::interfaceName() const
 
 void AudioProcessingEvent::trace(Visitor* visitor)
 {
+    visitor->trace(m_inputBuffer);
+    visitor->trace(m_outputBuffer);
     Event::trace(visitor);
 }
 

third_party/WebKit/Source/modules/webaudio/AudioProcessingEvent.h

@@ -37,7 +37,7 @@ class AudioBuffer;
 class AudioProcessingEvent FINAL : public Event {
 public:
     static PassRefPtrWillBeRawPtr<AudioProcessingEvent> create();
-    static PassRefPtrWillBeRawPtr<AudioProcessingEvent> create(PassRefPtr<AudioBuffer> inputBuffer, PassRefPtr<AudioBuffer> outputBuffer, double playbackTime);
+    static PassRefPtrWillBeRawPtr<AudioProcessingEvent> create(PassRefPtrWillBeRawPtr<AudioBuffer> inputBuffer, PassRefPtrWillBeRawPtr<AudioBuffer> outputBuffer, double playbackTime);
 
     virtual ~AudioProcessingEvent();
 
@@ -51,10 +51,10 @@ public:
 
 private:
     AudioProcessingEvent();
-    AudioProcessingEvent(PassRefPtr<AudioBuffer> inputBuffer, PassRefPtr<AudioBuffer> outputBuffer, double playbackTime);
+    AudioProcessingEvent(PassRefPtrWillBeRawPtr<AudioBuffer> inputBuffer, PassRefPtrWillBeRawPtr<AudioBuffer> outputBuffer, double playbackTime);
 
-    RefPtr<AudioBuffer> m_inputBuffer;
-    RefPtr<AudioBuffer> m_outputBuffer;
+    RefPtrWillBeMember<AudioBuffer> m_inputBuffer;
+    RefPtrWillBeMember<AudioBuffer> m_outputBuffer;
     double m_playbackTime;
 };
 

third_party/WebKit/Source/modules/webaudio/AudioSummingJunction.h

@@ -26,6 +26,7 @@
 #define AudioSummingJunction_h
 
 #include "platform/audio/AudioBus.h"
+#include "platform/heap/Handle.h"
 #include "wtf/HashSet.h"
 #include "wtf/Vector.h"
 
@@ -60,7 +61,7 @@ public:
     virtual void didUpdate() = 0;
 
 protected:
-    RefPtr<AudioContext> m_context;
+    RefPtrWillBePersistent<AudioContext> m_context;
 
     // m_outputs contains the AudioNodeOutputs representing current connections which are not disabled.
     // The rendering code should never use this directly, but instead uses m_renderingOutputs.

third_party/WebKit/Source/modules/webaudio/BiquadFilterNode.h

@@ -46,9 +46,9 @@ public:
         ALLPASS = 7
     };
 
-    static PassRefPtr<BiquadFilterNode> create(AudioContext* context, float sampleRate)
+    static PassRefPtrWillBeRawPtr<BiquadFilterNode> create(AudioContext* context, float sampleRate)
     {
-        return adoptRef(new BiquadFilterNode(context, sampleRate));
+        return adoptRefWillBeNoop(new BiquadFilterNode(context, sampleRate));
     }
 
     String type() const;

third_party/WebKit/Source/modules/webaudio/BiquadProcessor.h

@@ -80,10 +80,10 @@ public:
 private:
     FilterType m_type;
 
-    RefPtr<AudioParam> m_parameter1;
-    RefPtr<AudioParam> m_parameter2;
-    RefPtr<AudioParam> m_parameter3;
-    RefPtr<AudioParam> m_parameter4;
+    RefPtrWillBePersistent<AudioParam> m_parameter1;
+    RefPtrWillBePersistent<AudioParam> m_parameter2;
+    RefPtrWillBePersistent<AudioParam> m_parameter3;
+    RefPtrWillBePersistent<AudioParam> m_parameter4;
 
     // so DSP kernels know when to re-compute coefficients
     bool m_filterCoefficientsDirty;

third_party/WebKit/Source/modules/webaudio/ChannelMergerNode.cpp

@@ -40,12 +40,12 @@ const unsigned DefaultNumberOfOutputChannels = 1;
 
 namespace WebCore {
 
-PassRefPtr<ChannelMergerNode> ChannelMergerNode::create(AudioContext* context, float sampleRate, unsigned numberOfInputs)
+PassRefPtrWillBeRawPtr<ChannelMergerNode> ChannelMergerNode::create(AudioContext* context, float sampleRate, unsigned numberOfInputs)
 {
     if (!numberOfInputs || numberOfInputs > AudioContext::maxNumberOfChannels())
         return nullptr;
 
-    return adoptRef(new ChannelMergerNode(context, sampleRate, numberOfInputs));
+    return adoptRefWillBeNoop(new ChannelMergerNode(context, sampleRate, numberOfInputs));
 }
 
 ChannelMergerNode::ChannelMergerNode(AudioContext* context, float sampleRate, unsigned numberOfInputs)

third_party/WebKit/Source/modules/webaudio/ChannelMergerNode.h

@@ -38,7 +38,7 @@ class AudioContext;
 
 class ChannelMergerNode FINAL : public AudioNode {
 public:
-    static PassRefPtr<ChannelMergerNode> create(AudioContext*, float sampleRate, unsigned numberOfInputs);
+    static PassRefPtrWillBeRawPtr<ChannelMergerNode> create(AudioContext*, float sampleRate, unsigned numberOfInputs);
 
     // AudioNode
     virtual void process(size_t framesToProcess) OVERRIDE;

third_party/WebKit/Source/modules/webaudio/ChannelSplitterNode.cpp

@@ -34,12 +34,12 @@
 
 namespace WebCore {
 
-PassRefPtr<ChannelSplitterNode> ChannelSplitterNode::create(AudioContext* context, float sampleRate, unsigned numberOfOutputs)
+PassRefPtrWillBeRawPtr<ChannelSplitterNode> ChannelSplitterNode::create(AudioContext* context, float sampleRate, unsigned numberOfOutputs)
 {
     if (!numberOfOutputs || numberOfOutputs > AudioContext::maxNumberOfChannels())
         return nullptr;
 
-    return adoptRef(new ChannelSplitterNode(context, sampleRate, numberOfOutputs));
+    return adoptRefWillBeNoop(new ChannelSplitterNode(context, sampleRate, numberOfOutputs));
 }
 
 ChannelSplitterNode::ChannelSplitterNode(AudioContext* context, float sampleRate, unsigned numberOfOutputs)

third_party/WebKit/Source/modules/webaudio/ChannelSplitterNode.h

@@ -34,7 +34,7 @@ class AudioContext;
 
 class ChannelSplitterNode FINAL : public AudioNode {
 public:
-    static PassRefPtr<ChannelSplitterNode> create(AudioContext*, float sampleRate, unsigned numberOfOutputs);
+    static PassRefPtrWillBeRawPtr<ChannelSplitterNode> create(AudioContext*, float sampleRate, unsigned numberOfOutputs);
 
     // AudioNode
     virtual void process(size_t framesToProcess) OVERRIDE;

third_party/WebKit/Source/modules/webaudio/ConvolverNode.cpp

@@ -179,6 +179,12 @@ double ConvolverNode::latencyTime() const
     return std::numeric_limits<double>::infinity();
 }
 
+void ConvolverNode::trace(Visitor* visitor)
+{
+    visitor->trace(m_buffer);
+    AudioNode::trace(visitor);
+}
+
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)

third_party/WebKit/Source/modules/webaudio/ConvolverNode.h

@@ -38,9 +38,9 @@ class Reverb;
 
 class ConvolverNode FINAL : public AudioNode {
 public:
-    static PassRefPtr<ConvolverNode> create(AudioContext* context, float sampleRate)
+    static PassRefPtrWillBeRawPtr<ConvolverNode> create(AudioContext* context, float sampleRate)
     {
-        return adoptRef(new ConvolverNode(context, sampleRate));
+        return adoptRefWillBeNoop(new ConvolverNode(context, sampleRate));
     }
 
     virtual ~ConvolverNode();
@@ -57,6 +57,8 @@ public:
     bool normalize() const { return m_normalize; }
     void setNormalize(bool normalize) { m_normalize = normalize; }
 
+    virtual void trace(Visitor*) OVERRIDE;
+
 private:
     ConvolverNode(AudioContext*, float sampleRate);
 
@@ -64,7 +66,7 @@ private:
     virtual double latencyTime() const OVERRIDE;
 
     OwnPtr<Reverb> m_reverb;
-    RefPtr<AudioBuffer> m_buffer;
+    RefPtrWillBeMember<AudioBuffer> m_buffer;
 
     // This synchronizes dynamic changes to the convolution impulse response with process().
     mutable Mutex m_processLock;

third_party/WebKit/Source/modules/webaudio/DefaultAudioDestinationNode.h

@@ -36,9 +36,9 @@ class ExceptionState;
 
 class DefaultAudioDestinationNode FINAL : public AudioDestinationNode {
 public:
-    static PassRefPtr<DefaultAudioDestinationNode> create(AudioContext* context)
+    static PassRefPtrWillBeRawPtr<DefaultAudioDestinationNode> create(AudioContext* context)
     {
-        return adoptRef(new DefaultAudioDestinationNode(context));
+        return adoptRefWillBeNoop(new DefaultAudioDestinationNode(context));
     }
 
     virtual ~DefaultAudioDestinationNode();

third_party/WebKit/Source/modules/webaudio/DelayNode.h

@@ -36,9 +36,9 @@ class ExceptionState;
 
 class DelayNode FINAL : public AudioBasicProcessorNode {
 public:
-    static PassRefPtr<DelayNode> create(AudioContext* context, float sampleRate, double maxDelayTime, ExceptionState& exceptionState)
+    static PassRefPtrWillBeRawPtr<DelayNode> create(AudioContext* context, float sampleRate, double maxDelayTime, ExceptionState& exceptionState)
     {
-        return adoptRef(new DelayNode(context, sampleRate, maxDelayTime, exceptionState));
+        return adoptRefWillBeNoop(new DelayNode(context, sampleRate, maxDelayTime, exceptionState));
     }
 
     AudioParam* delayTime();

third_party/WebKit/Source/modules/webaudio/DelayProcessor.h

@@ -46,7 +46,7 @@ public:
     double maxDelayTime() { return m_maxDelayTime; }
 private:
 
-    RefPtr<AudioParam> m_delayTime;
+    RefPtrWillBePersistent<AudioParam> m_delayTime;
     double m_maxDelayTime;
 };
 

third_party/WebKit/Source/modules/webaudio/DynamicsCompressorNode.cpp

@@ -113,6 +113,17 @@ double DynamicsCompressorNode::latencyTime() const
     return m_dynamicsCompressor->latencyTime();
 }
 
+void DynamicsCompressorNode::trace(Visitor* visitor)
+{
+    visitor->trace(m_threshold);
+    visitor->trace(m_knee);
+    visitor->trace(m_ratio);
+    visitor->trace(m_reduction);
+    visitor->trace(m_attack);
+    visitor->trace(m_release);
+    AudioNode::trace(visitor);
+}
+
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)

third_party/WebKit/Source/modules/webaudio/DynamicsCompressorNode.h

@@ -35,9 +35,9 @@ class DynamicsCompressor;
 
 class DynamicsCompressorNode FINAL : public AudioNode {
 public:
-    static PassRefPtr<DynamicsCompressorNode> create(AudioContext* context, float sampleRate)
+    static PassRefPtrWillBeRawPtr<DynamicsCompressorNode> create(AudioContext* context, float sampleRate)
     {
-        return adoptRef(new DynamicsCompressorNode(context, sampleRate));
+        return adoptRefWillBeNoop(new DynamicsCompressorNode(context, sampleRate));
     }
 
     virtual ~DynamicsCompressorNode();
@@ -57,6 +57,8 @@ public:
     // Amount by which the compressor is currently compressing the signal in decibels.
     AudioParam* reduction() { return m_reduction.get(); }
 
+    virtual void trace(Visitor*) OVERRIDE;
+
 private:
     virtual double tailTime() const OVERRIDE;
     virtual double latencyTime() const OVERRIDE;
@@ -64,12 +66,12 @@ private:
     DynamicsCompressorNode(AudioContext*, float sampleRate);
 
     OwnPtr<DynamicsCompressor> m_dynamicsCompressor;
-    RefPtr<AudioParam> m_threshold;
-    RefPtr<AudioParam> m_knee;
-    RefPtr<AudioParam> m_ratio;
-    RefPtr<AudioParam> m_reduction;
-    RefPtr<AudioParam> m_attack;
-    RefPtr<AudioParam> m_release;
+    RefPtrWillBeMember<AudioParam> m_threshold;
+    RefPtrWillBeMember<AudioParam> m_knee;
+    RefPtrWillBeMember<AudioParam> m_ratio;
+    RefPtrWillBeMember<AudioParam> m_reduction;
+    RefPtrWillBeMember<AudioParam> m_attack;
+    RefPtrWillBeMember<AudioParam> m_release;
 };
 
 } // namespace WebCore

third_party/WebKit/Source/modules/webaudio/GainNode.cpp

@@ -108,6 +108,12 @@ void GainNode::checkNumberOfChannelsForInput(AudioNodeInput* input)
     AudioNode::checkNumberOfChannelsForInput(input);
 }
 
+void GainNode::trace(Visitor* visitor)
+{
+    visitor->trace(m_gain);
+    AudioNode::trace(visitor);
+}
+
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)

third_party/WebKit/Source/modules/webaudio/GainNode.h

@@ -39,9 +39,9 @@ class AudioContext;
 
 class GainNode FINAL : public AudioNode {
 public:
-    static PassRefPtr<GainNode> create(AudioContext* context, float sampleRate)
+    static PassRefPtrWillBeRawPtr<GainNode> create(AudioContext* context, float sampleRate)
     {
-        return adoptRef(new GainNode(context, sampleRate));
+        return adoptRefWillBeNoop(new GainNode(context, sampleRate));
     }
 
     // AudioNode
@@ -53,6 +53,8 @@ public:
     // JavaScript interface
     AudioParam* gain() { return m_gain.get(); }
 
+    virtual void trace(Visitor*) OVERRIDE;
+
 private:
     virtual double tailTime() const OVERRIDE { return 0; }
     virtual double latencyTime() const OVERRIDE { return 0; }
@@ -60,7 +62,7 @@ private:
     GainNode(AudioContext*, float sampleRate);
 
     float m_lastGain; // for de-zippering
-    RefPtr<AudioParam> m_gain;
+    RefPtrWillBeMember<AudioParam> m_gain;
 
     AudioFloatArray m_sampleAccurateGainValues;
 };

third_party/WebKit/Source/modules/webaudio/MediaElementAudioSourceNode.cpp

@@ -41,9 +41,9 @@ const unsigned maxSampleRate = 192000;
 
 namespace WebCore {
 
-PassRefPtr<MediaElementAudioSourceNode> MediaElementAudioSourceNode::create(AudioContext* context, HTMLMediaElement* mediaElement)
+PassRefPtrWillBeRawPtr<MediaElementAudioSourceNode> MediaElementAudioSourceNode::create(AudioContext* context, HTMLMediaElement* mediaElement)
 {
-    return adoptRef(new MediaElementAudioSourceNode(context, mediaElement));
+    return adoptRefWillBeNoop(new MediaElementAudioSourceNode(context, mediaElement));
 }
 
 MediaElementAudioSourceNode::MediaElementAudioSourceNode(AudioContext* context, HTMLMediaElement* mediaElement)

third_party/WebKit/Source/modules/webaudio/MediaElementAudioSourceNode.h

@@ -41,7 +41,7 @@ class HTMLMediaElement;
 
 class MediaElementAudioSourceNode FINAL : public AudioSourceNode, public AudioSourceProviderClient {
 public:
-    static PassRefPtr<MediaElementAudioSourceNode> create(AudioContext*, HTMLMediaElement*);
+    static PassRefPtrWillBeRawPtr<MediaElementAudioSourceNode> create(AudioContext*, HTMLMediaElement*);
 
     virtual ~MediaElementAudioSourceNode();
 

third_party/WebKit/Source/modules/webaudio/MediaStreamAudioDestinationNode.cpp

@@ -37,9 +37,9 @@
 
 namespace WebCore {
 
-PassRefPtr<MediaStreamAudioDestinationNode> MediaStreamAudioDestinationNode::create(AudioContext* context, size_t numberOfChannels)
+PassRefPtrWillBeRawPtr<MediaStreamAudioDestinationNode> MediaStreamAudioDestinationNode::create(AudioContext* context, size_t numberOfChannels)
 {
-    return adoptRef(new MediaStreamAudioDestinationNode(context, numberOfChannels));
+    return adoptRefWillBeNoop(new MediaStreamAudioDestinationNode(context, numberOfChannels));
 }
 
 MediaStreamAudioDestinationNode::MediaStreamAudioDestinationNode(AudioContext* context, size_t numberOfChannels)

third_party/WebKit/Source/modules/webaudio/MediaStreamAudioDestinationNode.h

@@ -39,7 +39,7 @@ class AudioContext;
 
 class MediaStreamAudioDestinationNode FINAL : public AudioBasicInspectorNode {
 public:
-    static PassRefPtr<MediaStreamAudioDestinationNode> create(AudioContext*, size_t numberOfChannels);
+    static PassRefPtrWillBeRawPtr<MediaStreamAudioDestinationNode> create(AudioContext*, size_t numberOfChannels);
 
     virtual ~MediaStreamAudioDestinationNode();
 

third_party/WebKit/Source/modules/webaudio/MediaStreamAudioSourceNode.cpp

@@ -35,9 +35,9 @@
 
 namespace WebCore {
 
-PassRefPtr<MediaStreamAudioSourceNode> MediaStreamAudioSourceNode::create(AudioContext* context, MediaStream* mediaStream, MediaStreamTrack* audioTrack, PassOwnPtr<AudioSourceProvider> audioSourceProvider)
+PassRefPtrWillBeRawPtr<MediaStreamAudioSourceNode> MediaStreamAudioSourceNode::create(AudioContext* context, MediaStream* mediaStream, MediaStreamTrack* audioTrack, PassOwnPtr<AudioSourceProvider> audioSourceProvider)
 {
-    return adoptRef(new MediaStreamAudioSourceNode(context, mediaStream, audioTrack, audioSourceProvider));
+    return adoptRefWillBeNoop(new MediaStreamAudioSourceNode(context, mediaStream, audioTrack, audioSourceProvider));
 }
 
 MediaStreamAudioSourceNode::MediaStreamAudioSourceNode(AudioContext* context, MediaStream* mediaStream, MediaStreamTrack* audioTrack, PassOwnPtr<AudioSourceProvider> audioSourceProvider)

third_party/WebKit/Source/modules/webaudio/MediaStreamAudioSourceNode.h

@@ -41,7 +41,7 @@ class AudioContext;
 
 class MediaStreamAudioSourceNode FINAL : public AudioSourceNode, public AudioSourceProviderClient {
 public:
-    static PassRefPtr<MediaStreamAudioSourceNode> create(AudioContext*, MediaStream*, MediaStreamTrack*, PassOwnPtr<AudioSourceProvider>);
+    static PassRefPtrWillBeRawPtr<MediaStreamAudioSourceNode> create(AudioContext*, MediaStream*, MediaStreamTrack*, PassOwnPtr<AudioSourceProvider>);
 
     virtual ~MediaStreamAudioSourceNode();
 

third_party/WebKit/Source/modules/webaudio/OfflineAudioCompletionEvent.cpp

@@ -35,7 +35,7 @@ PassRefPtrWillBeRawPtr<OfflineAudioCompletionEvent> OfflineAudioCompletionEvent:
     return adoptRefWillBeNoop(new OfflineAudioCompletionEvent);
 }
 
-PassRefPtrWillBeRawPtr<OfflineAudioCompletionEvent> OfflineAudioCompletionEvent::create(PassRefPtr<AudioBuffer> renderedBuffer)
+PassRefPtrWillBeRawPtr<OfflineAudioCompletionEvent> OfflineAudioCompletionEvent::create(PassRefPtrWillBeRawPtr<AudioBuffer> renderedBuffer)
 {
     return adoptRefWillBeNoop(new OfflineAudioCompletionEvent(renderedBuffer));
 }
@@ -45,7 +45,7 @@ OfflineAudioCompletionEvent::OfflineAudioCompletionEvent()
     ScriptWrappable::init(this);
 }
 
-OfflineAudioCompletionEvent::OfflineAudioCompletionEvent(PassRefPtr<AudioBuffer> renderedBuffer)
+OfflineAudioCompletionEvent::OfflineAudioCompletionEvent(PassRefPtrWillBeRawPtr<AudioBuffer> renderedBuffer)
     : Event(EventTypeNames::complete, true, false)
     , m_renderedBuffer(renderedBuffer)
 {
@@ -63,6 +63,7 @@ const AtomicString& OfflineAudioCompletionEvent::interfaceName() const
 
 void OfflineAudioCompletionEvent::trace(Visitor* visitor)
 {
+    visitor->trace(m_renderedBuffer);
     Event::trace(visitor);
 }
 

third_party/WebKit/Source/modules/webaudio/OfflineAudioCompletionEvent.h

@@ -37,7 +37,7 @@ class AudioBuffer;
 class OfflineAudioCompletionEvent FINAL : public Event {
 public:
     static PassRefPtrWillBeRawPtr<OfflineAudioCompletionEvent> create();
-    static PassRefPtrWillBeRawPtr<OfflineAudioCompletionEvent> create(PassRefPtr<AudioBuffer> renderedBuffer);
+    static PassRefPtrWillBeRawPtr<OfflineAudioCompletionEvent> create(PassRefPtrWillBeRawPtr<AudioBuffer> renderedBuffer);
 
     virtual ~OfflineAudioCompletionEvent();
 
@@ -49,9 +49,9 @@ public:
 
 private:
     OfflineAudioCompletionEvent();
-    explicit OfflineAudioCompletionEvent(PassRefPtr<AudioBuffer> renderedBuffer);
+    explicit OfflineAudioCompletionEvent(PassRefPtrWillBeRawPtr<AudioBuffer> renderedBuffer);
 
-    RefPtr<AudioBuffer> m_renderedBuffer;
+    RefPtrWillBeMember<AudioBuffer> m_renderedBuffer;
 };
 
 } // namespace WebCore

third_party/WebKit/Source/modules/webaudio/OfflineAudioContext.cpp

@@ -36,7 +36,7 @@
 
 namespace WebCore {
 
-PassRefPtr<OfflineAudioContext> OfflineAudioContext::create(ExecutionContext* context, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<OfflineAudioContext> OfflineAudioContext::create(ExecutionContext* context, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState)
 {
     // FIXME: add support for workers.
     if (!context || !context->isDocument()) {
@@ -71,7 +71,7 @@ PassRefPtr<OfflineAudioContext> OfflineAudioContext::create(ExecutionContext* co
         return nullptr;
     }
 
-    RefPtr<OfflineAudioContext> audioContext(adoptRef(new OfflineAudioContext(document, numberOfChannels, numberOfFrames, sampleRate)));
+    RefPtrWillBeRawPtr<OfflineAudioContext> audioContext(adoptRefWillBeThreadSafeRefCountedGarbageCollected(new OfflineAudioContext(document, numberOfChannels, numberOfFrames, sampleRate)));
 
     if (!audioContext->destination()) {
         exceptionState.throwDOMException(

third_party/WebKit/Source/modules/webaudio/OfflineAudioContext.h

@@ -33,7 +33,7 @@ class ExceptionState;
 
 class OfflineAudioContext FINAL : public AudioContext {
 public:
-    static PassRefPtr<OfflineAudioContext> create(ExecutionContext*, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState&);
+    static PassRefPtrWillBeRawPtr<OfflineAudioContext> create(ExecutionContext*, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState&);
 
     virtual ~OfflineAudioContext();
 

third_party/WebKit/Source/modules/webaudio/OfflineAudioContext.idl

@@ -23,6 +23,7 @@
  */
 
 [
+    WillBeGarbageCollected,
     Conditional=WEB_AUDIO,
     Constructor(unsigned long numberOfChannels, unsigned long numberOfFrames, float sampleRate),
     ConstructorCallWith=ExecutionContext,

third_party/WebKit/Source/modules/webaudio/OfflineAudioDestinationNode.cpp

@@ -153,6 +153,12 @@ void OfflineAudioDestinationNode::notifyComplete()
     context()->fireCompletionEvent();
 }
 
+void OfflineAudioDestinationNode::trace(Visitor* visitor)
+{
+    visitor->trace(m_renderTarget);
+    AudioDestinationNode::trace(visitor);
+}
+
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)

third_party/WebKit/Source/modules/webaudio/OfflineAudioDestinationNode.h

@@ -38,9 +38,9 @@ class AudioContext;
 
 class OfflineAudioDestinationNode FINAL : public AudioDestinationNode {
 public:
-    static PassRefPtr<OfflineAudioDestinationNode> create(AudioContext* context, AudioBuffer* renderTarget)
+    static PassRefPtrWillBeRawPtr<OfflineAudioDestinationNode> create(AudioContext* context, AudioBuffer* renderTarget)
     {
-        return adoptRef(new OfflineAudioDestinationNode(context, renderTarget));
+        return adoptRefWillBeNoop(new OfflineAudioDestinationNode(context, renderTarget));
     }
 
     virtual ~OfflineAudioDestinationNode();
@@ -54,6 +54,8 @@ public:
 
     virtual float sampleRate()  const OVERRIDE { return m_renderTarget->sampleRate(); }
 
+    virtual void trace(Visitor*) OVERRIDE;
+
 private:
     class OfflineRenderingTask;
     friend class OfflineRenderingTask;
@@ -61,7 +63,7 @@ private:
     OfflineAudioDestinationNode(AudioContext*, AudioBuffer* renderTarget);
 
     // This AudioNode renders into this AudioBuffer.
-    RefPtr<AudioBuffer> m_renderTarget;
+    RefPtrWillBeMember<AudioBuffer> m_renderTarget;
 
     // Temporary AudioBus for each render quantum.
     RefPtr<AudioBus> m_renderBus;

third_party/WebKit/Source/modules/webaudio/OscillatorNode.cpp

@@ -43,9 +43,9 @@ namespace WebCore {
 
 using namespace VectorMath;
 
-PassRefPtr<OscillatorNode> OscillatorNode::create(AudioContext* context, float sampleRate)
+PassRefPtrWillBeRawPtr<OscillatorNode> OscillatorNode::create(AudioContext* context, float sampleRate)
 {
-    return adoptRef(new OscillatorNode(context, sampleRate));
+    return adoptRefWillBeNoop(new OscillatorNode(context, sampleRate));
 }
 
 OscillatorNode::OscillatorNode(AudioContext* context, float sampleRate)
@@ -116,22 +116,22 @@ bool OscillatorNode::setType(unsigned type)
 
     switch (type) {
     case SINE: {
-        DEFINE_STATIC_REF(PeriodicWave, periodicWaveSine, (PeriodicWave::createSine(sampleRate)));
+        DEFINE_STATIC_REF_WILL_BE_PERSISTENT(PeriodicWave, periodicWaveSine, (PeriodicWave::createSine(sampleRate)));
         periodicWave = periodicWaveSine;
         break;
     }
     case SQUARE: {
-        DEFINE_STATIC_REF(PeriodicWave, periodicWaveSquare, (PeriodicWave::createSquare(sampleRate)));
+        DEFINE_STATIC_REF_WILL_BE_PERSISTENT(PeriodicWave, periodicWaveSquare, (PeriodicWave::createSquare(sampleRate)));
         periodicWave = periodicWaveSquare;
         break;
     }
     case SAWTOOTH: {
-        DEFINE_STATIC_REF(PeriodicWave, periodicWaveSawtooth, (PeriodicWave::createSawtooth(sampleRate)));
+        DEFINE_STATIC_REF_WILL_BE_PERSISTENT(PeriodicWave, periodicWaveSawtooth, (PeriodicWave::createSawtooth(sampleRate)));
         periodicWave = periodicWaveSawtooth;
         break;
     }
     case TRIANGLE: {
-        DEFINE_STATIC_REF(PeriodicWave, periodicWaveTriangle, (PeriodicWave::createTriangle(sampleRate)));
+        DEFINE_STATIC_REF_WILL_BE_PERSISTENT(PeriodicWave, periodicWaveTriangle, (PeriodicWave::createTriangle(sampleRate)));
         periodicWave = periodicWaveTriangle;
         break;
     }
@@ -341,6 +341,14 @@ bool OscillatorNode::propagatesSilence() const
     return !isPlayingOrScheduled() || hasFinished() || !m_periodicWave.get();
 }
 
+void OscillatorNode::trace(Visitor* visitor)
+{
+    visitor->trace(m_frequency);
+    visitor->trace(m_detune);
+    visitor->trace(m_periodicWave);
+    AudioScheduledSourceNode::trace(visitor);
+}
+
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)

third_party/WebKit/Source/modules/webaudio/OscillatorNode.h

@@ -52,7 +52,7 @@ public:
         CUSTOM = 4
     };
 
-    static PassRefPtr<OscillatorNode> create(AudioContext*, float sampleRate);
+    static PassRefPtrWillBeRawPtr<OscillatorNode> create(AudioContext*, float sampleRate);
 
     virtual ~OscillatorNode();
 
@@ -68,6 +68,8 @@ public:
 
     void setPeriodicWave(PeriodicWave*);
 
+    virtual void trace(Visitor*) OVERRIDE;
+
 private:
     OscillatorNode(AudioContext*, float sampleRate);
 
@@ -82,10 +84,10 @@ private:
     unsigned short m_type;
 
     // Frequency value in Hertz.
-    RefPtr<AudioParam> m_frequency;
+    RefPtrWillBeMember<AudioParam> m_frequency;
 
     // Detune value (deviating from the frequency) in Cents.
-    RefPtr<AudioParam> m_detune;
+    RefPtrWillBeMember<AudioParam> m_detune;
 
     bool m_firstRender;
 
@@ -100,7 +102,7 @@ private:
     AudioFloatArray m_phaseIncrements;
     AudioFloatArray m_detuneValues;
 
-    RefPtr<PeriodicWave> m_periodicWave;
+    RefPtrWillBeMember<PeriodicWave> m_periodicWave;
 };
 
 } // namespace WebCore

third_party/WebKit/Source/modules/webaudio/PannerNode.h

@@ -54,9 +54,9 @@ public:
         DopplerRateDirty = 0x4,
     };
 
-    static PassRefPtr<PannerNode> create(AudioContext* context, float sampleRate)
+    static PassRefPtrWillBeRawPtr<PannerNode> create(AudioContext* context, float sampleRate)
     {
-        return adoptRef(new PannerNode(context, sampleRate));
+        return adoptRefWillBeNoop(new PannerNode(context, sampleRate));
     }
 
     virtual ~PannerNode();

third_party/WebKit/Source/modules/webaudio/PeriodicWave.cpp

@@ -45,12 +45,12 @@ namespace WebCore {
 
 using namespace VectorMath;
 
-PassRefPtr<PeriodicWave> PeriodicWave::create(float sampleRate, Float32Array* real, Float32Array* imag)
+PassRefPtrWillBeRawPtr<PeriodicWave> PeriodicWave::create(float sampleRate, Float32Array* real, Float32Array* imag)
 {
     bool isGood = real && imag && real->length() == imag->length();
     ASSERT(isGood);
     if (isGood) {
-        RefPtr<PeriodicWave> periodicWave = adoptRef(new PeriodicWave(sampleRate));
+        RefPtrWillBeRawPtr<PeriodicWave> periodicWave = adoptRefWillBeNoop(new PeriodicWave(sampleRate));
         size_t numberOfComponents = real->length();
         periodicWave->createBandLimitedTables(real->data(), imag->data(), numberOfComponents);
         return periodicWave;
@@ -58,30 +58,30 @@ PassRefPtr<PeriodicWave> PeriodicWave::create(float sampleRate, Float32Array* re
     return nullptr;
 }
 
-PassRefPtr<PeriodicWave> PeriodicWave::createSine(float sampleRate)
+PassRefPtrWillBeRawPtr<PeriodicWave> PeriodicWave::createSine(float sampleRate)
 {
-    RefPtr<PeriodicWave> periodicWave = adoptRef(new PeriodicWave(sampleRate));
+    RefPtrWillBeRawPtr<PeriodicWave> periodicWave = adoptRefWillBeNoop(new PeriodicWave(sampleRate));
     periodicWave->generateBasicWaveform(OscillatorNode::SINE);
     return periodicWave;
 }
 
-PassRefPtr<PeriodicWave> PeriodicWave::createSquare(float sampleRate)
+PassRefPtrWillBeRawPtr<PeriodicWave> PeriodicWave::createSquare(float sampleRate)
 {
-    RefPtr<PeriodicWave> periodicWave = adoptRef(new PeriodicWave(sampleRate));
+    RefPtrWillBeRawPtr<PeriodicWave> periodicWave = adoptRefWillBeNoop(new PeriodicWave(sampleRate));
     periodicWave->generateBasicWaveform(OscillatorNode::SQUARE);
     return periodicWave;
 }
 
-PassRefPtr<PeriodicWave> PeriodicWave::createSawtooth(float sampleRate)
+PassRefPtrWillBeRawPtr<PeriodicWave> PeriodicWave::createSawtooth(float sampleRate)
 {
-    RefPtr<PeriodicWave> periodicWave = adoptRef(new PeriodicWave(sampleRate));
+    RefPtrWillBeRawPtr<PeriodicWave> periodicWave = adoptRefWillBeNoop(new PeriodicWave(sampleRate));
     periodicWave->generateBasicWaveform(OscillatorNode::SAWTOOTH);
     return periodicWave;
 }
 
-PassRefPtr<PeriodicWave> PeriodicWave::createTriangle(float sampleRate)
+PassRefPtrWillBeRawPtr<PeriodicWave> PeriodicWave::createTriangle(float sampleRate)
 {
-    RefPtr<PeriodicWave> periodicWave = adoptRef(new PeriodicWave(sampleRate));
+    RefPtrWillBeRawPtr<PeriodicWave> periodicWave = adoptRefWillBeNoop(new PeriodicWave(sampleRate));
     periodicWave->generateBasicWaveform(OscillatorNode::TRIANGLE);
     return periodicWave;
 }

third_party/WebKit/Source/modules/webaudio/PeriodicWave.h

@@ -40,15 +40,15 @@
 
 namespace WebCore {
 
-class PeriodicWave : public ScriptWrappable, public RefCounted<PeriodicWave> {
+class PeriodicWave : public RefCountedWillBeGarbageCollectedFinalized<PeriodicWave>, public ScriptWrappable {
 public:
-    static PassRefPtr<PeriodicWave> createSine(float sampleRate);
-    static PassRefPtr<PeriodicWave> createSquare(float sampleRate);
-    static PassRefPtr<PeriodicWave> createSawtooth(float sampleRate);
-    static PassRefPtr<PeriodicWave> createTriangle(float sampleRate);
+    static PassRefPtrWillBeRawPtr<PeriodicWave> createSine(float sampleRate);
+    static PassRefPtrWillBeRawPtr<PeriodicWave> createSquare(float sampleRate);
+    static PassRefPtrWillBeRawPtr<PeriodicWave> createSawtooth(float sampleRate);
+    static PassRefPtrWillBeRawPtr<PeriodicWave> createTriangle(float sampleRate);
 
     // Creates an arbitrary periodic wave given the frequency components (Fourier coefficients).
-    static PassRefPtr<PeriodicWave> create(float sampleRate, Float32Array* real, Float32Array* imag);
+    static PassRefPtrWillBeRawPtr<PeriodicWave> create(float sampleRate, Float32Array* real, Float32Array* imag);
 
     // Returns pointers to the lower and higher wave data for the pitch range containing
     // the given fundamental frequency. These two tables are in adjacent "pitch" ranges
@@ -63,6 +63,8 @@ public:
 
     unsigned periodicWaveSize() const { return m_periodicWaveSize; }
 
+    void trace(Visitor*) { }
+
 private:
     explicit PeriodicWave(float sampleRate);
 

third_party/WebKit/Source/modules/webaudio/PeriodicWave.idl

@@ -24,6 +24,7 @@
 
 // PeriodicWave represents a periodic audio waveform given by its Fourier coefficients.
 [
+    WillBeGarbageCollected,
     Conditional=WEB_AUDIO
 ] interface PeriodicWave {
 

third_party/WebKit/Source/modules/webaudio/ScriptProcessorNode.cpp

@@ -56,7 +56,7 @@ static size_t chooseBufferSize()
     return bufferSize;
 }
 
-PassRefPtr<ScriptProcessorNode> ScriptProcessorNode::create(AudioContext* context, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels)
+PassRefPtrWillBeRawPtr<ScriptProcessorNode> ScriptProcessorNode::create(AudioContext* context, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels)
 {
     // Check for valid buffer size.
     switch (bufferSize) {
@@ -84,7 +84,7 @@ PassRefPtr<ScriptProcessorNode> ScriptProcessorNode::create(AudioContext* contex
     if (numberOfOutputChannels > AudioContext::maxNumberOfChannels())
         return nullptr;
 
-    return adoptRef(new ScriptProcessorNode(context, sampleRate, bufferSize, numberOfInputChannels, numberOfOutputChannels));
+    return adoptRefWillBeNoop(new ScriptProcessorNode(context, sampleRate, bufferSize, numberOfInputChannels, numberOfOutputChannels));
 }
 
 ScriptProcessorNode::ScriptProcessorNode(AudioContext* context, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels)
@@ -127,8 +127,8 @@ void ScriptProcessorNode::initialize()
     // Create double buffers on both the input and output sides.
     // These AudioBuffers will be directly accessed in the main thread by JavaScript.
     for (unsigned i = 0; i < 2; ++i) {
-        RefPtr<AudioBuffer> inputBuffer = m_numberOfInputChannels ? AudioBuffer::create(m_numberOfInputChannels, bufferSize(), sampleRate) : nullptr;
-        RefPtr<AudioBuffer> outputBuffer = m_numberOfOutputChannels ? AudioBuffer::create(m_numberOfOutputChannels, bufferSize(), sampleRate) : nullptr;
+        RefPtrWillBeRawPtr<AudioBuffer> inputBuffer = m_numberOfInputChannels ? AudioBuffer::create(m_numberOfInputChannels, bufferSize(), sampleRate) : nullptr;
+        RefPtrWillBeRawPtr<AudioBuffer> outputBuffer = m_numberOfOutputChannels ? AudioBuffer::create(m_numberOfOutputChannels, bufferSize(), sampleRate) : nullptr;
 
         m_inputBuffers.append(inputBuffer);
         m_outputBuffers.append(outputBuffer);
@@ -284,6 +284,13 @@ double ScriptProcessorNode::latencyTime() const
     return std::numeric_limits<double>::infinity();
 }
 
+void ScriptProcessorNode::trace(Visitor* visitor)
+{
+    visitor->trace(m_inputBuffers);
+    visitor->trace(m_outputBuffers);
+    AudioNode::trace(visitor);
+}
+
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)

third_party/WebKit/Source/modules/webaudio/ScriptProcessorNode.h

@@ -50,7 +50,7 @@ public:
     // This value controls how frequently the onaudioprocess event handler is called and how many sample-frames need to be processed each call.
     // Lower numbers for bufferSize will result in a lower (better) latency. Higher numbers will be necessary to avoid audio breakup and glitches.
     // The value chosen must carefully balance between latency and audio quality.
-    static PassRefPtr<ScriptProcessorNode> create(AudioContext*, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels);
+    static PassRefPtrWillBeRawPtr<ScriptProcessorNode> create(AudioContext*, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels);
 
     virtual ~ScriptProcessorNode();
 
@@ -63,6 +63,7 @@ public:
 
     DEFINE_ATTRIBUTE_EVENT_LISTENER(audioprocess);
 
+    void trace(Visitor*);
 
 private:
     virtual double tailTime() const OVERRIDE;
@@ -78,8 +79,8 @@ private:
     void swapBuffers() { m_doubleBufferIndex = 1 - m_doubleBufferIndex; }
     unsigned m_doubleBufferIndex;
     unsigned m_doubleBufferIndexForEvent;
-    Vector<RefPtr<AudioBuffer> > m_inputBuffers;
-    Vector<RefPtr<AudioBuffer> > m_outputBuffers;
+    WillBeHeapVector<RefPtrWillBeMember<AudioBuffer> > m_inputBuffers;
+    WillBeHeapVector<RefPtrWillBeMember<AudioBuffer> > m_outputBuffers;
 
     size_t m_bufferSize;
     unsigned m_bufferReadWriteIndex;

third_party/WebKit/Source/modules/webaudio/WaveShaperNode.h

@@ -36,9 +36,9 @@ class ExceptionState;
 
 class WaveShaperNode FINAL : public AudioBasicProcessorNode {
 public:
-    static PassRefPtr<WaveShaperNode> create(AudioContext* context)
+    static PassRefPtrWillBeRawPtr<WaveShaperNode> create(AudioContext* context)
     {
-        return adoptRef(new WaveShaperNode(context));
+        return adoptRefWillBeNoop(new WaveShaperNode(context));
     }
 
     // setCurve() is called on the main thread.

third_party/WebKit/Source/platform/heap/Handle.h

@@ -777,6 +777,7 @@ template<typename T, typename U> inline bool operator!=(const PassRefPtr<T>& a,
 #define RefCountedWillBeRefCountedGarbageCollected WebCore::RefCountedGarbageCollected
 #define ThreadSafeRefCountedWillBeGarbageCollected WebCore::GarbageCollected
 #define ThreadSafeRefCountedWillBeGarbageCollectedFinalized WebCore::GarbageCollectedFinalized
+#define ThreadSafeRefCountedWillBeThreadSafeRefCountedGarbageCollected WebCore::ThreadSafeRefCountedGarbageCollected
 #define TreeSharedWillBeRefCountedGarbageCollected WebCore::RefCountedGarbageCollected
 #define PersistentWillBeMember WebCore::Member
 #define RefPtrWillBePersistent WebCore::Persistent
@@ -830,6 +831,13 @@ template<typename T> PassRefPtrWillBeRawPtr<T> adoptRefWillBeRefCountedGarbageCo
     return PassRefPtrWillBeRawPtr<T>(adoptRefCountedGarbageCollected(ptr));
 }
 
+template<typename T> PassRefPtrWillBeRawPtr<T> adoptRefWillBeThreadSafeRefCountedGarbageCollected(T* ptr)
+{
+    static const bool isThreadSafeRefCountedGarbageCollected = WTF::IsSubclassOfTemplate<T, ThreadSafeRefCountedGarbageCollected>::value;
+    COMPILE_ASSERT(isThreadSafeRefCountedGarbageCollected, useAdoptRefWillBeNoop);
+    return PassRefPtrWillBeRawPtr<T>(adoptRefCountedGarbageCollected(ptr));
+}
+
 template<typename T> PassOwnPtrWillBeRawPtr<T> adoptPtrWillBeNoop(T* ptr)
 {
     static const bool notRefCountedGarbageCollected = !WTF::IsSubclassOfTemplate<T, RefCountedGarbageCollected>::value;
@@ -862,6 +870,7 @@ public:
 #define RefCountedWillBeRefCountedGarbageCollected WTF::RefCounted
 #define ThreadSafeRefCountedWillBeGarbageCollected WTF::ThreadSafeRefCounted
 #define ThreadSafeRefCountedWillBeGarbageCollectedFinalized WTF::ThreadSafeRefCounted
+#define ThreadSafeRefCountedWillBeThreadSafeRefCountedGarbageCollected WTF::ThreadSafeRefCounted
 #define TreeSharedWillBeRefCountedGarbageCollected WebCore::TreeShared
 #define PersistentWillBeMember WebCore::Persistent
 #define RefPtrWillBePersistent WTF::RefPtr
@@ -901,6 +910,7 @@ public:
 
 template<typename T> PassRefPtrWillBeRawPtr<T> adoptRefWillBeNoop(T* ptr) { return adoptRef(ptr); }
 template<typename T> PassRefPtrWillBeRawPtr<T> adoptRefWillBeRefCountedGarbageCollected(T* ptr) { return adoptRef(ptr); }
+template<typename T> PassRefPtrWillBeRawPtr<T> adoptRefWillBeThreadSafeRefCountedGarbageCollected(T* ptr) { return adoptRef(ptr); }
 template<typename T> PassOwnPtrWillBeRawPtr<T> adoptPtrWillBeNoop(T* ptr) { return adoptPtr(ptr); }
 
 #define WTF_MAKE_FAST_ALLOCATED_WILL_BE_REMOVED WTF_MAKE_FAST_ALLOCATED

third_party/WebKit/Source/platform/heap/Heap.h

@@ -42,6 +42,7 @@
 #include "wtf/ListHashSet.h"
 #include "wtf/OwnPtr.h"
 #include "wtf/PassRefPtr.h"
+#include "wtf/ThreadSafeRefCounted.h"
 
 #include <stdint.h>
 
@@ -70,6 +71,7 @@ class HeapStats;
 class PageMemory;
 template<ThreadAffinity affinity> class ThreadLocalPersistents;
 template<typename T, typename RootsAccessor = ThreadLocalPersistents<ThreadingTrait<T>::Affinity > > class Persistent;
+template<typename T> class CrossThreadPersistent;
 
 PLATFORM_EXPORT size_t osPageSize();
 
@@ -559,6 +561,56 @@ private:
     friend class ThreadState;
 };
 
+// FIXME: This is currently used by the WebAudio code.
+// We should attempt to restructure the WebAudio code so that the main thread
+// alone determines life-time and receives messages about life-time from the
+// audio thread.
+template<typename T>
+class ThreadSafeRefCountedGarbageCollected : public GarbageCollectedFinalized<T>, public WTF::ThreadSafeRefCountedBase {
+    WTF_MAKE_NONCOPYABLE(ThreadSafeRefCountedGarbageCollected);
+
+public:
+    ThreadSafeRefCountedGarbageCollected()
+    {
+        m_keepAlive = adoptPtr(new CrossThreadPersistent<T>(static_cast<T*>(this)));
+    }
+
+    // Override ref to deal with a case where a reference count goes up
+    // from 0 to 1. This can happen in the following scenario:
+    // (1) The reference count becomes 0, but on-stack pointers keep references to the object.
+    // (2) The on-stack pointer is assigned to a RefPtr. The reference count becomes 1.
+    // In this case, we have to resurrect m_keepAlive.
+    void ref()
+    {
+        MutexLocker lock(m_mutex);
+        if (UNLIKELY(!refCount())) {
+            ASSERT(!m_keepAlive);
+            m_keepAlive = adoptPtr(new CrossThreadPersistent<T>(static_cast<T*>(this)));
+        }
+        WTF::ThreadSafeRefCountedBase::ref();
+    }
+
+    // Override deref to deal with our own deallocation based on ref counting.
+    void deref()
+    {
+        MutexLocker lock(m_mutex);
+        if (derefBase()) {
+            ASSERT(m_keepAlive);
+            m_keepAlive.clear();
+        }
+    }
+
+    using GarbageCollectedFinalized<T>::operator new;
+    using GarbageCollectedFinalized<T>::operator delete;
+
+protected:
+    ~ThreadSafeRefCountedGarbageCollected() { }
+
+private:
+    OwnPtr<CrossThreadPersistent<T> > m_keepAlive;
+    mutable Mutex m_mutex;
+};
+
 // The CallbackStack contains all the visitor callbacks used to trace and mark
 // objects. A specific CallbackStack instance contains at most bufferSize elements.
 // If more space is needed a new CallbackStack instance is created and chained