Update Chromium to beta version 37.0.2062.68

Change-Id: I188e3b5aff1bec75566014291b654eb19f5bc8ca Reviewed-by: Andras Becsi <andras.becsi@digia.com>
author: Jocelyn Turcotte <jocelyn.turcotte@digia.com> 2014-08-08 14:30:41 +0200
committer: Jocelyn Turcotte <jocelyn.turcotte@digia.com> 2014-08-12 13:49:54 +0200
commit: ab0a50979b9eb4dfa3320eff7e187e41efedf7a9 (patch)
tree: 498dfb8a97ff3361a9f7486863a52bb4e26bb898 /chromium/third_party/WebKit/Source/modules/webaudio
parent: 4ce69f7403811819800e7c5ae1318b2647e778d1 (diff)
100 files changed, 1486 insertions, 1233 deletions
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AnalyserNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/AnalyserNode.cpp
index 83e6dba02ef..53cdd441f57 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AnalyserNode.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AnalyserNode.cpp
@@ -69,70 +69,47 @@ void AnalyserNode::process(size_t framesToProcess)
         outputBus->copyFrom(*inputBus);
 }
 
-void AnalyserNode::reset()
-{
-    m_analyser.reset();
-}
-
 void AnalyserNode::setFftSize(unsigned size, ExceptionState& exceptionState)
 {
     if (!m_analyser.setFftSize(size)) {
         exceptionState.throwDOMException(
             IndexSizeError,
-            ExceptionMessages::failedToSet(
-                "fftSize",
-                "AnalyserNode",
-                "FFT size (" + String::number(size)
-                + ") must be a power of two between "
-                + String::number(RealtimeAnalyser::MinFFTSize) + " and "
-                + String::number(RealtimeAnalyser::MaxFFTSize) + ", inclusive"));
+            (size < RealtimeAnalyser::MinFFTSize || size > RealtimeAnalyser::MaxFFTSize) ?
+                ExceptionMessages::indexOutsideRange("FFT size", size, RealtimeAnalyser::MinFFTSize, ExceptionMessages::InclusiveBound, RealtimeAnalyser::MaxFFTSize, ExceptionMessages::InclusiveBound)
+                : ("The value provided (" + String::number(size) + ") is not a power of two."));
     }
 }
 
-void AnalyserNode::setMinDecibels(float k, ExceptionState& exceptionState)
+void AnalyserNode::setMinDecibels(double k, ExceptionState& exceptionState)
 {
-    if (k <= maxDecibels()) {
+    if (k < maxDecibels()) {
         m_analyser.setMinDecibels(k);
     } else {
         exceptionState.throwDOMException(
             IndexSizeError,
-            ExceptionMessages::failedToSet(
-                "minDecibels",
-                "AnalyserNode",
-                "minDecibels (" + String::number(k)
-                + ") must be less than or equal maxDecibels (" + String::number(maxDecibels())
-                + ")."));
+            ExceptionMessages::indexExceedsMaximumBound("minDecibels", k, maxDecibels()));
     }
 }
 
-void AnalyserNode::setMaxDecibels(float k, ExceptionState& exceptionState)
+void AnalyserNode::setMaxDecibels(double k, ExceptionState& exceptionState)
 {
-    if (k >= minDecibels()) {
+    if (k > minDecibels()) {
         m_analyser.setMaxDecibels(k);
     } else {
         exceptionState.throwDOMException(
             IndexSizeError,
-            ExceptionMessages::failedToSet(
-                "maxDecibels",
-                "AnalyserNode",
-                "maxDecibels (" + String::number(k)
-                + ") must be greater than or equal minDecibels (" + String::number(minDecibels())
-                + ")."));
+            ExceptionMessages::indexExceedsMinimumBound("maxDecibels", k, minDecibels()));
     }
 }
 
-void AnalyserNode::setSmoothingTimeConstant(float k, ExceptionState& exceptionState)
+void AnalyserNode::setSmoothingTimeConstant(double k, ExceptionState& exceptionState)
 {
     if (k >= 0 && k <= 1) {
         m_analyser.setSmoothingTimeConstant(k);
     } else {
         exceptionState.throwDOMException(
             IndexSizeError,
-            ExceptionMessages::failedToSet(
-                "smoothingTimeConstant",
-                "AnalyserNode",
-                "smoothing value (" + String::number(k)
-                + ") must be between 0 and 1, inclusive."));
+            ExceptionMessages::indexOutsideRange("smoothing value", k, 0.0, ExceptionMessages::InclusiveBound, 1.0, ExceptionMessages::InclusiveBound));
     }
 }
 
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AnalyserNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/AnalyserNode.h
index db2dacb7c83..fba97158f67 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AnalyserNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AnalyserNode.h
@@ -33,18 +33,17 @@ namespace WebCore {
 
 class ExceptionState;
 
-class AnalyserNode : public AudioBasicInspectorNode {
+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();
 
     // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void reset();
+    virtual void process(size_t framesToProcess) OVERRIDE;
 
     // Javascript bindings
     unsigned fftSize() const { return m_analyser.fftSize(); }
@@ -52,19 +51,19 @@ public:
 
     unsigned frequencyBinCount() const { return m_analyser.frequencyBinCount(); }
 
-    void setMinDecibels(float k, ExceptionState&);
-    float minDecibels() const { return m_analyser.minDecibels(); }
+    void setMinDecibels(double k, ExceptionState&);
+    double minDecibels() const { return m_analyser.minDecibels(); }
 
-    void setMaxDecibels(float k, ExceptionState&);
-    float maxDecibels() const { return m_analyser.maxDecibels(); }
+    void setMaxDecibels(double k, ExceptionState&);
+    double maxDecibels() const { return m_analyser.maxDecibels(); }
 
-    void setSmoothingTimeConstant(float k, ExceptionState&);
-    float smoothingTimeConstant() const { return m_analyser.smoothingTimeConstant(); }
+    void setSmoothingTimeConstant(double k, ExceptionState&);
+    double smoothingTimeConstant() const { return m_analyser.smoothingTimeConstant(); }
 
     void getFloatFrequencyData(Float32Array* array) { m_analyser.getFloatFrequencyData(array); }
     void getByteFrequencyData(Uint8Array* array) { m_analyser.getByteFrequencyData(array); }
+    void getFloatTimeDomainData(Float32Array* array) { m_analyser.getFloatTimeDomainData(array); }
     void getByteTimeDomainData(Uint8Array* array) { m_analyser.getByteTimeDomainData(array); }
-
 private:
     virtual double tailTime() const OVERRIDE { return 0; }
     virtual double latencyTime() const OVERRIDE { return 0; }
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AnalyserNode.idl b/chromium/third_party/WebKit/Source/modules/webaudio/AnalyserNode.idl
index f3add662997..4de5f299edc 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AnalyserNode.idl
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AnalyserNode.idl
@@ -29,11 +29,11 @@
     readonly attribute unsigned long frequencyBinCount;
 
     // minDecibels / maxDecibels represent the range to scale the FFT analysis data for conversion to unsigned byte values.
-    [RaisesException=Setter] attribute float minDecibels;
-    [RaisesException=Setter] attribute float maxDecibels;
+    [RaisesException=Setter] attribute double minDecibels;
+    [RaisesException=Setter] attribute double maxDecibels;
 
     // A value from 0.0 -> 1.0 where 0.0 represents no time averaging with the last analysis frame.
-    [RaisesException=Setter] attribute float smoothingTimeConstant;
+    [RaisesException=Setter] attribute double smoothingTimeConstant;
 
     // Copies the current frequency data into the passed array.
     // If the array has fewer elements than the frequencyBinCount, the excess elements will be dropped.
@@ -41,5 +41,6 @@
     void getByteFrequencyData(Uint8Array array);
 
     // Real-time waveform data
+    void getFloatTimeDomainData(Float32Array array);
     void getByteTimeDomainData(Uint8Array array);
 };
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AsyncAudioDecoder.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/AsyncAudioDecoder.cpp
index f1ea34f4c71..db7cedaaf62 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AsyncAudioDecoder.cpp
+++ b/chromium/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
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AsyncAudioDecoder.h b/chromium/third_party/WebKit/Source/modules/webaudio/AsyncAudioDecoder.h
index a615787bd72..7f774564635 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AsyncAudioDecoder.h
+++ b/chromium/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;
 };
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioBasicInspectorNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/AudioBasicInspectorNode.h
index 44d03cb103f..1e7cf824eb7 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioBasicInspectorNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioBasicInspectorNode.h
@@ -39,10 +39,10 @@ public:
     AudioBasicInspectorNode(AudioContext*, float sampleRate, unsigned outputChannelCount);
 
     // AudioNode
-    virtual void pullInputs(size_t framesToProcess);
-    virtual void connect(AudioNode*, unsigned outputIndex, unsigned inputIndex, ExceptionState&);
-    virtual void disconnect(unsigned outputIndex, ExceptionState&);
-    virtual void checkNumberOfChannelsForInput(AudioNodeInput*);
+    virtual void pullInputs(size_t framesToProcess) OVERRIDE FINAL;
+    virtual void connect(AudioNode*, unsigned outputIndex, unsigned inputIndex, ExceptionState&) OVERRIDE FINAL;
+    virtual void disconnect(unsigned outputIndex, ExceptionState&) OVERRIDE FINAL;
+    virtual void checkNumberOfChannelsForInput(AudioNodeInput*) OVERRIDE FINAL;
 
 private:
     void updatePullStatus();
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioBasicProcessorNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/AudioBasicProcessorNode.cpp
index 6428bae8899..8a5330e2f64 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioBasicProcessorNode.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioBasicProcessorNode.cpp
@@ -91,12 +91,6 @@ void AudioBasicProcessorNode::pullInputs(size_t framesToProcess)
     input(0)->pull(output(0)->bus(), framesToProcess);
 }
 
-void AudioBasicProcessorNode::reset()
-{
-    if (processor())
-        processor()->reset();
-}
-
 // As soon as we know the channel count of our input, we can lazily initialize.
 // Sometimes this may be called more than once with different channel counts, in which case we must safely
 // uninitialize and then re-initialize with the new channel count.
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioBasicProcessorNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/AudioBasicProcessorNode.h
index eb94cac2221..ad37fb6e3bd 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioBasicProcessorNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioBasicProcessorNode.h
@@ -42,21 +42,20 @@ public:
     AudioBasicProcessorNode(AudioContext*, float sampleRate);
 
     // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void pullInputs(size_t framesToProcess);
-    virtual void reset();
-    virtual void initialize();
-    virtual void uninitialize();
+    virtual void process(size_t framesToProcess) OVERRIDE FINAL;
+    virtual void pullInputs(size_t framesToProcess) OVERRIDE FINAL;
+    virtual void initialize() OVERRIDE FINAL;
+    virtual void uninitialize() OVERRIDE FINAL;
 
     // Called in the main thread when the number of channels for the input may have changed.
-    virtual void checkNumberOfChannelsForInput(AudioNodeInput*);
+    virtual void checkNumberOfChannelsForInput(AudioNodeInput*) OVERRIDE FINAL;
 
     // Returns the number of channels for both the input and the output.
     unsigned numberOfChannels();
 
 protected:
-    virtual double tailTime() const OVERRIDE;
-    virtual double latencyTime() const OVERRIDE;
+    virtual double tailTime() const OVERRIDE FINAL;
+    virtual double latencyTime() const OVERRIDE FINAL;
 
     AudioProcessor* processor() { return m_processor.get(); }
     OwnPtr<AudioProcessor> m_processor;
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioBuffer.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/AudioBuffer.cpp
index b520512e054..d473aae6cc4 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioBuffer.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioBuffer.cpp
@@ -32,6 +32,7 @@
 
 #include "modules/webaudio/AudioBuffer.h"
 
+#include "bindings/v8/ExceptionMessages.h"
 #include "bindings/v8/ExceptionState.h"
 #include "core/dom/ExceptionCode.h"
 #include "platform/audio/AudioBus.h"
@@ -42,36 +43,101 @@ namespace WebCore {
 
 float AudioBuffer::minAllowedSampleRate()
 {
-    return 22050;
+    // crbug.com/344375
+    return 3000;
 }
 
 float AudioBuffer::maxAllowedSampleRate()
 {
-    return 96000;
+    // Windows can support up to this rate.
+    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() || !numberOfFrames)
-        return 0;
+    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 0;
+        return nullptr;
     return buffer;
 }
 
-PassRefPtr<AudioBuffer> AudioBuffer::createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, float sampleRate)
+PassRefPtrWillBeRawPtr<AudioBuffer> AudioBuffer::create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState)
+{
+    if (!numberOfChannels || numberOfChannels > AudioContext::maxNumberOfChannels()) {
+        exceptionState.throwDOMException(
+            NotSupportedError,
+            ExceptionMessages::indexOutsideRange(
+                "number of channels",
+                numberOfChannels,
+                1u,
+                ExceptionMessages::InclusiveBound,
+                AudioContext::maxNumberOfChannels(),
+                ExceptionMessages::InclusiveBound));
+        return nullptr;
+    }
+
+    if (sampleRate < AudioBuffer::minAllowedSampleRate() || sampleRate > AudioBuffer::maxAllowedSampleRate()) {
+        exceptionState.throwDOMException(
+            NotSupportedError,
+            ExceptionMessages::indexOutsideRange(
+                "sample rate",
+                sampleRate,
+                AudioBuffer::minAllowedSampleRate(),
+                ExceptionMessages::InclusiveBound,
+                AudioBuffer::maxAllowedSampleRate(),
+                ExceptionMessages::InclusiveBound));
+        return nullptr;
+    }
+
+    if (!numberOfFrames) {
+        exceptionState.throwDOMException(
+            NotSupportedError,
+            ExceptionMessages::indexExceedsMinimumBound(
+                "number of frames",
+                numberOfFrames,
+                static_cast<size_t>(0)));
+        return nullptr;
+    }
+
+    RefPtrWillBeRawPtr<AudioBuffer> audioBuffer = create(numberOfChannels, numberOfFrames, sampleRate);
+
+    if (!audioBuffer.get()) {
+        exceptionState.throwDOMException(
+            NotSupportedError,
+            "createBuffer("
+            + String::number(numberOfChannels) + ", "
+            + String::number(numberOfFrames) + ", "
+            + String::number(sampleRate)
+            + ") failed.");
+    }
+
+    return audioBuffer;
+}
+
+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;
     }
 
-    return 0;
+    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
@@ -80,8 +146,7 @@ bool AudioBuffer::createdSuccessfully(unsigned desiredNumberOfChannels) const
 }
 
 AudioBuffer::AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate)
-    : m_gain(1.0)
-    , m_sampleRate(sampleRate)
+    : m_sampleRate(sampleRate)
     , m_length(numberOfFrames)
 {
     ScriptWrappable::init(this);
@@ -101,8 +166,7 @@ AudioBuffer::AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, float
 }
 
 AudioBuffer::AudioBuffer(AudioBus* bus)
-    : m_gain(1.0)
-    , m_sampleRate(bus->sampleRate())
+    : m_sampleRate(bus->sampleRate())
     , m_length(bus->length())
 {
     ScriptWrappable::init(this);
@@ -122,16 +186,11 @@ AudioBuffer::AudioBuffer(AudioBus* bus)
     }
 }
 
-void AudioBuffer::releaseMemory()
-{
-    m_channels.clear();
-}
-
 PassRefPtr<Float32Array> AudioBuffer::getChannelData(unsigned channelIndex, ExceptionState& exceptionState)
 {
     if (channelIndex >= m_channels.size()) {
         exceptionState.throwDOMException(IndexSizeError, "channel index (" + String::number(channelIndex) + ") exceeds number of channels (" + String::number(m_channels.size()) + ")");
-        return 0;
+        return nullptr;
     }
 
     Float32Array* channelData = m_channels[channelIndex].get();
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioBuffer.h b/chromium/third_party/WebKit/Source/modules/webaudio/AudioBuffer.h
index 8ffb2aceb11..cd0f87958bb 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioBuffer.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioBuffer.h
@@ -41,16 +41,19 @@ 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 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; }
-    double duration() const { return length() / sampleRate(); }
+    double duration() const { return length() / static_cast<double>(sampleRate()); }
     float sampleRate() const { return m_sampleRate; }
 
     // Channel data access
@@ -59,23 +62,16 @@ public:
     Float32Array* getChannelData(unsigned channelIndex);
     void zero();
 
-    // Scalar gain
-    double gain() const { return m_gain; }
-    void setGain(double gain) { m_gain = gain; }
-
-    // Because an AudioBuffer has a JavaScript wrapper, which will be garbage collected, it may take awhile for this object to be deleted.
-    // releaseMemory() can be called when the AudioContext goes away, so we can release the memory earlier than when the garbage collection happens.
-    // Careful! Only call this when the page unloads, after the AudioContext is no longer processing.
-    void releaseMemory();
-
     static float minAllowedSampleRate();
     static float maxAllowedSampleRate();
+
+    void trace(Visitor*) { }
+
 protected:
     AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
     explicit AudioBuffer(AudioBus*);
     bool createdSuccessfully(unsigned desiredNumberOfChannels) const;
 
-    double m_gain; // scalar gain
     float m_sampleRate;
     size_t m_length;
 
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioBuffer.idl b/chromium/third_party/WebKit/Source/modules/webaudio/AudioBuffer.idl
index 19f66a5f2a0..7ec7c60bb4c 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioBuffer.idl
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioBuffer.idl
@@ -27,14 +27,13 @@
  */
 
 [
+    WillBeGarbageCollected,
     Conditional=WEB_AUDIO
 ] interface AudioBuffer {
     readonly attribute long length; // in sample-frames
-    readonly attribute float duration; // in seconds
+    readonly attribute double duration; // in seconds
     readonly attribute float sampleRate; // in sample-frames per second
 
-    attribute float gain; // linear gain (default 1.0)
-
     // Channel access
     readonly attribute unsigned long numberOfChannels;
     [RaisesException] Float32Array getChannelData(unsigned long channelIndex);
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioBufferSourceNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/AudioBufferSourceNode.cpp
index 07a63a57bc1..c2dbc32bf24 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioBufferSourceNode.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioBufferSourceNode.cpp
@@ -30,7 +30,6 @@
 
 #include "bindings/v8/ExceptionState.h"
 #include "core/dom/ExceptionCode.h"
-#include "core/page/PageConsole.h"
 #include "platform/audio/AudioUtilities.h"
 #include "modules/webaudio/AudioContext.h"
 #include "modules/webaudio/AudioNodeOutput.h"
@@ -50,14 +49,14 @@ 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)
     : AudioScheduledSourceNode(context, sampleRate)
-    , m_buffer(0)
+    , m_buffer(nullptr)
     , m_isLooping(false)
     , m_loopStart(0)
     , m_loopEnd(0)
@@ -65,13 +64,11 @@ AudioBufferSourceNode::AudioBufferSourceNode(AudioContext* context, float sample
     , m_isGrain(false)
     , m_grainOffset(0.0)
     , m_grainDuration(DefaultGrainDuration)
-    , m_lastGain(1.0)
     , m_pannerNode(0)
 {
     ScriptWrappable::init(this);
     setNodeType(NodeTypeAudioBufferSource);
 
-    m_gain = AudioParam::create(context, "gain", 1.0, 0.0, 1.0);
     m_playbackRate = AudioParam::create(context, "playbackRate", 1.0, 0.0, MaxRate);
 
     // Default to mono.  A call to setBuffer() will set the number of output channels to that of the buffer.
@@ -133,9 +130,6 @@ void AudioBufferSourceNode::process(size_t framesToProcess)
             return;
         }
 
-        // Apply the gain (in-place) to the output bus.
-        float totalGain = gain()->value() * m_buffer->gain();
-        outputBus->copyWithGainFrom(*outputBus, &m_lastGain, totalGain);
         outputBus->clearSilentFlag();
     } else {
         // Too bad - the tryLock() failed.  We must be in the middle of changing buffers and were already outputting silence anyway.
@@ -331,21 +325,9 @@ bool AudioBufferSourceNode::renderFromBuffer(AudioBus* bus, unsigned destination
 }
 
 
-void AudioBufferSourceNode::reset()
-{
-    m_virtualReadIndex = 0;
-    m_lastGain = gain()->value();
-}
-
 void AudioBufferSourceNode::setBuffer(AudioBuffer* buffer, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
-    // FIXME: It does not look like we should throw if the buffer is null as
-    // the attribute is nullable in the specification.
-    if (!buffer) {
-        exceptionState.throwTypeError("buffer cannot be null");
-        return;
-    }
 
     // The context must be locked since changing the buffer can re-configure the number of channels that are output.
     AudioContext::AutoLocker contextLocker(context());
@@ -382,28 +364,18 @@ unsigned AudioBufferSourceNode::numberOfChannels()
     return output(0)->numberOfChannels();
 }
 
-void AudioBufferSourceNode::start(ExceptionState& exceptionState)
-{
-    startPlaying(false, 0, 0, buffer() ? buffer()->duration() : 0, exceptionState);
-}
-
 void AudioBufferSourceNode::start(double when, ExceptionState& exceptionState)
 {
-    startPlaying(false, when, 0, buffer() ? buffer()->duration() : 0, exceptionState);
+    AudioScheduledSourceNode::start(when, exceptionState);
 }
 
 void AudioBufferSourceNode::start(double when, double grainOffset, ExceptionState& exceptionState)
 {
-    startPlaying(true, when, grainOffset, buffer() ? buffer()->duration() : 0, exceptionState);
+    start(when, grainOffset, buffer() ? buffer()->duration() : 0, exceptionState);
 }
 
 void AudioBufferSourceNode::start(double when, double grainOffset, double grainDuration, ExceptionState& exceptionState)
 {
-    startPlaying(true, when, grainOffset, grainDuration, exceptionState);
-}
-
-void AudioBufferSourceNode::startPlaying(bool isGrain, double when, double grainOffset, double grainDuration, ExceptionState& exceptionState)
-{
     ASSERT(isMainThread());
 
     if (m_playbackState != UNSCHEDULED_STATE) {
@@ -416,22 +388,20 @@ void AudioBufferSourceNode::startPlaying(bool isGrain, double when, double grain
     if (!buffer())
         return;
 
-    if (isGrain) {
-        // Do sanity checking of grain parameters versus buffer size.
-        double bufferDuration = buffer()->duration();
+    // Do sanity checking of grain parameters versus buffer size.
+    double bufferDuration = buffer()->duration();
 
-        grainOffset = max(0.0, grainOffset);
-        grainOffset = min(bufferDuration, grainOffset);
-        m_grainOffset = grainOffset;
+    grainOffset = max(0.0, grainOffset);
+    grainOffset = min(bufferDuration, grainOffset);
+    m_grainOffset = grainOffset;
 
-        double maxDuration = bufferDuration - grainOffset;
+    double maxDuration = bufferDuration - grainOffset;
 
-        grainDuration = max(0.0, grainDuration);
-        grainDuration = min(maxDuration, grainDuration);
-        m_grainDuration = grainDuration;
-    }
+    grainDuration = max(0.0, grainDuration);
+    grainDuration = min(maxDuration, grainDuration);
+    m_grainDuration = grainDuration;
 
-    m_isGrain = isGrain;
+    m_isGrain = true;
     m_startTime = when;
 
     // We call timeToSampleFrame here since at playbackRate == 1 we don't want to go through linear interpolation
@@ -443,14 +413,6 @@ void AudioBufferSourceNode::startPlaying(bool isGrain, double when, double grain
     m_playbackState = SCHEDULED_STATE;
 }
 
-void AudioBufferSourceNode::noteGrainOn(double when, double grainOffset, double grainDuration, ExceptionState& exceptionState)
-{
-    // Handle unspecified duration where 0 means the rest of the buffer.
-    if (!grainDuration && buffer())
-        grainDuration = buffer()->duration();
-    startPlaying(true, when, grainOffset, grainDuration, exceptionState);
-}
-
 double AudioBufferSourceNode::totalPitchRate()
 {
     double dopplerRate = 1.0;
@@ -513,6 +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)
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioBufferSourceNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/AudioBufferSourceNode.h
index 9eafdb201b4..cb24a38c09a 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioBufferSourceNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioBufferSourceNode.h
@@ -42,15 +42,14 @@ class AudioContext;
 // AudioBufferSourceNode is an AudioNode representing an audio source from an in-memory audio asset represented by an AudioBuffer.
 // It generally will be used for short sounds which require a high degree of scheduling flexibility (can playback in rhythmically perfect ways).
 
-class AudioBufferSourceNode : public AudioScheduledSourceNode {
+class AudioBufferSourceNode FINAL : public AudioScheduledSourceNode {
 public:
-    static PassRefPtr<AudioBufferSourceNode> create(AudioContext*, float sampleRate);
+    static PassRefPtrWillBeRawPtr<AudioBufferSourceNode> create(AudioContext*, float sampleRate);
 
     virtual ~AudioBufferSourceNode();
 
     // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void reset();
+    virtual void process(size_t framesToProcess) OVERRIDE;
 
     // setBuffer() is called on the main thread. This is the buffer we use for playback.
     void setBuffer(AudioBuffer*, ExceptionState&);
@@ -61,13 +60,11 @@ public:
     unsigned numberOfChannels();
 
     // Play-state
-    void start(ExceptionState&);
+    void start(ExceptionState& exceptionState) { start(0, exceptionState); }
     void start(double when, ExceptionState&);
     void start(double when, double grainOffset, ExceptionState&);
     void start(double when, double grainOffset, double grainDuration, ExceptionState&);
 
-    void noteGrainOn(double when, double grainOffset, double grainDuration, ExceptionState&);
-
     // Note: the attribute was originally exposed as .looping, but to be more consistent in naming with <audio>
     // and with how it's described in the specification, the proper attribute name is .loop
     // The old attribute is kept for backwards compatibility.
@@ -80,7 +77,6 @@ public:
     void setLoopStart(double loopStart) { m_loopStart = loopStart; }
     void setLoopEnd(double loopEnd) { m_loopEnd = loopEnd; }
 
-    AudioParam* gain() { return m_gain.get(); }
     AudioParam* playbackRate() { return m_playbackRate.get(); }
 
     // If a panner node is set, then we can incorporate doppler shift into the playback pitch rate.
@@ -88,16 +84,16 @@ public:
     void clearPannerNode();
 
     // If we are no longer playing, propogate silence ahead to downstream nodes.
-    virtual bool propagatesSilence() const;
+    virtual bool propagatesSilence() const OVERRIDE;
 
     // AudioScheduledSourceNode
     virtual void finish() OVERRIDE;
 
+    virtual void trace(Visitor*) OVERRIDE;
+
 private:
     AudioBufferSourceNode(AudioContext*, float sampleRate);
 
-    void startPlaying(bool isGrain, double when, double grainOffset, double grainDuration, ExceptionState&);
-
     // Returns true on success.
     bool renderFromBuffer(AudioBus*, unsigned destinationFrameOffset, size_t numberOfFrames);
 
@@ -105,15 +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 "gain" and "playbackRate" attributes.
-    RefPtr<AudioParam> m_gain;
-    RefPtr<AudioParam> m_playbackRate;
+    // Used for the "playbackRate" attributes.
+    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.
@@ -135,9 +130,6 @@ private:
     // It incorporates the base pitch rate, any sample-rate conversion factor from the buffer, and any doppler shift from an associated panner node.
     double totalPitchRate();
 
-    // m_lastGain provides continuity when we dynamically adjust the gain.
-    float m_lastGain;
-
     // We optionally keep track of a panner node which has a doppler shift that is incorporated into
     // the pitch rate. We manually manage ref-counting because we want to use RefTypeConnection.
     PannerNode* m_pannerNode;
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioBufferSourceNode.idl b/chromium/third_party/WebKit/Source/modules/webaudio/AudioBufferSourceNode.idl
index ab58774ead6..135f1ce1be2 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioBufferSourceNode.idl
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioBufferSourceNode.idl
@@ -24,33 +24,19 @@
 
 // A cached (non-streamed), memory-resident audio source
 [
-    Conditional=WEB_AUDIO
+    Conditional=WEB_AUDIO,
+    TypeChecking=Interface|Nullable,
 ] interface AudioBufferSourceNode : AudioSourceNode {
-    [StrictTypeChecking, RaisesException=Setter] attribute AudioBuffer buffer;
+    [RaisesException=Setter] attribute AudioBuffer buffer;
 
-    const unsigned short UNSCHEDULED_STATE = 0;
-    const unsigned short SCHEDULED_STATE = 1;
-    const unsigned short PLAYING_STATE = 2;
-    const unsigned short FINISHED_STATE = 3;
-
-    readonly attribute unsigned short playbackState;
-
-    readonly attribute AudioParam gain;
     readonly attribute AudioParam playbackRate;
 
-    attribute boolean loop; // This is the proper attribute name from the specification.
-
+    attribute boolean loop;
     attribute double loopStart;
     attribute double loopEnd;
 
-    [RaisesException, MeasureAs=WebAudioStart] void start(optional double when, optional double grainOffset, optional double grainDuration);
+    [RaisesException] void start(optional double when, optional double grainOffset, optional double grainDuration);
     [RaisesException] void stop(optional double when);
 
-    [DeprecateAs=WebAudioLooping, ImplementedAs=loop] attribute boolean looping; // This is an alias for the .loop attribute for backwards compatibility.
-
-    [RaisesException, ImplementedAs=start, MeasureAs=LegacyWebAudio] void noteOn(double when);
-    [RaisesException, MeasureAs=LegacyWebAudio] void noteGrainOn(double when, double grainOffset, double grainDuration);
-    [RaisesException, ImplementedAs=stop] void noteOff(double when);
-
     attribute EventHandler onended;
 };
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioContext.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/AudioContext.cpp
index 5ae5ffe99f1..a77c9988d43 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioContext.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioContext.cpp
@@ -86,91 +86,67 @@ bool AudioContext::isSampleRateRangeGood(float sampleRate)
 }
 
 // Don't allow more than this number of simultaneous AudioContexts talking to hardware.
-const unsigned MaxHardwareContexts = 4;
+const unsigned MaxHardwareContexts = 6;
 unsigned AudioContext::s_hardwareContextCount = 0;
 
-PassRefPtr<AudioContext> AudioContext::create(Document& document, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<AudioContext> AudioContext::create(Document& document, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
     if (s_hardwareContextCount >= MaxHardwareContexts) {
         exceptionState.throwDOMException(
             SyntaxError,
             "number of hardware contexts reached maximum (" + String::number(MaxHardwareContexts) + ").");
-        return 0;
+        return nullptr;
     }
 
-    RefPtr<AudioContext> audioContext(adoptRef(new AudioContext(&document)));
+    RefPtrWillBeRawPtr<AudioContext> audioContext(adoptRefWillBeThreadSafeRefCountedGarbageCollected(new AudioContext(&document)));
     audioContext->suspendIfNeeded();
     return audioContext.release();
 }
 
-PassRefPtr<AudioContext> AudioContext::create(Document& document, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState)
-{
-    document.addConsoleMessage(JSMessageSource, WarningMessageLevel, "Deprecated AudioContext constructor: use OfflineAudioContext instead");
-    return OfflineAudioContext::create(&document, numberOfChannels, numberOfFrames, sampleRate, exceptionState);
-}
-
 // Constructor for rendering to the audio hardware.
 AudioContext::AudioContext(Document* document)
     : ActiveDOMObject(document)
     , m_isStopScheduled(false)
+    , m_isCleared(false)
     , m_isInitialized(false)
-    , m_isAudioThreadFinished(false)
-    , m_destinationNode(0)
+    , m_destinationNode(nullptr)
     , m_isDeletionScheduled(false)
     , m_automaticPullNodesNeedUpdating(false)
     , m_connectionCount(0)
     , m_audioThread(0)
     , m_graphOwnerThread(UndefinedThreadIdentifier)
     , m_isOfflineContext(false)
-    , m_activeSourceCount(0)
 {
-    constructCommon();
+    ScriptWrappable::init(this);
 
     m_destinationNode = DefaultAudioDestinationNode::create(this);
 
-    // This sets in motion an asynchronous loading mechanism on another thread.
-    // We can check m_hrtfDatabaseLoader->isLoaded() to find out whether or not it has been fully loaded.
-    // It's not that useful to have a callback function for this since the audio thread automatically starts rendering on the graph
-    // when this has finished (see AudioDestinationNode).
-    m_hrtfDatabaseLoader = HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(sampleRate());
+    initialize();
 }
 
 // Constructor for offline (non-realtime) rendering.
 AudioContext::AudioContext(Document* document, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate)
     : ActiveDOMObject(document)
     , m_isStopScheduled(false)
+    , m_isCleared(false)
     , m_isInitialized(false)
-    , m_isAudioThreadFinished(false)
-    , m_destinationNode(0)
+    , m_destinationNode(nullptr)
+    , m_isDeletionScheduled(false)
     , m_automaticPullNodesNeedUpdating(false)
     , m_connectionCount(0)
     , m_audioThread(0)
     , m_graphOwnerThread(UndefinedThreadIdentifier)
     , m_isOfflineContext(true)
-    , m_activeSourceCount(0)
 {
-    constructCommon();
-
-    m_hrtfDatabaseLoader = HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(sampleRate);
+    ScriptWrappable::init(this);
 
     // Create a new destination for offline rendering.
     m_renderTarget = AudioBuffer::create(numberOfChannels, numberOfFrames, sampleRate);
-    ASSERT(m_renderTarget);
-    m_destinationNode = OfflineAudioDestinationNode::create(this, m_renderTarget.get());
-    ASSERT(m_destinationNode);
-}
+    if (m_renderTarget.get())
+        m_destinationNode = OfflineAudioDestinationNode::create(this, m_renderTarget.get());
 
-void AudioContext::constructCommon()
-{
-    ScriptWrappable::init(this);
-    // According to spec AudioContext must die only after page navigate.
-    // Lets mark it as ActiveDOMObject with pending activity and unmark it in clear method.
-    setPendingActivity(this);
-
-    FFTFrame::initialize();
-
-    m_listener = AudioListener::create();
+    initialize();
 }
 
 AudioContext::~AudioContext()
@@ -180,7 +156,6 @@ AudioContext::~AudioContext()
 #endif
     // AudioNodes keep a reference to their context, so there should be no way to be in the destructor if there are still AudioNodes around.
     ASSERT(!m_isInitialized);
-    ASSERT(m_isStopScheduled);
     ASSERT(!m_nodesToDelete.size());
     ASSERT(!m_referencedNodes.size());
     ASSERT(!m_finishedNodes.size());
@@ -190,27 +165,27 @@ AudioContext::~AudioContext()
     ASSERT(!m_renderingAutomaticPullNodes.size());
 }
 
-void AudioContext::lazyInitialize()
+void AudioContext::initialize()
 {
-    if (!m_isInitialized) {
-        // Don't allow the context to initialize a second time after it's already been explicitly uninitialized.
-        ASSERT(!m_isAudioThreadFinished);
-        if (!m_isAudioThreadFinished) {
-            if (m_destinationNode.get()) {
-                m_destinationNode->initialize();
+    if (isInitialized())
+        return;
+
+    FFTFrame::initialize();
+    m_listener = AudioListener::create();
 
-                if (!isOfflineContext()) {
-                    // This starts the audio thread. The destination node's provideInput() method will now be called repeatedly to render audio.
-                    // Each time provideInput() is called, a portion of the audio stream is rendered. Let's call this time period a "render quantum".
-                    // NOTE: for now default AudioContext does not need an explicit startRendering() call from JavaScript.
-                    // We may want to consider requiring it for symmetry with OfflineAudioContext.
-                    m_destinationNode->startRendering();
-                    ++s_hardwareContextCount;
-                }
+    if (m_destinationNode.get()) {
+        m_destinationNode->initialize();
 
-            }
-            m_isInitialized = true;
+        if (!isOfflineContext()) {
+            // This starts the audio thread. The destination node's provideInput() method will now be called repeatedly to render audio.
+            // Each time provideInput() is called, a portion of the audio stream is rendered. Let's call this time period a "render quantum".
+            // NOTE: for now default AudioContext does not need an explicit startRendering() call from JavaScript.
+            // We may want to consider requiring it for symmetry with OfflineAudioContext.
+            m_destinationNode->startRendering();
+            ++s_hardwareContextCount;
         }
+
+        m_isInitialized = true;
     }
 }
 
@@ -222,28 +197,24 @@ void AudioContext::clear()
 
     // Audio thread is dead. Nobody will schedule node deletion action. Let's do it ourselves.
     do {
-        deleteMarkedNodes();
-        m_nodesToDelete.append(m_nodesMarkedForDeletion);
+        m_nodesToDelete.appendVector(m_nodesMarkedForDeletion);
         m_nodesMarkedForDeletion.clear();
+        deleteMarkedNodes();
     } while (m_nodesToDelete.size());
 
-    // It was set in constructCommon.
-    unsetPendingActivity(this);
+    m_isCleared = true;
 }
 
 void AudioContext::uninitialize()
 {
     ASSERT(isMainThread());
 
-    if (!m_isInitialized)
+    if (!isInitialized())
         return;
 
     // This stops the audio thread and all audio rendering.
     m_destinationNode->uninitialize();
 
-    // Don't allow the context to initialize a second time after it's already been explicitly uninitialized.
-    m_isAudioThreadFinished = true;
-
     if (!isOfflineContext()) {
         ASSERT(s_hardwareContextCount);
         --s_hardwareContextCount;
@@ -255,20 +226,6 @@ void AudioContext::uninitialize()
     m_isInitialized = false;
 }
 
-bool AudioContext::isInitialized() const
-{
-    return m_isInitialized;
-}
-
-bool AudioContext::isRunnable() const
-{
-    if (!isInitialized())
-        return false;
-
-    // Check with the HRTF spatialization system to see if it's finished loading.
-    return m_hrtfDatabaseLoader->isLoaded();
-}
-
 void AudioContext::stopDispatch(void* userData)
 {
     AudioContext* context = reinterpret_cast<AudioContext*>(userData);
@@ -294,55 +251,15 @@ void AudioContext::stop()
     callOnMainThread(stopDispatch, this);
 }
 
-PassRefPtr<AudioBuffer> AudioContext::createBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState)
+bool AudioContext::hasPendingActivity() const
 {
-    RefPtr<AudioBuffer> audioBuffer = AudioBuffer::create(numberOfChannels, numberOfFrames, sampleRate);
-    if (!audioBuffer.get()) {
-        if (numberOfChannels > AudioContext::maxNumberOfChannels()) {
-            exceptionState.throwDOMException(
-                NotSupportedError,
-                "requested number of channels (" + String::number(numberOfChannels) + ") exceeds maximum (" + String::number(AudioContext::maxNumberOfChannels()) + ")");
-        } else if (sampleRate < AudioBuffer::minAllowedSampleRate() || sampleRate > AudioBuffer::maxAllowedSampleRate()) {
-            exceptionState.throwDOMException(
-                NotSupportedError,
-                "requested sample rate (" + String::number(sampleRate)
-                + ") does not lie in the allowed range of "
-                + String::number(AudioBuffer::minAllowedSampleRate())
-                + "-" + String::number(AudioBuffer::maxAllowedSampleRate()) + " Hz");
-        } else if (!numberOfFrames) {
-            exceptionState.throwDOMException(
-                NotSupportedError,
-                "number of frames must be greater than 0.");
-        } else {
-            exceptionState.throwDOMException(
-                NotSupportedError,
-                "unable to create buffer of " + String::number(numberOfChannels)
-                + " channel(s) of " + String::number(numberOfFrames)
-                + " frames each.");
-        }
-        return 0;
-    }
-
-    return audioBuffer;
+    // According to spec AudioContext must die only after page navigates.
+    return !m_isCleared;
 }
 
-PassRefPtr<AudioBuffer> AudioContext::createBuffer(ArrayBuffer* arrayBuffer, bool mixToMono, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<AudioBuffer> AudioContext::createBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState)
 {
-    ASSERT(arrayBuffer);
-    if (!arrayBuffer) {
-        exceptionState.throwDOMException(
-            SyntaxError,
-            "invalid ArrayBuffer.");
-        return 0;
-    }
-
-    RefPtr<AudioBuffer> audioBuffer = AudioBuffer::createFromAudioFileData(arrayBuffer->data(), arrayBuffer->byteLength(), mixToMono, sampleRate());
-    if (!audioBuffer.get()) {
-        exceptionState.throwDOMException(
-            SyntaxError,
-            "invalid audio data in ArrayBuffer.");
-        return 0;
-    }
+    RefPtrWillBeRawPtr<AudioBuffer> audioBuffer = AudioBuffer::create(numberOfChannels, numberOfFrames, sampleRate, exceptionState);
 
     return audioBuffer;
 }
@@ -358,11 +275,10 @@ void AudioContext::decodeAudioData(ArrayBuffer* audioData, PassOwnPtr<AudioBuffe
     m_audioDecoder.decodeAsync(audioData, sampleRate(), successCallback, errorCallback);
 }
 
-PassRefPtr<AudioBufferSourceNode> AudioContext::createBufferSource()
+PassRefPtrWillBeRawPtr<AudioBufferSourceNode> AudioContext::createBufferSource()
 {
     ASSERT(isMainThread());
-    lazyInitialize();
-    RefPtr<AudioBufferSourceNode> node = AudioBufferSourceNode::create(this, m_destinationNode->sampleRate());
+    RefPtrWillBeRawPtr<AudioBufferSourceNode> node = AudioBufferSourceNode::create(this, m_destinationNode->sampleRate());
 
     // Because this is an AudioScheduledSourceNode, the context keeps a reference until it has finished playing.
     // When this happens, AudioScheduledSourceNode::finish() calls AudioContext::notifyNodeFinishedProcessing().
@@ -371,27 +287,25 @@ PassRefPtr<AudioBufferSourceNode> AudioContext::createBufferSource()
     return node;
 }
 
-PassRefPtr<MediaElementAudioSourceNode> AudioContext::createMediaElementSource(HTMLMediaElement* mediaElement, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<MediaElementAudioSourceNode> AudioContext::createMediaElementSource(HTMLMediaElement* mediaElement, ExceptionState& exceptionState)
 {
+    ASSERT(isMainThread());
     if (!mediaElement) {
         exceptionState.throwDOMException(
             InvalidStateError,
             "invalid HTMLMedialElement.");
-        return 0;
+        return nullptr;
     }
 
-    ASSERT(isMainThread());
-    lazyInitialize();
-
     // First check if this media element already has a source node.
     if (mediaElement->audioSourceNode()) {
         exceptionState.throwDOMException(
             InvalidStateError,
             "invalid HTMLMediaElement.");
-        return 0;
+        return nullptr;
     }
 
-    RefPtr<MediaElementAudioSourceNode> node = MediaElementAudioSourceNode::create(this, mediaElement);
+    RefPtrWillBeRawPtr<MediaElementAudioSourceNode> node = MediaElementAudioSourceNode::create(this, mediaElement);
 
     mediaElement->setAudioSourceNode(node.get());
 
@@ -399,33 +313,28 @@ PassRefPtr<MediaElementAudioSourceNode> AudioContext::createMediaElementSource(H
     return node;
 }
 
-PassRefPtr<MediaStreamAudioSourceNode> AudioContext::createMediaStreamSource(MediaStream* mediaStream, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<MediaStreamAudioSourceNode> AudioContext::createMediaStreamSource(MediaStream* mediaStream, ExceptionState& exceptionState)
 {
+    ASSERT(isMainThread());
     if (!mediaStream) {
         exceptionState.throwDOMException(
             InvalidStateError,
             "invalid MediaStream source");
-        return 0;
+        return nullptr;
     }
 
-    ASSERT(isMainThread());
-    lazyInitialize();
-
-    AudioSourceProvider* provider = 0;
-
     MediaStreamTrackVector audioTracks = mediaStream->getAudioTracks();
-    RefPtr<MediaStreamTrack> audioTrack;
-
-    // FIXME: get a provider for non-local MediaStreams (like from a remote peer).
-    for (size_t i = 0; i < audioTracks.size(); ++i) {
-        audioTrack = audioTracks[i];
-        if (audioTrack->component()->audioSourceProvider()) {
-            provider = audioTrack->component()->audioSourceProvider();
-            break;
-        }
+    if (audioTracks.isEmpty()) {
+        exceptionState.throwDOMException(
+            InvalidStateError,
+            "MediaStream has no audio track");
+        return nullptr;
     }
 
-    RefPtr<MediaStreamAudioSourceNode> node = MediaStreamAudioSourceNode::create(this, mediaStream, audioTrack.get(), provider);
+    // Use the first audio track in the media stream.
+    RefPtrWillBeRawPtr<MediaStreamTrack> audioTrack = audioTracks[0];
+    OwnPtr<AudioSourceProvider> provider = audioTrack->createWebAudioSource();
+    RefPtrWillBeRawPtr<MediaStreamAudioSourceNode> node = MediaStreamAudioSourceNode::create(this, mediaStream, audioTrack.get(), provider.release());
 
     // FIXME: Only stereo streams are supported right now. We should be able to accept multi-channel streams.
     node->setFormat(2, sampleRate());
@@ -434,36 +343,34 @@ PassRefPtr<MediaStreamAudioSourceNode> AudioContext::createMediaStreamSource(Med
     return node;
 }
 
-PassRefPtr<MediaStreamAudioDestinationNode> AudioContext::createMediaStreamDestination()
+PassRefPtrWillBeRawPtr<MediaStreamAudioDestinationNode> AudioContext::createMediaStreamDestination()
 {
-    // FIXME: Add support for an optional argument which specifies the number of channels.
-    // FIXME: The default should probably be stereo instead of mono.
-    return MediaStreamAudioDestinationNode::create(this, 1);
+    // Set number of output channels to stereo by default.
+    return MediaStreamAudioDestinationNode::create(this, 2);
 }
 
-PassRefPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(ExceptionState& exceptionState)
 {
     // Set number of input/output channels to stereo by default.
     return createScriptProcessor(0, 2, 2, exceptionState);
 }
 
-PassRefPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t bufferSize, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t bufferSize, ExceptionState& exceptionState)
 {
     // Set number of input/output channels to stereo by default.
     return createScriptProcessor(bufferSize, 2, 2, exceptionState);
 }
 
-PassRefPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t bufferSize, size_t numberOfInputChannels, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t bufferSize, size_t numberOfInputChannels, ExceptionState& exceptionState)
 {
     // Set number of output channels to stereo by default.
     return createScriptProcessor(bufferSize, numberOfInputChannels, 2, exceptionState);
 }
 
-PassRefPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t bufferSize, size_t numberOfInputChannels, size_t numberOfOutputChannels, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t bufferSize, size_t numberOfInputChannels, size_t numberOfOutputChannels, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
-    lazyInitialize();
-    RefPtr<ScriptProcessorNode> node = ScriptProcessorNode::create(this, m_destinationNode->sampleRate(), bufferSize, numberOfInputChannels, numberOfOutputChannels);
+    RefPtrWillBeRawPtr<ScriptProcessorNode> node = ScriptProcessorNode::create(this, m_destinationNode->sampleRate(), bufferSize, numberOfInputChannels, numberOfOutputChannels);
 
     if (!node.get()) {
         if (!numberOfInputChannels && !numberOfOutputChannels) {
@@ -488,90 +395,81 @@ PassRefPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t buffe
                 "buffer size (" + String::number(bufferSize)
                 + ") must be a power of two between 256 and 16384.");
         }
-        return 0;
+        return nullptr;
     }
 
     refNode(node.get()); // context keeps reference until we stop making javascript rendering callbacks
     return node;
 }
 
-PassRefPtr<BiquadFilterNode> AudioContext::createBiquadFilter()
+PassRefPtrWillBeRawPtr<BiquadFilterNode> AudioContext::createBiquadFilter()
 {
     ASSERT(isMainThread());
-    lazyInitialize();
     return BiquadFilterNode::create(this, m_destinationNode->sampleRate());
 }
 
-PassRefPtr<WaveShaperNode> AudioContext::createWaveShaper()
+PassRefPtrWillBeRawPtr<WaveShaperNode> AudioContext::createWaveShaper()
 {
     ASSERT(isMainThread());
-    lazyInitialize();
     return WaveShaperNode::create(this);
 }
 
-PassRefPtr<PannerNode> AudioContext::createPanner()
+PassRefPtrWillBeRawPtr<PannerNode> AudioContext::createPanner()
 {
     ASSERT(isMainThread());
-    lazyInitialize();
     return PannerNode::create(this, m_destinationNode->sampleRate());
 }
 
-PassRefPtr<ConvolverNode> AudioContext::createConvolver()
+PassRefPtrWillBeRawPtr<ConvolverNode> AudioContext::createConvolver()
 {
     ASSERT(isMainThread());
-    lazyInitialize();
     return ConvolverNode::create(this, m_destinationNode->sampleRate());
 }
 
-PassRefPtr<DynamicsCompressorNode> AudioContext::createDynamicsCompressor()
+PassRefPtrWillBeRawPtr<DynamicsCompressorNode> AudioContext::createDynamicsCompressor()
 {
     ASSERT(isMainThread());
-    lazyInitialize();
     return DynamicsCompressorNode::create(this, m_destinationNode->sampleRate());
 }
 
-PassRefPtr<AnalyserNode> AudioContext::createAnalyser()
+PassRefPtrWillBeRawPtr<AnalyserNode> AudioContext::createAnalyser()
 {
     ASSERT(isMainThread());
-    lazyInitialize();
     return AnalyserNode::create(this, m_destinationNode->sampleRate());
 }
 
-PassRefPtr<GainNode> AudioContext::createGain()
+PassRefPtrWillBeRawPtr<GainNode> AudioContext::createGain()
 {
     ASSERT(isMainThread());
-    lazyInitialize();
     return GainNode::create(this, m_destinationNode->sampleRate());
 }
 
-PassRefPtr<DelayNode> AudioContext::createDelay(ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<DelayNode> AudioContext::createDelay(ExceptionState& exceptionState)
 {
     const double defaultMaxDelayTime = 1;
     return createDelay(defaultMaxDelayTime, exceptionState);
 }
 
-PassRefPtr<DelayNode> AudioContext::createDelay(double maxDelayTime, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<DelayNode> AudioContext::createDelay(double maxDelayTime, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
-    lazyInitialize();
-    RefPtr<DelayNode> node = DelayNode::create(this, m_destinationNode->sampleRate(), maxDelayTime, exceptionState);
+    RefPtrWillBeRawPtr<DelayNode> node = DelayNode::create(this, m_destinationNode->sampleRate(), maxDelayTime, exceptionState);
     if (exceptionState.hadException())
-        return 0;
+        return nullptr;
     return node;
 }
 
-PassRefPtr<ChannelSplitterNode> AudioContext::createChannelSplitter(ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<ChannelSplitterNode> AudioContext::createChannelSplitter(ExceptionState& exceptionState)
 {
     const unsigned ChannelSplitterDefaultNumberOfOutputs = 6;
     return createChannelSplitter(ChannelSplitterDefaultNumberOfOutputs, exceptionState);
 }
 
-PassRefPtr<ChannelSplitterNode> AudioContext::createChannelSplitter(size_t numberOfOutputs, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<ChannelSplitterNode> AudioContext::createChannelSplitter(size_t numberOfOutputs, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
-    lazyInitialize();
 
-    RefPtr<ChannelSplitterNode> node = ChannelSplitterNode::create(this, m_destinationNode->sampleRate(), numberOfOutputs);
+    RefPtrWillBeRawPtr<ChannelSplitterNode> node = ChannelSplitterNode::create(this, m_destinationNode->sampleRate(), numberOfOutputs);
 
     if (!node.get()) {
         exceptionState.throwDOMException(
@@ -579,24 +477,23 @@ PassRefPtr<ChannelSplitterNode> AudioContext::createChannelSplitter(size_t numbe
             "number of outputs (" + String::number(numberOfOutputs)
             + ") must be between 1 and "
             + String::number(AudioContext::maxNumberOfChannels()) + ".");
-        return 0;
+        return nullptr;
     }
 
     return node;
 }
 
-PassRefPtr<ChannelMergerNode> AudioContext::createChannelMerger(ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<ChannelMergerNode> AudioContext::createChannelMerger(ExceptionState& exceptionState)
 {
     const unsigned ChannelMergerDefaultNumberOfInputs = 6;
     return createChannelMerger(ChannelMergerDefaultNumberOfInputs, exceptionState);
 }
 
-PassRefPtr<ChannelMergerNode> AudioContext::createChannelMerger(size_t numberOfInputs, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<ChannelMergerNode> AudioContext::createChannelMerger(size_t numberOfInputs, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
-    lazyInitialize();
 
-    RefPtr<ChannelMergerNode> node = ChannelMergerNode::create(this, m_destinationNode->sampleRate(), numberOfInputs);
+    RefPtrWillBeRawPtr<ChannelMergerNode> node = ChannelMergerNode::create(this, m_destinationNode->sampleRate(), numberOfInputs);
 
     if (!node.get()) {
         exceptionState.throwDOMException(
@@ -604,18 +501,17 @@ PassRefPtr<ChannelMergerNode> AudioContext::createChannelMerger(size_t numberOfI
             "number of inputs (" + String::number(numberOfInputs)
             + ") must be between 1 and "
             + String::number(AudioContext::maxNumberOfChannels()) + ".");
-        return 0;
+        return nullptr;
     }
 
     return node;
 }
 
-PassRefPtr<OscillatorNode> AudioContext::createOscillator()
+PassRefPtrWillBeRawPtr<OscillatorNode> AudioContext::createOscillator()
 {
     ASSERT(isMainThread());
-    lazyInitialize();
 
-    RefPtr<OscillatorNode> node = OscillatorNode::create(this, m_destinationNode->sampleRate());
+    RefPtrWillBeRawPtr<OscillatorNode> node = OscillatorNode::create(this, m_destinationNode->sampleRate());
 
     // Because this is an AudioScheduledSourceNode, the context keeps a reference until it has finished playing.
     // When this happens, AudioScheduledSourceNode::finish() calls AudioContext::notifyNodeFinishedProcessing().
@@ -624,7 +520,7 @@ PassRefPtr<OscillatorNode> AudioContext::createOscillator()
     return node;
 }
 
-PassRefPtr<PeriodicWave> AudioContext::createPeriodicWave(Float32Array* real, Float32Array* imag, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<PeriodicWave> AudioContext::createPeriodicWave(Float32Array* real, Float32Array* imag, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
 
@@ -632,14 +528,14 @@ PassRefPtr<PeriodicWave> AudioContext::createPeriodicWave(Float32Array* real, Fl
         exceptionState.throwDOMException(
             SyntaxError,
             "invalid real array");
-        return 0;
+        return nullptr;
     }
 
     if (!imag) {
         exceptionState.throwDOMException(
             SyntaxError,
             "invalid imaginary array");
-        return 0;
+        return nullptr;
     }
 
     if (real->length() != imag->length()) {
@@ -648,7 +544,7 @@ PassRefPtr<PeriodicWave> AudioContext::createPeriodicWave(Float32Array* real, Fl
             "length of real array (" + String::number(real->length())
             + ") and length of imaginary array (" +  String::number(imag->length())
             + ") must match.");
-        return 0;
+        return nullptr;
     }
 
     if (real->length() > 4096) {
@@ -656,7 +552,7 @@ PassRefPtr<PeriodicWave> AudioContext::createPeriodicWave(Float32Array* real, Fl
             IndexSizeError,
             "length of real array (" + String::number(real->length())
             + ") exceeds allowed maximum of 4096");
-        return 0;
+        return nullptr;
     }
 
     if (imag->length() > 4096) {
@@ -664,10 +560,9 @@ PassRefPtr<PeriodicWave> AudioContext::createPeriodicWave(Float32Array* real, Fl
             IndexSizeError,
             "length of imaginary array (" + String::number(imag->length())
             + ") exceeds allowed maximum of 4096");
-        return 0;
+        return nullptr;
     }
 
-    lazyInitialize();
     return PeriodicWave::create(sampleRate(), real, imag);
 }
 
@@ -680,7 +575,7 @@ void AudioContext::notifyNodeFinishedProcessing(AudioNode* node)
 void AudioContext::derefFinishedSourceNodes()
 {
     ASSERT(isGraphOwner());
-    ASSERT(isAudioThread() || isAudioThreadFinished());
+    ASSERT(isAudioThread());
     for (unsigned i = 0; i < m_finishedNodes.size(); i++)
         derefNode(m_finishedNodes[i]);
 
@@ -712,7 +607,7 @@ void AudioContext::derefNode(AudioNode* node)
 
 void AudioContext::derefUnfinishedSourceNodes()
 {
-    ASSERT(isMainThread() && isAudioThreadFinished());
+    ASSERT(isMainThread());
     for (unsigned i = 0; i < m_referencedNodes.size(); ++i)
         m_referencedNodes[i]->deref(AudioNode::RefTypeConnection);
 
@@ -743,7 +638,7 @@ bool AudioContext::tryLock(bool& mustReleaseLock)
     bool isAudioThread = thisThread == audioThread();
 
     // Try to catch cases of using try lock on main thread - it should use regular lock.
-    ASSERT(isAudioThread || isAudioThreadFinished());
+    ASSERT(isAudioThread);
 
     if (!isAudioThread) {
         // In release build treat tryLock() as lock() (since above ASSERT(isAudioThread) never fires) - this is the best we can do.
@@ -858,7 +753,7 @@ void AudioContext::markForDeletion(AudioNode* node)
 {
     ASSERT(isGraphOwner());
 
-    if (isAudioThreadFinished())
+    if (!isInitialized())
         m_nodesToDelete.append(node);
     else
         m_nodesMarkedForDeletion.append(node);
@@ -872,14 +767,14 @@ void AudioContext::markForDeletion(AudioNode* node)
 
 void AudioContext::scheduleNodeDeletion()
 {
-    bool isGood = m_isInitialized && isGraphOwner();
+    bool isGood = isInitialized() && isGraphOwner();
     ASSERT(isGood);
     if (!isGood)
         return;
 
     // Make sure to call deleteMarkedNodes() on main thread.
     if (m_nodesMarkedForDeletion.size() && !m_isDeletionScheduled) {
-        m_nodesToDelete.append(m_nodesMarkedForDeletion);
+        m_nodesToDelete.appendVector(m_nodesMarkedForDeletion);
         m_nodesMarkedForDeletion.clear();
 
         m_isDeletionScheduled = true;
@@ -907,7 +802,7 @@ void AudioContext::deleteMarkedNodes()
     ASSERT(isMainThread());
 
     // Protect this object from being deleted before we release the mutex locked by AutoLocker.
-    RefPtr<AudioContext> protect(this);
+    RefPtrWillBeRawPtr<AudioContext> protect(this);
     {
         AutoLocker locker(this);
 
@@ -924,9 +819,14 @@ void AudioContext::deleteMarkedNodes()
             unsigned numberOfOutputs = node->numberOfOutputs();
             for (unsigned i = 0; i < numberOfOutputs; ++i)
                 m_dirtyAudioNodeOutputs.remove(node->output(i));
-
+#if ENABLE(OILPAN)
+            // Finally, clear the keep alive handle that keeps this
+            // object from being collected.
+            node->clearKeepAlive();
+#else
             // Finally, delete it.
             delete node;
+#endif
         }
         m_isDeletionScheduled = false;
     }
@@ -955,7 +855,7 @@ void AudioContext::handleDirtyAudioSummingJunctions()
 {
     ASSERT(isGraphOwner());
 
-    for (HashSet<AudioSummingJunction*>::iterator i = m_dirtySummingJunctions.begin(); i != m_dirtySummingJunctions.end(); ++i)
+    for (HashSet<AudioSummingJunction* >::iterator i = m_dirtySummingJunctions.begin(); i != m_dirtySummingJunctions.end(); ++i)
         (*i)->updateRenderingState();
 
     m_dirtySummingJunctions.clear();
@@ -1051,14 +951,13 @@ void AudioContext::fireCompletionEvent()
     }
 }
 
-void AudioContext::incrementActiveSourceCount()
-{
-    atomicIncrement(&m_activeSourceCount);
-}
-
-void AudioContext::decrementActiveSourceCount()
+void AudioContext::trace(Visitor* visitor)
 {
-    atomicDecrement(&m_activeSourceCount);
+    visitor->trace(m_renderTarget);
+    visitor->trace(m_destinationNode);
+    visitor->trace(m_listener);
+    visitor->trace(m_dirtySummingJunctions);
+    EventTargetWithInlineData::trace(visitor);
 }
 
 } // namespace WebCore
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioContext.h b/chromium/third_party/WebKit/Source/modules/webaudio/AudioContext.h
index a6ece93825d..d6c9f8b94dd 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioContext.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioContext.h
@@ -28,11 +28,11 @@
 #include "bindings/v8/ScriptWrappable.h"
 #include "core/dom/ActiveDOMObject.h"
 #include "core/events/EventListener.h"
-#include "core/events/EventTarget.h"
-#include "platform/audio/AudioBus.h"
-#include "platform/audio/HRTFDatabaseLoader.h"
+#include "modules/EventTargetModules.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"
@@ -74,43 +74,30 @@ 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>);
+    WILL_BE_USING_GARBAGE_COLLECTED_MIXIN(AudioContext);
 public:
     // Create an AudioContext for rendering to the audio hardware.
-    static PassRefPtr<AudioContext> create(Document&, ExceptionState&);
-
-    // Deprecated: create an AudioContext for offline (non-realtime) rendering.
-    static PassRefPtr<AudioContext> create(Document&, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState&);
+    static PassRefPtrWillBeRawPtr<AudioContext> create(Document&, ExceptionState&);
 
     virtual ~AudioContext();
 
-    bool isInitialized() const;
+    virtual void trace(Visitor*) OVERRIDE;
 
+    bool isInitialized() const { return m_isInitialized; }
     bool isOfflineContext() { return m_isOfflineContext; }
 
-    // Returns true when initialize() was called AND all asynchronous initialization has completed.
-    bool isRunnable() const;
-
-    HRTFDatabaseLoader* hrtfDatabaseLoader() const { return m_hrtfDatabaseLoader.get(); }
-
     // Document notification
-    virtual void stop();
-
-    Document* document() const; // ASSERTs if document no longer exists.
-    bool hasDocument();
+    virtual void stop() OVERRIDE FINAL;
+    virtual bool hasPendingActivity() const OVERRIDE;
 
     AudioDestinationNode* destination() { return m_destinationNode.get(); }
     size_t currentSampleFrame() const { return m_destinationNode->currentSampleFrame(); }
     double currentTime() const { return m_destinationNode->currentTime(); }
     float sampleRate() const { return m_destinationNode->sampleRate(); }
-    unsigned long activeSourceCount() const { return static_cast<unsigned long>(m_activeSourceCount); }
 
-    void incrementActiveSourceCount();
-    void decrementActiveSourceCount();
-
-    PassRefPtr<AudioBuffer> createBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState&);
-    PassRefPtr<AudioBuffer> createBuffer(ArrayBuffer*, bool mixToMono, 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&);
@@ -118,29 +105,29 @@ 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*);
@@ -183,9 +170,6 @@ public:
     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);
 
@@ -234,8 +218,8 @@ public:
     void removeMarkedSummingJunction(AudioSummingJunction*);
 
     // EventTarget
-    virtual const AtomicString& interfaceName() const OVERRIDE;
-    virtual ExecutionContext* executionContext() const OVERRIDE;
+    virtual const AtomicString& interfaceName() const OVERRIDE FINAL;
+    virtual ExecutionContext* executionContext() const OVERRIDE FINAL;
 
     DEFINE_ATTRIBUTE_EVENT_LISTENER(complete);
 
@@ -251,22 +235,21 @@ protected:
     static bool isSampleRateRangeGood(float sampleRate);
 
 private:
-    void constructCommon();
-
-    void lazyInitialize();
+    void initialize();
     void uninitialize();
 
     // ExecutionContext calls stop twice.
     // We'd like to schedule only one stop action for them.
     bool m_isStopScheduled;
     static void stopDispatch(void* userData);
+    bool m_isCleared;
     void clear();
 
     void scheduleNodeDeletion();
     static void deleteMarkedNodesDispatch(void* userData);
 
+    // Set to true when the destination node has been initialized and is ready to process data.
     bool m_isInitialized;
-    bool m_isAudioThreadFinished;
 
     // 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.
@@ -279,8 +262,8 @@ 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;
@@ -301,7 +284,7 @@ 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();
@@ -324,10 +307,7 @@ private:
     // Only accessed in the audio thread.
     Vector<AudioNode*> m_deferredFinishDerefList;
 
-    // HRTF Database loader
-    RefPtr<HRTFDatabaseLoader> m_hrtfDatabaseLoader;
-
-    RefPtr<AudioBuffer> m_renderTarget;
+    RefPtrWillBeMember<AudioBuffer> m_renderTarget;
 
     bool m_isOfflineContext;
 
@@ -336,9 +316,6 @@ private:
     // 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).
-    int m_activeSourceCount;
 };
 
 } // WebCore
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioContext.idl b/chromium/third_party/WebKit/Source/modules/webaudio/AudioContext.idl
index 52da1a57277..f5a932a181c 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioContext.idl
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioContext.idl
@@ -24,19 +24,20 @@
  */
 
 [
+    WillBeGarbageCollected,
     ActiveDOMObject,
     Conditional=WEB_AUDIO,
     Constructor,
-    Constructor(unsigned long numberOfChannels, unsigned long numberOfFrames, float sampleRate),
     ConstructorCallWith=Document,
     NoInterfaceObject,
     RaisesException=Constructor,
 ] interface AudioContext : EventTarget {
+    
     // All rendered audio ultimately connects to destination, which represents the audio hardware.
     readonly attribute AudioDestinationNode destination;
 
     // All scheduled times are relative to this time in seconds.
-    readonly attribute float currentTime;
+    readonly attribute double currentTime;
 
     // All AudioNodes in the context run at this sample-rate (sample-frames per second).
     readonly attribute float sampleRate;
@@ -44,11 +45,7 @@
     // All panning is relative to this listener.
     readonly attribute AudioListener listener;
 
-    // Number of AudioBufferSourceNodes that are currently playing.
-    readonly attribute unsigned long activeSourceCount;
-
     [RaisesException] AudioBuffer createBuffer(unsigned long numberOfChannels, unsigned long numberOfFrames, float sampleRate);
-    [RaisesException] AudioBuffer createBuffer(ArrayBuffer? buffer, boolean mixToMono);
 
     // Asynchronous audio file data decoding.
     [RaisesException] void decodeAudioData(ArrayBuffer audioData, AudioBufferCallback successCallback, optional AudioBufferCallback errorCallback);
@@ -82,10 +79,4 @@
     // void prepareOfflineBufferRendering(unsigned long numberOfChannels, unsigned long numberOfFrames, float sampleRate);
     attribute EventHandler oncomplete;
     void startRendering();
-
-    [MeasureAs=LegacyWebAudio, ImplementedAs=createGain] GainNode createGainNode();
-    [MeasureAs=LegacyWebAudio, ImplementedAs=createDelay, RaisesException] DelayNode createDelayNode(optional double maxDelayTime);
-
-    [MeasureAs=LegacyWebAudio, ImplementedAs=createScriptProcessor, RaisesException] ScriptProcessorNode createJavaScriptNode(unsigned long bufferSize, optional unsigned long numberOfInputChannels, optional unsigned long numberOfOutputChannels);
-
 };
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioDestinationNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/AudioDestinationNode.cpp
index 68a6d7285e1..807c1d7a8ff 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioDestinationNode.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioDestinationNode.cpp
@@ -60,7 +60,7 @@ void AudioDestinationNode::render(AudioBus* sourceBus, AudioBus* destinationBus,
 
     context()->setAudioThread(currentThread());
 
-    if (!context()->isRunnable()) {
+    if (!context()->isInitialized()) {
         destinationBus->zero();
         return;
     }
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioDestinationNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/AudioDestinationNode.h
index 801d7fd656d..182648732f8 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioDestinationNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioDestinationNode.h
@@ -42,29 +42,23 @@ public:
     virtual ~AudioDestinationNode();
 
     // AudioNode
-    virtual void process(size_t) { }; // we're pulled by hardware so this is never called
-    virtual void reset() { m_currentSampleFrame = 0; };
+    virtual void process(size_t) OVERRIDE FINAL { } // we're pulled by hardware so this is never called
 
     // The audio hardware calls render() to get the next render quantum of audio into destinationBus.
     // It will optionally give us local/live audio input in sourceBus (if it's not 0).
-    virtual void render(AudioBus* sourceBus, AudioBus* destinationBus, size_t numberOfFrames);
+    virtual void render(AudioBus* sourceBus, AudioBus* destinationBus, size_t numberOfFrames) OVERRIDE FINAL;
 
     size_t currentSampleFrame() const { return m_currentSampleFrame; }
     double currentTime() const { return currentSampleFrame() / static_cast<double>(sampleRate()); }
 
     virtual unsigned long maxChannelCount() const { return 0; }
 
-    // Enable local/live input for the specified device.
-    virtual void enableInput(const String& inputDeviceId) = 0;
-
     virtual void startRendering() = 0;
 
-    AudioSourceProvider* localAudioInputProvider() { return &m_localAudioInputProvider; }
-
 protected:
     // LocalAudioInputProvider allows us to expose an AudioSourceProvider for local/live audio input.
     // If there is local/live audio input, we call set() with the audio input data every render quantum.
-    class LocalAudioInputProvider : public AudioSourceProvider {
+    class LocalAudioInputProvider FINAL : public AudioSourceProvider {
     public:
         LocalAudioInputProvider()
             : m_sourceBus(AudioBus::create(2, AudioNode::ProcessingSizeInFrames)) // FIXME: handle non-stereo local input.
@@ -78,7 +72,7 @@ protected:
         }
 
         // AudioSourceProvider.
-        virtual void provideInput(AudioBus* destinationBus, size_t numberOfFrames)
+        virtual void provideInput(AudioBus* destinationBus, size_t numberOfFrames) OVERRIDE
         {
             bool isGood = destinationBus && destinationBus->length() == numberOfFrames && m_sourceBus->length() == numberOfFrames;
             ASSERT(isGood);
@@ -90,8 +84,8 @@ protected:
         RefPtr<AudioBus> m_sourceBus;
     };
 
-    virtual double tailTime() const OVERRIDE { return 0; }
-    virtual double latencyTime() const OVERRIDE { return 0; }
+    virtual double tailTime() const OVERRIDE FINAL { return 0; }
+    virtual double latencyTime() const OVERRIDE FINAL { return 0; }
 
     // Counts the number of sample-frames processed by the destination.
     size_t m_currentSampleFrame;
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioListener.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/AudioListener.cpp
index e13794597bb..94466c9a19f 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioListener.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioListener.cpp
@@ -32,6 +32,7 @@
 
 #include "modules/webaudio/AudioListener.h"
 
+#include "modules/webaudio/PannerNode.h"
 #include "platform/audio/AudioBus.h"
 
 namespace WebCore {
@@ -47,6 +48,101 @@ AudioListener::AudioListener()
     ScriptWrappable::init(this);
 }
 
+AudioListener::~AudioListener()
+{
+    m_panners.clear();
+}
+
+void AudioListener::addPanner(PannerNode* panner)
+{
+    if (!panner)
+        return;
+
+    m_panners.append(panner);
+}
+
+void AudioListener::removePanner(PannerNode* panner)
+{
+    for (unsigned i = 0; i < m_panners.size(); ++i) {
+        if (panner == m_panners[i]) {
+            m_panners.remove(i);
+            break;
+        }
+    }
+}
+
+void AudioListener::markPannersAsDirty(unsigned type)
+{
+    for (unsigned i = 0; i < m_panners.size(); ++i)
+        m_panners[i]->markPannerAsDirty(type);
+}
+
+void AudioListener::setPosition(const FloatPoint3D &position)
+{
+    if (m_position == position)
+        return;
+
+    // This synchronizes with panner's process().
+    MutexLocker listenerLocker(m_listenerLock);
+    m_position = position;
+    markPannersAsDirty(PannerNode::AzimuthElevationDirty | PannerNode::DistanceConeGainDirty | PannerNode::DopplerRateDirty);
+}
+
+void AudioListener::setOrientation(const FloatPoint3D &orientation)
+{
+    if (m_orientation == orientation)
+        return;
+
+    // This synchronizes with panner's process().
+    MutexLocker listenerLocker(m_listenerLock);
+    m_orientation = orientation;
+    markPannersAsDirty(PannerNode::AzimuthElevationDirty);
+}
+
+void AudioListener::setUpVector(const FloatPoint3D &upVector)
+{
+    if (m_upVector == upVector)
+        return;
+
+    // This synchronizes with panner's process().
+    MutexLocker listenerLocker(m_listenerLock);
+    m_upVector = upVector;
+    markPannersAsDirty(PannerNode::AzimuthElevationDirty);
+}
+
+void AudioListener::setVelocity(const FloatPoint3D &velocity)
+{
+    if (m_velocity == velocity)
+        return;
+
+    // This synchronizes with panner's process().
+    MutexLocker listenerLocker(m_listenerLock);
+    m_velocity = velocity;
+    markPannersAsDirty(PannerNode::DopplerRateDirty);
+}
+
+void AudioListener::setDopplerFactor(double dopplerFactor)
+{
+    if (m_dopplerFactor == dopplerFactor)
+        return;
+
+    // This synchronizes with panner's process().
+    MutexLocker listenerLocker(m_listenerLock);
+    m_dopplerFactor = dopplerFactor;
+    markPannersAsDirty(PannerNode::DopplerRateDirty);
+}
+
+void AudioListener::setSpeedOfSound(double speedOfSound)
+{
+    if (m_speedOfSound == speedOfSound)
+        return;
+
+    // This synchronizes with panner's process().
+    MutexLocker listenerLocker(m_listenerLock);
+    m_speedOfSound = speedOfSound;
+    markPannersAsDirty(PannerNode::DopplerRateDirty);
+}
+
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioListener.h b/chromium/third_party/WebKit/Source/modules/webaudio/AudioListener.h
index dcd68dd42ba..214db390b70 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioListener.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioListener.h
@@ -31,63 +31,78 @@
 
 #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"
 
 namespace WebCore {
 
+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();
 
     // Position
     void setPosition(float x, float y, float z) { setPosition(FloatPoint3D(x, y, z)); }
-    void setPosition(const FloatPoint3D &position) { m_position = position; }
     const FloatPoint3D& position() const { return m_position; }
 
-    // Orientation
+    // Orientation and Up-vector
     void setOrientation(float x, float y, float z, float upX, float upY, float upZ)
     {
         setOrientation(FloatPoint3D(x, y, z));
         setUpVector(FloatPoint3D(upX, upY, upZ));
     }
-    void setOrientation(const FloatPoint3D &orientation) { m_orientation = orientation; }
     const FloatPoint3D& orientation() const { return m_orientation; }
-
-    // Up-vector
-    void setUpVector(const FloatPoint3D &upVector) { m_upVector = upVector; }
     const FloatPoint3D& upVector() const { return m_upVector; }
 
     // Velocity
     void setVelocity(float x, float y, float z) { setVelocity(FloatPoint3D(x, y, z)); }
-    void setVelocity(const FloatPoint3D &velocity) { m_velocity = velocity; }
     const FloatPoint3D& velocity() const { return m_velocity; }
 
     // Doppler factor
-    void setDopplerFactor(double dopplerFactor) { m_dopplerFactor = dopplerFactor; }
+    void setDopplerFactor(double);
     double dopplerFactor() const { return m_dopplerFactor; }
 
     // Speed of sound
-    void setSpeedOfSound(double speedOfSound) { m_speedOfSound = speedOfSound; }
+    void setSpeedOfSound(double);
     double speedOfSound() const { return m_speedOfSound; }
 
+    Mutex& listenerLock() { return m_listenerLock; }
+    void addPanner(PannerNode*);
+    void removePanner(PannerNode*);
+
+    void trace(Visitor*) { }
+
 private:
     AudioListener();
 
-    // Position / Orientation
+    void setPosition(const FloatPoint3D&);
+    void setOrientation(const FloatPoint3D&);
+    void setUpVector(const FloatPoint3D&);
+    void setVelocity(const FloatPoint3D&);
+
+    void markPannersAsDirty(unsigned);
+
     FloatPoint3D m_position;
     FloatPoint3D m_orientation;
     FloatPoint3D m_upVector;
-
     FloatPoint3D m_velocity;
-
     double m_dopplerFactor;
     double m_speedOfSound;
+
+    // Synchronize a panner's process() with setting of the state of the listener.
+    mutable Mutex m_listenerLock;
+
+    // List for pannerNodes in context.
+    Vector<PannerNode*> m_panners;
 };
 
 } // WebCore
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioListener.idl b/chromium/third_party/WebKit/Source/modules/webaudio/AudioListener.idl
index 0011fe0f560..b9668f513fa 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioListener.idl
+++ b/chromium/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)
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/AudioNode.cpp
index 7e4b075f998..e48936c7335 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioNode.cpp
+++ b/chromium/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)
@@ -139,12 +142,6 @@ void AudioNode::setNodeType(NodeType type)
 #endif
 }
 
-void AudioNode::lazyInitialize()
-{
-    if (!isInitialized())
-        initialize();
-}
-
 void AudioNode::addInput(PassOwnPtr<AudioNodeInput> input)
 {
     m_inputs.append(input);
@@ -309,9 +306,7 @@ void AudioNode::setChannelCountMode(const String& mode, ExceptionState& exceptio
     } else if (mode == "explicit") {
         m_channelCountMode = Explicit;
     } else {
-        exceptionState.throwDOMException(
-            InvalidStateError,
-            "invalid mode '" + mode + "'; must be 'max', 'clamped-max', or 'explicit'.");
+        ASSERT_NOT_REACHED();
     }
 
     if (m_channelCountMode != oldMode)
@@ -340,9 +335,7 @@ void AudioNode::setChannelInterpretation(const String& interpretation, Exception
     } else if (interpretation == "discrete") {
         m_channelInterpretation = AudioBus::Discrete;
     } else {
-        exceptionState.throwDOMException(
-            InvalidStateError,
-            "invalid interpretation '" + interpretation + "'; must be 'speakers' or 'discrete'.");
+        ASSERT_NOT_REACHED();
     }
 }
 
@@ -478,6 +471,9 @@ void AudioNode::disableOutputsIfNecessary()
 
 void AudioNode::ref(RefType refType)
 {
+#if ENABLE(OILPAN)
+    ASSERT(m_keepAlive);
+#endif
     switch (refType) {
     case RefTypeNormal:
         atomicIncrement(&m_normalRefCount);
@@ -531,7 +527,7 @@ void AudioNode::deref(RefType refType)
     // Once AudioContext::uninitialize() is called there's no more chances for deleteMarkedNodes() to get called, so we call here.
     // We can't call in AudioContext::~AudioContext() since it will never be called as long as any AudioNode is alive
     // because AudioNodes keep a reference to the context.
-    if (context()->isAudioThreadFinished())
+    if (!context()->isInitialized())
         context()->deleteMarkedNodes();
 }
 
@@ -592,6 +588,25 @@ void AudioNode::printNodeCounts()
 
 #endif // DEBUG_AUDIONODE_REFERENCES
 
+void AudioNode::trace(Visitor* visitor)
+{
+    visitor->trace(m_context);
+    EventTargetWithInlineData::trace(visitor);
+}
+
+#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)
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/AudioNode.h
index 61fc047bdd5..fb63c918e40 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioNode.h
@@ -26,7 +26,7 @@
 #define AudioNode_h
 
 #include "bindings/v8/ScriptWrappable.h"
-#include "core/events/EventTarget.h"
+#include "modules/EventTargetModules.h"
 #include "platform/audio/AudioBus.h"
 #include "wtf/Forward.h"
 #include "wtf/OwnPtr.h"
@@ -50,7 +50,9 @@ 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 {
+    WILL_BE_USING_GARBAGE_COLLECTED_MIXIN(AudioNode);
 public:
     enum { ProcessingSizeInFrames = 128 };
 
@@ -109,17 +111,12 @@ public:
     // Called from context's audio thread.
     virtual void process(size_t framesToProcess) = 0;
 
-    // Resets DSP processing state (clears delay lines, filter memory, etc.)
-    // Called from context's audio thread.
-    virtual void reset() = 0;
-
     // No significant resources should be allocated until initialize() is called.
     // Processing may not occur until a node is initialized.
     virtual void initialize();
     virtual void uninitialize();
 
     bool isInitialized() const { return m_isInitialized; }
-    void lazyInitialize();
 
     unsigned numberOfInputs() const { return m_inputs.size(); }
     unsigned numberOfOutputs() const { return m_outputs.size(); }
@@ -181,8 +178,14 @@ public:
     AudioBus::ChannelInterpretation internalChannelInterpretation() const { return m_channelInterpretation; }
 
     // EventTarget
-    virtual const AtomicString& interfaceName() const OVERRIDE;
-    virtual ExecutionContext* executionContext() const OVERRIDE;
+    virtual const AtomicString& interfaceName() const OVERRIDE FINAL;
+    virtual ExecutionContext* executionContext() const OVERRIDE FINAL;
+
+    virtual void trace(Visitor*) OVERRIDE;
+
+#if ENABLE(OILPAN)
+    void clearKeepAlive();
+#endif
 
 protected:
     // Inputs and outputs must be created before the AudioNode is initialized.
@@ -200,11 +203,23 @@ 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;
 
@@ -220,8 +235,10 @@ private:
     static int s_nodeCount[NodeTypeEnd];
 #endif
 
-    virtual void refEventTarget() OVERRIDE { ref(); }
-    virtual void derefEventTarget() OVERRIDE { deref(); }
+#if !ENABLE(OILPAN)
+    virtual void refEventTarget() OVERRIDE FINAL { ref(); }
+    virtual void derefEventTarget() OVERRIDE FINAL { deref(); }
+#endif
 
 protected:
     unsigned m_channelCount;
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioNode.idl b/chromium/third_party/WebKit/Source/modules/webaudio/AudioNode.idl
index 35d98ccf302..4ed4d40b0db 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioNode.idl
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioNode.idl
@@ -22,23 +22,32 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
+enum ChannelCountMode {
+    "max",
+    "clamped-max",
+    "explicit"
+};
+
+enum ChannelInterpretation {
+    "speakers",
+    "discrete"
+};
+
+// http://webaudio.github.io/web-audio-api/#idl-def-AudioNode
+
 [
     Conditional=WEB_AUDIO,
     Custom=Wrap,
+    WillBeGarbageCollected,
 ] interface AudioNode : EventTarget {
-    readonly attribute AudioContext context;
-    readonly attribute unsigned long numberOfInputs;
-    readonly attribute unsigned long numberOfOutputs;
-
-    [RaisesException=Setter] attribute unsigned long channelCount;
-
-    [RaisesException=Setter] attribute DOMString channelCountMode;
-
-    [RaisesException=Setter] attribute DOMString channelInterpretation;
-
+    // FIXME: AudioNode argument should not be nullable
     [RaisesException] void connect(AudioNode? destination, [Default=Undefined] optional unsigned long output, [Default=Undefined] optional unsigned long input);
-
-    [RaisesException] void connect(AudioParam? destination, [Default=Undefined] optional unsigned long output);
-
+    [RaisesException] void connect(AudioParam destination, [Default=Undefined] optional unsigned long output);
     [RaisesException] void disconnect([Default=Undefined] optional unsigned long output);
+    readonly attribute AudioContext  context;
+    readonly attribute unsigned long numberOfInputs;
+    readonly attribute unsigned long numberOfOutputs;
+    [RaisesException=Setter] attribute unsigned long         channelCount;
+    [RaisesException=Setter] attribute ChannelCountMode      channelCountMode;
+    [RaisesException=Setter] attribute ChannelInterpretation channelInterpretation;
 };
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioNodeInput.h b/chromium/third_party/WebKit/Source/modules/webaudio/AudioNodeInput.h
index 9f171a26ecf..8952863d554 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioNodeInput.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioNodeInput.h
@@ -40,7 +40,7 @@ class AudioNodeOutput;
 // In the case of multiple connections, the input will act as a unity-gain summing junction, mixing all the outputs.
 // The number of channels of the input's bus is the maximum of the number of channels of all its connections.
 
-class AudioNodeInput : public AudioSummingJunction {
+class AudioNodeInput FINAL : public AudioSummingJunction {
 public:
     explicit AudioNodeInput(AudioNode*);
 
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioNodeOutput.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/AudioNodeOutput.cpp
index 9902699bb95..ef7a7ab2a01 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioNodeOutput.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioNodeOutput.cpp
@@ -146,11 +146,6 @@ unsigned AudioNodeOutput::renderingFanOutCount() const
     return m_renderingFanOutCount;
 }
 
-unsigned AudioNodeOutput::renderingParamFanOutCount() const
-{
-    return m_renderingParamFanOutCount;
-}
-
 void AudioNodeOutput::addInput(AudioNodeInput* input)
 {
     ASSERT(context()->isGraphOwner());
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioNodeOutput.h b/chromium/third_party/WebKit/Source/modules/webaudio/AudioNodeOutput.h
index 14e9272a306..31f9637ecc7 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioNodeOutput.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioNodeOutput.h
@@ -62,10 +62,6 @@ public:
     // Unlike fanOutCount() it will not change during the course of a render quantum.
     unsigned renderingFanOutCount() const;
 
-    // renderingParamFanOutCount() is the number of AudioParams that we're connected to during rendering.
-    // Unlike paramFanOutCount() it will not change during the course of a render quantum.
-    unsigned renderingParamFanOutCount() const;
-
     // Must be called with the context's graph lock.
     void disconnectAll();
 
@@ -144,8 +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
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioParam.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/AudioParam.cpp
index 8ff48e2ed9e..a356c93bd54 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioParam.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioParam.cpp
@@ -84,7 +84,7 @@ bool AudioParam::smooth()
         m_smoothedValue = m_value;
     else {
         // Dezipper - exponential approach.
-        m_smoothedValue += (m_value - m_smoothedValue) * m_smoothingConstant;
+        m_smoothedValue += (m_value - m_smoothedValue) * DefaultSmoothingConstant;
 
         // If we get close enough then snap to actual value.
         if (fabs(m_smoothedValue - m_value) < SnapThreshold) // FIXME: the threshold needs to be adjustable depending on range - but this is OK general purpose value.
@@ -96,7 +96,7 @@ bool AudioParam::smooth()
 
 float AudioParam::finalValue()
 {
-    float value;
+    float value = m_value;
     calculateFinalValues(&value, 1, false);
     return value;
 }
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioParam.h b/chromium/third_party/WebKit/Source/modules/webaudio/AudioParam.h
index b6c30cbaeb8..ed25afabf79 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioParam.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioParam.h
@@ -43,14 +43,14 @@ namespace WebCore {
 
 class AudioNodeOutput;
 
-class AudioParam : 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
@@ -83,12 +83,11 @@ public:
     bool smooth();
 
     void resetSmoothedValue() { m_smoothedValue = m_value; }
-    void setSmoothingConstant(double k) { m_smoothingConstant = k; }
 
     // Parameter automation.
     void setValueAtTime(float value, double time) { m_timeline.setValueAtTime(value, time); }
     void linearRampToValueAtTime(float value, double time) { m_timeline.linearRampToValueAtTime(value, time); }
-    void exponentialRampToValueAtTime(float value, double time) { m_timeline.exponentialRampToValueAtTime(value, time); }
+    void exponentialRampToValueAtTime(float value, double time, ExceptionState& es) { m_timeline.exponentialRampToValueAtTime(value, time, es); }
     void setTargetAtTime(float target, double time, double timeConstant) { m_timeline.setTargetAtTime(target, time, timeConstant); }
     void setValueCurveAtTime(Float32Array* curve, double time, double duration) { m_timeline.setValueCurveAtTime(curve, time, duration); }
     void cancelScheduledValues(double startTime) { m_timeline.cancelScheduledValues(startTime); }
@@ -103,6 +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)
@@ -113,7 +114,6 @@ protected:
         , m_maxValue(maxValue)
         , m_units(units)
         , m_smoothedValue(defaultValue)
-        , m_smoothingConstant(DefaultSmoothingConstant)
     {
         ScriptWrappable::init(this);
     }
@@ -132,7 +132,6 @@ private:
 
     // Smoothing (de-zippering)
     double m_smoothedValue;
-    double m_smoothingConstant;
 
     AudioParamTimeline m_timeline;
 };
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioParam.idl b/chromium/third_party/WebKit/Source/modules/webaudio/AudioParam.idl
index bb903874f3b..2d55ad31792 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioParam.idl
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioParam.idl
@@ -27,6 +27,7 @@
  */
 
 [
+    WillBeGarbageCollected,
     Conditional=WEB_AUDIO
 ] interface AudioParam {
     attribute float value;
@@ -42,7 +43,7 @@
     // Parameter automation.
     void setValueAtTime(float value, double time);
     void linearRampToValueAtTime(float value, double time);
-    void exponentialRampToValueAtTime(float value, double time);
+    [RaisesException] void exponentialRampToValueAtTime(float value, double time);
 
     // Exponentially approach the target with a rate having the given time constant.
     void setTargetAtTime(float target, double time, double timeConstant);
@@ -54,6 +55,4 @@
     // Cancels all scheduled parameter changes with times greater than or equal to startTime.
     void cancelScheduledValues(double startTime);
 
-    [MeasureAs=LegacyWebAudio, ImplementedAs=setTargetAtTime] void setTargetValueAtTime(float targetValue, double time, double timeConstant);
-
 };
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioParamTimeline.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/AudioParamTimeline.cpp
index 51d5293244a..eb107c04c1a 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioParamTimeline.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioParamTimeline.cpp
@@ -29,6 +29,8 @@
 
 #include "modules/webaudio/AudioParamTimeline.h"
 
+#include "bindings/v8/ExceptionState.h"
+#include "core/dom/ExceptionCode.h"
 #include "platform/audio/AudioUtilities.h"
 #include "platform/FloatConversion.h"
 #include "wtf/MathExtras.h"
@@ -40,22 +42,30 @@ namespace WebCore {
 
 void AudioParamTimeline::setValueAtTime(float value, double time)
 {
-    insertEvent(ParamEvent(ParamEvent::SetValue, value, time, 0, 0, 0));
+    insertEvent(ParamEvent(ParamEvent::SetValue, value, time, 0, 0, nullptr));
 }
 
 void AudioParamTimeline::linearRampToValueAtTime(float value, double time)
 {
-    insertEvent(ParamEvent(ParamEvent::LinearRampToValue, value, time, 0, 0, 0));
+    insertEvent(ParamEvent(ParamEvent::LinearRampToValue, value, time, 0, 0, nullptr));
 }
 
-void AudioParamTimeline::exponentialRampToValueAtTime(float value, double time)
+void AudioParamTimeline::exponentialRampToValueAtTime(float value, double time, ExceptionState& exceptionState)
 {
-    insertEvent(ParamEvent(ParamEvent::ExponentialRampToValue, value, time, 0, 0, 0));
+    ASSERT(isMainThread());
+    if (value <= 0) {
+        exceptionState.throwDOMException(
+            InvalidStateError,
+            "Target value for exponential ramp must be positive: " + String::number(value));
+        return;
+    }
+
+    insertEvent(ParamEvent(ParamEvent::ExponentialRampToValue, value, time, 0, 0, nullptr));
 }
 
 void AudioParamTimeline::setTargetAtTime(float target, double time, double timeConstant)
 {
-    insertEvent(ParamEvent(ParamEvent::SetTarget, target, time, timeConstant, 0, 0));
+    insertEvent(ParamEvent(ParamEvent::SetTarget, target, time, timeConstant, 0, nullptr));
 }
 
 void AudioParamTimeline::setValueCurveAtTime(Float32Array* curve, double time, double duration)
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioParamTimeline.h b/chromium/third_party/WebKit/Source/modules/webaudio/AudioParamTimeline.h
index 335a9ac2116..350d3bcb89f 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioParamTimeline.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioParamTimeline.h
@@ -46,7 +46,7 @@ public:
 
     void setValueAtTime(float value, double time);
     void linearRampToValueAtTime(float value, double time);
-    void exponentialRampToValueAtTime(float value, double time);
+    void exponentialRampToValueAtTime(float value, double time, ExceptionState&);
     void setTargetAtTime(float target, double time, double timeConstant);
     void setValueCurveAtTime(Float32Array* curve, double time, double duration);
     void cancelScheduledValues(double startTime);
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioProcessingEvent.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/AudioProcessingEvent.cpp
index ac8eb99bcbc..112994af876 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioProcessingEvent.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioProcessingEvent.cpp
@@ -28,18 +28,16 @@
 
 #include "modules/webaudio/AudioProcessingEvent.h"
 
-#include "core/events/ThreadLocalEventNames.h"
-
 namespace WebCore {
 
-PassRefPtr<AudioProcessingEvent> AudioProcessingEvent::create()
+PassRefPtrWillBeRawPtr<AudioProcessingEvent> AudioProcessingEvent::create()
 {
-    return adoptRef(new AudioProcessingEvent);
+    return adoptRefWillBeNoop(new AudioProcessingEvent);
 }
 
-PassRefPtr<AudioProcessingEvent> AudioProcessingEvent::create(PassRefPtr<AudioBuffer> inputBuffer, PassRefPtr<AudioBuffer> outputBuffer)
+PassRefPtrWillBeRawPtr<AudioProcessingEvent> AudioProcessingEvent::create(PassRefPtrWillBeRawPtr<AudioBuffer> inputBuffer, PassRefPtrWillBeRawPtr<AudioBuffer> outputBuffer, double playbackTime)
 {
-    return adoptRef(new AudioProcessingEvent(inputBuffer, outputBuffer));
+    return adoptRefWillBeNoop(new AudioProcessingEvent(inputBuffer, outputBuffer, playbackTime));
 }
 
 AudioProcessingEvent::AudioProcessingEvent()
@@ -47,10 +45,11 @@ AudioProcessingEvent::AudioProcessingEvent()
     ScriptWrappable::init(this);
 }
 
-AudioProcessingEvent::AudioProcessingEvent(PassRefPtr<AudioBuffer> inputBuffer, PassRefPtr<AudioBuffer> outputBuffer)
+AudioProcessingEvent::AudioProcessingEvent(PassRefPtrWillBeRawPtr<AudioBuffer> inputBuffer, PassRefPtrWillBeRawPtr<AudioBuffer> outputBuffer, double playbackTime)
     : Event(EventTypeNames::audioprocess, true, false)
     , m_inputBuffer(inputBuffer)
     , m_outputBuffer(outputBuffer)
+    , m_playbackTime(playbackTime)
 {
     ScriptWrappable::init(this);
 }
@@ -64,6 +63,13 @@ const AtomicString& AudioProcessingEvent::interfaceName() const
     return EventNames::AudioProcessingEvent;
 }
 
+void AudioProcessingEvent::trace(Visitor* visitor)
+{
+    visitor->trace(m_inputBuffer);
+    visitor->trace(m_outputBuffer);
+    Event::trace(visitor);
+}
+
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioProcessingEvent.h b/chromium/third_party/WebKit/Source/modules/webaudio/AudioProcessingEvent.h
index de2d78332f5..e938fe75748 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioProcessingEvent.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioProcessingEvent.h
@@ -25,7 +25,7 @@
 #ifndef AudioProcessingEvent_h
 #define AudioProcessingEvent_h
 
-#include "core/events/Event.h"
+#include "modules/EventModules.h"
 #include "modules/webaudio/AudioBuffer.h"
 #include "wtf/PassRefPtr.h"
 #include "wtf/RefPtr.h"
@@ -34,24 +34,28 @@ namespace WebCore {
 
 class AudioBuffer;
 
-class AudioProcessingEvent : public Event {
+class AudioProcessingEvent FINAL : public Event {
 public:
-    static PassRefPtr<AudioProcessingEvent> create();
-    static PassRefPtr<AudioProcessingEvent> create(PassRefPtr<AudioBuffer> inputBuffer, PassRefPtr<AudioBuffer> outputBuffer);
+    static PassRefPtrWillBeRawPtr<AudioProcessingEvent> create();
+    static PassRefPtrWillBeRawPtr<AudioProcessingEvent> create(PassRefPtrWillBeRawPtr<AudioBuffer> inputBuffer, PassRefPtrWillBeRawPtr<AudioBuffer> outputBuffer, double playbackTime);
 
     virtual ~AudioProcessingEvent();
 
     AudioBuffer* inputBuffer() { return m_inputBuffer.get(); }
     AudioBuffer* outputBuffer() { return m_outputBuffer.get(); }
+    double playbackTime() const { return m_playbackTime; }
 
-    virtual const AtomicString& interfaceName() const;
+    virtual const AtomicString& interfaceName() const OVERRIDE;
+
+    virtual void trace(Visitor*) OVERRIDE;
 
 private:
     AudioProcessingEvent();
-    AudioProcessingEvent(PassRefPtr<AudioBuffer> inputBuffer, PassRefPtr<AudioBuffer> outputBuffer);
+    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;
 };
 
 } // namespace WebCore
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioProcessingEvent.idl b/chromium/third_party/WebKit/Source/modules/webaudio/AudioProcessingEvent.idl
index ed8f90a0a3a..2b812825dd0 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioProcessingEvent.idl
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioProcessingEvent.idl
@@ -25,6 +25,7 @@
 [
     Conditional=WEB_AUDIO
 ] interface AudioProcessingEvent : Event {
+    readonly attribute double playbackTime;
     readonly attribute AudioBuffer inputBuffer;
     readonly attribute AudioBuffer outputBuffer;
 };
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioScheduledSourceNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/AudioScheduledSourceNode.cpp
index f53e18cf44e..e96f2c89ff9 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioScheduledSourceNode.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioScheduledSourceNode.cpp
@@ -30,9 +30,9 @@
 
 #include "bindings/v8/ExceptionState.h"
 #include "core/dom/ExceptionCode.h"
-#include "core/events/Event.h"
-#include "platform/audio/AudioUtilities.h"
+#include "modules/EventModules.h"
 #include "modules/webaudio/AudioContext.h"
+#include "platform/audio/AudioUtilities.h"
 #include <algorithm>
 #include "wtf/MathExtras.h"
 
@@ -48,7 +48,6 @@ AudioScheduledSourceNode::AudioScheduledSourceNode(AudioContext* context, float
     , m_startTime(0)
     , m_endTime(UnknownTime)
     , m_hasEndedListener(false)
-    , m_stopCalled(false)
 {
 }
 
@@ -91,7 +90,6 @@ void AudioScheduledSourceNode::updateSchedulingInfo(size_t quantumFrameSize,
     if (m_playbackState == SCHEDULED_STATE) {
         // Increment the active source count only if we're transitioning from SCHEDULED_STATE to PLAYING_STATE.
         m_playbackState = PLAYING_STATE;
-        context()->incrementActiveSourceCount();
     }
 
     quantumFrameOffset = startFrame > quantumStartFrame ? startFrame - quantumStartFrame : 0;
@@ -137,7 +135,6 @@ void AudioScheduledSourceNode::updateSchedulingInfo(size_t quantumFrameSize,
     return;
 }
 
-
 void AudioScheduledSourceNode::start(double when, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
@@ -157,28 +154,23 @@ void AudioScheduledSourceNode::stop(double when, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
 
-    if (m_stopCalled) {
-        exceptionState.throwDOMException(
-            InvalidStateError,
-            "cannot call stop more than once.");
-    } else if (m_playbackState == UNSCHEDULED_STATE) {
+    if (m_playbackState == UNSCHEDULED_STATE) {
         exceptionState.throwDOMException(
             InvalidStateError,
             "cannot call stop without calling start first.");
     } else {
-        // This can only happen from the SCHEDULED_STATE or PLAYING_STATE. The UNSCHEDULED_STATE is
-        // handled above, and the FINISHED_STATE is only reachable after stop() has been called, and
-        // hence m_stopCalled is true. But that case is handled above.
+        // stop() can be called more than once, with the last call to stop taking effect, unless the
+        // source has already stopped due to earlier calls to stop. No exceptions are thrown in any
+        // case.
         when = max(0.0, when);
         m_endTime = when;
-        m_stopCalled = true;
     }
 }
 
-void AudioScheduledSourceNode::setOnended(PassRefPtr<EventListener> listener, DOMWrapperWorld* isolatedWorld)
+void AudioScheduledSourceNode::setOnended(PassRefPtr<EventListener> listener)
 {
     m_hasEndedListener = listener;
-    setAttributeEventListener(EventTypeNames::ended, listener, isolatedWorld);
+    setAttributeEventListener(EventTypeNames::ended, listener);
 }
 
 void AudioScheduledSourceNode::finish()
@@ -187,7 +179,6 @@ void AudioScheduledSourceNode::finish()
         // Let the context dereference this AudioNode.
         context()->notifyNodeFinishedProcessing(this);
         m_playbackState = FINISHED_STATE;
-        context()->decrementActiveSourceCount();
     }
 
     if (m_hasEndedListener) {
@@ -211,8 +202,8 @@ AudioScheduledSourceNode::NotifyEndedTask::NotifyEndedTask(PassRefPtr<AudioSched
 
 void AudioScheduledSourceNode::NotifyEndedTask::notifyEnded()
 {
-    RefPtr<Event> event = Event::create(EventTypeNames::ended);
-    event->setTarget(m_scheduledNode);
+    RefPtrWillBeRawPtr<Event> event = Event::create(EventTypeNames::ended);
+    event->setTarget(m_scheduledNode.get());
     m_scheduledNode->dispatchEvent(event.get());
 }
 
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioScheduledSourceNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/AudioScheduledSourceNode.h
index e89cd76bf0d..3942ecd3589 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioScheduledSourceNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioScheduledSourceNode.h
@@ -40,7 +40,7 @@ public:
     // These are the possible states an AudioScheduledSourceNode can be in:
     //
     // UNSCHEDULED_STATE - Initial playback state. Created, but not yet scheduled.
-    // SCHEDULED_STATE - Scheduled to play (via noteOn() or noteGrainOn()), but not yet playing.
+    // SCHEDULED_STATE - Scheduled to play (via start()), but not yet playing.
     // PLAYING_STATE - Generating sound.
     // FINISHED_STATE - Finished generating sound.
     //
@@ -77,8 +77,8 @@ public:
     bool isPlayingOrScheduled() const { return m_playbackState == PLAYING_STATE || m_playbackState == SCHEDULED_STATE; }
     bool hasFinished() const { return m_playbackState == FINISHED_STATE; }
 
-    EventListener* onended(DOMWrapperWorld* isolatedWorld) { return getAttributeEventListener(EventTypeNames::ended, isolatedWorld); }
-    void setOnended(PassRefPtr<EventListener>, DOMWrapperWorld* isolatedWorld = 0);
+    EventListener* onended() { return getAttributeEventListener(EventTypeNames::ended); }
+    void setOnended(PassRefPtr<EventListener>);
 
 protected:
     // Get frame information for the current time quantum.
@@ -109,7 +109,6 @@ protected:
     double m_endTime; // in seconds
 
     bool m_hasEndedListener;
-    bool m_stopCalled;
 
     static const double UnknownTime;
 };
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioSourceNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/AudioSourceNode.h
index b0883e10c5a..a8105b39993 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioSourceNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioSourceNode.h
@@ -41,8 +41,8 @@ public:
         ScriptWrappable::init(this);
     }
 protected:
-    virtual double tailTime() const OVERRIDE { return 0; }
-    virtual double latencyTime() const OVERRIDE { return 0; }
+    virtual double tailTime() const OVERRIDE FINAL { return 0; }
+    virtual double latencyTime() const OVERRIDE FINAL { return 0; }
 };
 
 } // namespace WebCore
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioSummingJunction.h b/chromium/third_party/WebKit/Source/modules/webaudio/AudioSummingJunction.h
index a2e789f04b2..ca21fe6cb02 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioSummingJunction.h
+++ b/chromium/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"
 
@@ -54,23 +55,18 @@ public:
     // Rendering code accesses its version of the current connections here.
     unsigned numberOfRenderingConnections() const { return m_renderingOutputs.size(); }
     AudioNodeOutput* renderingOutput(unsigned i) { return m_renderingOutputs[i]; }
-    const AudioNodeOutput* renderingOutput(unsigned i) const { return m_renderingOutputs[i]; }
     bool isConnected() const { return numberOfRenderingConnections() > 0; }
 
     virtual bool canUpdateState() = 0;
     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.
     HashSet<AudioNodeOutput*> m_outputs;
 
-    // numberOfConnections() should never be called from the audio rendering thread.
-    // Instead numberOfRenderingConnections() and renderingOutput() should be used.
-    unsigned numberOfConnections() const { return m_outputs.size(); }
-
     // m_renderingOutputs is a copy of m_outputs which will never be modified during the graph rendering on the audio thread.
     // This is the list which is used by the rendering code.
     // Whenever m_outputs is modified, the context is told so it can later update m_renderingOutputs from m_outputs at a safe time.
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/BiquadDSPKernel.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/BiquadDSPKernel.cpp
index 5136ae118cd..45b663be10b 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/BiquadDSPKernel.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/BiquadDSPKernel.cpp
@@ -40,73 +40,73 @@ namespace WebCore {
 // settings of the Biquad.
 static const double MaxBiquadDelayTime = 0.2;
 
-void BiquadDSPKernel::updateCoefficientsIfNecessary(bool useSmoothing, bool forceUpdate)
+void BiquadDSPKernel::updateCoefficientsIfNecessary()
 {
-    if (forceUpdate || biquadProcessor()->filterCoefficientsDirty()) {
-        double value1;
-        double value2;
+    if (biquadProcessor()->filterCoefficientsDirty()) {
+        double cutoffFrequency;
+        double Q;
         double gain;
         double detune; // in Cents
 
         if (biquadProcessor()->hasSampleAccurateValues()) {
-            value1 = biquadProcessor()->parameter1()->finalValue();
-            value2 = biquadProcessor()->parameter2()->finalValue();
+            cutoffFrequency = biquadProcessor()->parameter1()->finalValue();
+            Q = biquadProcessor()->parameter2()->finalValue();
             gain = biquadProcessor()->parameter3()->finalValue();
             detune = biquadProcessor()->parameter4()->finalValue();
-        } else if (useSmoothing) {
-            value1 = biquadProcessor()->parameter1()->smoothedValue();
-            value2 = biquadProcessor()->parameter2()->smoothedValue();
+        } else {
+            cutoffFrequency = biquadProcessor()->parameter1()->smoothedValue();
+            Q = biquadProcessor()->parameter2()->smoothedValue();
             gain = biquadProcessor()->parameter3()->smoothedValue();
             detune = biquadProcessor()->parameter4()->smoothedValue();
-        } else {
-            value1 = biquadProcessor()->parameter1()->value();
-            value2 = biquadProcessor()->parameter2()->value();
-            gain = biquadProcessor()->parameter3()->value();
-            detune = biquadProcessor()->parameter4()->value();
         }
 
-        // Convert from Hertz to normalized frequency 0 -> 1.
-        double nyquist = this->nyquist();
-        double normalizedFrequency = value1 / nyquist;
+        updateCoefficients(cutoffFrequency, Q, gain, detune);
+    }
+}
 
-        // Offset frequency by detune.
-        if (detune)
-            normalizedFrequency *= pow(2, detune / 1200);
+void BiquadDSPKernel::updateCoefficients(double cutoffFrequency, double Q, double gain, double detune)
+{
+    // Convert from Hertz to normalized frequency 0 -> 1.
+    double nyquist = this->nyquist();
+    double normalizedFrequency = cutoffFrequency / nyquist;
 
-        // Configure the biquad with the new filter parameters for the appropriate type of filter.
-        switch (biquadProcessor()->type()) {
-        case BiquadProcessor::LowPass:
-            m_biquad.setLowpassParams(normalizedFrequency, value2);
-            break;
+    // Offset frequency by detune.
+    if (detune)
+        normalizedFrequency *= pow(2, detune / 1200);
 
-        case BiquadProcessor::HighPass:
-            m_biquad.setHighpassParams(normalizedFrequency, value2);
-            break;
+    // Configure the biquad with the new filter parameters for the appropriate type of filter.
+    switch (biquadProcessor()->type()) {
+    case BiquadProcessor::LowPass:
+        m_biquad.setLowpassParams(normalizedFrequency, Q);
+        break;
 
-        case BiquadProcessor::BandPass:
-            m_biquad.setBandpassParams(normalizedFrequency, value2);
-            break;
+    case BiquadProcessor::HighPass:
+        m_biquad.setHighpassParams(normalizedFrequency, Q);
+        break;
 
-        case BiquadProcessor::LowShelf:
-            m_biquad.setLowShelfParams(normalizedFrequency, gain);
-            break;
+    case BiquadProcessor::BandPass:
+        m_biquad.setBandpassParams(normalizedFrequency, Q);
+        break;
 
-        case BiquadProcessor::HighShelf:
-            m_biquad.setHighShelfParams(normalizedFrequency, gain);
-            break;
+    case BiquadProcessor::LowShelf:
+        m_biquad.setLowShelfParams(normalizedFrequency, gain);
+        break;
 
-        case BiquadProcessor::Peaking:
-            m_biquad.setPeakingParams(normalizedFrequency, value2, gain);
-            break;
+    case BiquadProcessor::HighShelf:
+        m_biquad.setHighShelfParams(normalizedFrequency, gain);
+        break;
 
-        case BiquadProcessor::Notch:
-            m_biquad.setNotchParams(normalizedFrequency, value2);
-            break;
+    case BiquadProcessor::Peaking:
+        m_biquad.setPeakingParams(normalizedFrequency, Q, gain);
+        break;
 
-        case BiquadProcessor::Allpass:
-            m_biquad.setAllpassParams(normalizedFrequency, value2);
-            break;
-        }
+    case BiquadProcessor::Notch:
+        m_biquad.setNotchParams(normalizedFrequency, Q);
+        break;
+
+    case BiquadProcessor::Allpass:
+        m_biquad.setAllpassParams(normalizedFrequency, Q);
+        break;
     }
 }
 
@@ -118,7 +118,14 @@ void BiquadDSPKernel::process(const float* source, float* destination, size_t fr
     // FIXME: as an optimization, implement a way that a Biquad object can simply copy its internal filter coefficients from another Biquad object.
     // Then re-factor this code to only run for the first BiquadDSPKernel of each BiquadProcessor.
 
-    updateCoefficientsIfNecessary(true, false);
+
+    // The audio thread can't block on this lock; skip updating the coefficients for this block if
+    // necessary. We'll get them the next time around.
+    {
+        MutexTryLocker tryLocker(m_processLock);
+        if (tryLocker.locked())
+            updateCoefficientsIfNecessary();
+    }
 
     m_biquad.process(source, destination, framesToProcess);
 }
@@ -142,12 +149,33 @@ void BiquadDSPKernel::getFrequencyResponse(int nFrequencies,
     for (int k = 0; k < nFrequencies; ++k)
         frequency[k] = narrowPrecisionToFloat(frequencyHz[k] / nyquist);
 
-    // We want to get the final values of the coefficients and compute
-    // the response from that instead of some intermediate smoothed
-    // set. Forcefully update the coefficients even if they are not
-    // dirty.
+    double cutoffFrequency;
+    double Q;
+    double gain;
+    double detune; // in Cents
+
+    {
+        // Get a copy of the current biquad filter coefficients so we can update the biquad with
+        // these values. We need to synchronize with process() to prevent process() from updating
+        // the filter coefficients while we're trying to access them. The process will update it
+        // next time around.
+        //
+        // The BiquadDSPKernel object here (along with it's Biquad object) is for querying the
+        // frequency response and is NOT the same as the one in process() which is used for
+        // performing the actual filtering. This one is is created in
+        // BiquadProcessor::getFrequencyResponse for this purpose. Both, however, point to the same
+        // BiquadProcessor object.
+        //
+        // FIXME: Simplify this: crbug.com/390266
+        MutexLocker processLocker(m_processLock);
+
+        cutoffFrequency = biquadProcessor()->parameter1()->value();
+        Q = biquadProcessor()->parameter2()->value();
+        gain = biquadProcessor()->parameter3()->value();
+        detune = biquadProcessor()->parameter4()->value();
+    }
 
-    updateCoefficientsIfNecessary(false, true);
+    updateCoefficients(cutoffFrequency, Q, gain, detune);
 
     m_biquad.getFrequencyResponse(nFrequencies, frequency.data(), magResponse, phaseResponse);
 }
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/BiquadDSPKernel.h b/chromium/third_party/WebKit/Source/modules/webaudio/BiquadDSPKernel.h
index c2e8ac8cbb5..d0b12b332f0 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/BiquadDSPKernel.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/BiquadDSPKernel.h
@@ -35,7 +35,7 @@ class BiquadProcessor;
 
 // BiquadDSPKernel is an AudioDSPKernel and is responsible for filtering one channel of a BiquadProcessor using a Biquad object.
 
-class BiquadDSPKernel : public AudioDSPKernel {
+class BiquadDSPKernel FINAL : public AudioDSPKernel {
 public:
     explicit BiquadDSPKernel(BiquadProcessor* processor)
     : AudioDSPKernel(processor)
@@ -43,8 +43,8 @@ public:
     }
 
     // AudioDSPKernel
-    virtual void process(const float* source, float* dest, size_t framesToProcess);
-    virtual void reset() { m_biquad.reset(); }
+    virtual void process(const float* source, float* dest, size_t framesToProcess) OVERRIDE;
+    virtual void reset() OVERRIDE { m_biquad.reset(); }
 
     // Get the magnitude and phase response of the filter at the given
     // set of frequencies (in Hz). The phase response is in radians.
@@ -62,12 +62,13 @@ protected:
 
     // To prevent audio glitches when parameters are changed,
     // dezippering is used to slowly change the parameters.
-    // |useSmoothing| implies that we want to update using the
-    // smoothed values. Otherwise the final target values are
-    // used. If |forceUpdate| is true, we update the coefficients even
-    // if they are not dirty. (Used when computing the frequency
-    // response.)
-    void updateCoefficientsIfNecessary(bool useSmoothing, bool forceUpdate);
+    void updateCoefficientsIfNecessary();
+    // Update the biquad cofficients with the given parameters
+    void updateCoefficients(double frequency, double Q, double gain, double detune);
+
+private:
+    // Synchronize process() with getting and setting the filter coefficients.
+    mutable Mutex m_processLock;
 };
 
 } // namespace WebCore
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/BiquadFilterNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/BiquadFilterNode.cpp
index 1aa8f4a910a..7069d1f3f04 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/BiquadFilterNode.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/BiquadFilterNode.cpp
@@ -82,8 +82,6 @@ void BiquadFilterNode::setType(const String& type)
         setType(BiquadProcessor::Notch);
     else if (type == "allpass")
         setType(BiquadProcessor::Allpass);
-    else
-        ASSERT_NOT_REACHED();
 }
 
 bool BiquadFilterNode::setType(unsigned type)
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/BiquadFilterNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/BiquadFilterNode.h
index 0ba4681c304..94bd76e90d2 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/BiquadFilterNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/BiquadFilterNode.h
@@ -32,7 +32,7 @@ namespace WebCore {
 
 class AudioParam;
 
-class BiquadFilterNode : public AudioBasicProcessorNode {
+class BiquadFilterNode FINAL : public AudioBasicProcessorNode {
 public:
     // These must be defined as in the .idl file and must match those in the BiquadProcessor class.
     enum {
@@ -46,13 +46,12 @@ 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;
-    bool setType(unsigned); // Returns true on success.
     void setType(const String&);
 
     AudioParam* frequency() { return biquadProcessor()->parameter1(); }
@@ -70,6 +69,7 @@ private:
     BiquadFilterNode(AudioContext*, float sampleRate);
 
     BiquadProcessor* biquadProcessor() { return static_cast<BiquadProcessor*>(processor()); }
+    bool setType(unsigned); // Returns true on success.
 };
 
 } // namespace WebCore
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/BiquadFilterNode.idl b/chromium/third_party/WebKit/Source/modules/webaudio/BiquadFilterNode.idl
index 2422dfb1a98..9783ab2a22b 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/BiquadFilterNode.idl
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/BiquadFilterNode.idl
@@ -22,20 +22,21 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
+enum BiquadFilterType {
+    "lowpass",
+    "highpass",
+    "bandpass",
+    "lowshelf",
+    "highshelf",
+    "peaking",
+    "notch",
+    "allpass"
+};
+
 [
     Conditional=WEB_AUDIO
 ] interface BiquadFilterNode : AudioNode {
-    // Filter type.
-    const unsigned short LOWPASS = 0;
-    const unsigned short HIGHPASS = 1;
-    const unsigned short BANDPASS = 2;
-    const unsigned short LOWSHELF = 3;
-    const unsigned short HIGHSHELF = 4;
-    const unsigned short PEAKING = 5;
-    const unsigned short NOTCH = 6;
-    const unsigned short ALLPASS = 7;
-
-    [Custom=Setter] attribute DOMString type;
+    attribute BiquadFilterType type;
 
     readonly attribute AudioParam frequency; // in Hertz
     readonly attribute AudioParam detune; // in Cents
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/BiquadProcessor.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/BiquadProcessor.cpp
index 2655abeb215..f2410ce928d 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/BiquadProcessor.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/BiquadProcessor.cpp
@@ -35,10 +35,10 @@ namespace WebCore {
 BiquadProcessor::BiquadProcessor(AudioContext* context, float sampleRate, size_t numberOfChannels, bool autoInitialize)
     : AudioDSPKernelProcessor(sampleRate, numberOfChannels)
     , m_type(LowPass)
-    , m_parameter1(0)
-    , m_parameter2(0)
-    , m_parameter3(0)
-    , m_parameter4(0)
+    , m_parameter1(nullptr)
+    , m_parameter2(nullptr)
+    , m_parameter3(nullptr)
+    , m_parameter4(nullptr)
     , m_filterCoefficientsDirty(true)
     , m_hasSampleAccurateValues(false)
 {
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/BiquadProcessor.h b/chromium/third_party/WebKit/Source/modules/webaudio/BiquadProcessor.h
index 6c3e3307094..7647586c36e 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/BiquadProcessor.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/BiquadProcessor.h
@@ -36,7 +36,7 @@ namespace WebCore {
 
 // BiquadProcessor is an AudioDSPKernelProcessor which uses Biquad objects to implement several common filters.
 
-class BiquadProcessor : public AudioDSPKernelProcessor {
+class BiquadProcessor FINAL : public AudioDSPKernelProcessor {
 public:
     enum FilterType {
         LowPass = 0,
@@ -53,9 +53,9 @@ public:
 
     virtual ~BiquadProcessor();
 
-    virtual PassOwnPtr<AudioDSPKernel> createKernel();
+    virtual PassOwnPtr<AudioDSPKernel> createKernel() OVERRIDE;
 
-    virtual void process(const AudioBus* source, AudioBus* destination, size_t framesToProcess);
+    virtual void process(const AudioBus* source, AudioBus* destination, size_t framesToProcess) OVERRIDE;
 
     // Get the magnitude and phase response of the filter at the given
     // set of frequencies (in Hz). The phase response is in radians.
@@ -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;
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/ChannelMergerNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/ChannelMergerNode.cpp
index 43b86b4d389..96befdeb438 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/ChannelMergerNode.cpp
+++ b/chromium/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 0;
+        return nullptr;
 
-    return adoptRef(new ChannelMergerNode(context, sampleRate, numberOfInputs));
+    return adoptRefWillBeNoop(new ChannelMergerNode(context, sampleRate, numberOfInputs));
 }
 
 ChannelMergerNode::ChannelMergerNode(AudioContext* context, float sampleRate, unsigned numberOfInputs)
@@ -76,29 +76,33 @@ void ChannelMergerNode::process(size_t framesToProcess)
 
     // Merge all the channels from all the inputs into one output.
     unsigned outputChannelIndex = 0;
+    unsigned maxAllowedOutputChannels = output->numberOfChannels();
+
     for (unsigned i = 0; i < numberOfInputs(); ++i) {
         AudioNodeInput* input = this->input(i);
         if (input->isConnected()) {
             unsigned numberOfInputChannels = input->bus()->numberOfChannels();
 
-            // Merge channels from this particular input.
+            // Merge channels from this particular input, but be careful not to exceed the number of
+            // output channels.  (This can happen if there are many inputs with each input
+            // containing many channels.)
             for (unsigned j = 0; j < numberOfInputChannels; ++j) {
-                AudioChannel* inputChannel = input->bus()->channel(j);
-                AudioChannel* outputChannel = output->bus()->channel(outputChannelIndex);
-                outputChannel->copyFrom(inputChannel);
+                if (outputChannelIndex < maxAllowedOutputChannels) {
+                    AudioChannel* inputChannel = input->bus()->channel(j);
+                    AudioChannel* outputChannel = output->bus()->channel(outputChannelIndex);
+                    outputChannel->copyFrom(inputChannel);
 
-                ++outputChannelIndex;
+                    ++outputChannelIndex;
+                }
             }
         }
+        if (outputChannelIndex >= maxAllowedOutputChannels)
+            break;
     }
 
     ASSERT(outputChannelIndex == output->numberOfChannels());
 }
 
-void ChannelMergerNode::reset()
-{
-}
-
 // Any time a connection or disconnection happens on any of our inputs, we potentially need to change the
 // number of channels of our output.
 void ChannelMergerNode::checkNumberOfChannelsForInput(AudioNodeInput* input)
@@ -113,6 +117,9 @@ void ChannelMergerNode::checkNumberOfChannelsForInput(AudioNodeInput* input)
             numberOfOutputChannels += input->numberOfChannels();
     }
 
+    // If the actual number of channels exceeds the max allowed, just drop the excess.
+    numberOfOutputChannels = std::min(numberOfOutputChannels, AudioContext::maxNumberOfChannels());
+
     // Set the correct number of channels on the output
     AudioNodeOutput* output = this->output(0);
     ASSERT(output);
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/ChannelMergerNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/ChannelMergerNode.h
index a15b6fa5122..ea62eff3175 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/ChannelMergerNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/ChannelMergerNode.h
@@ -36,16 +36,15 @@ namespace WebCore {
 
 class AudioContext;
 
-class ChannelMergerNode : public AudioNode {
+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);
-    virtual void reset();
+    virtual void process(size_t framesToProcess) OVERRIDE;
 
     // Called in the audio thread (pre-rendering task) when the number of channels for an input may have changed.
-    virtual void checkNumberOfChannelsForInput(AudioNodeInput*);
+    virtual void checkNumberOfChannelsForInput(AudioNodeInput*) OVERRIDE;
 
 private:
     unsigned m_desiredNumberOfOutputChannels;
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/ChannelSplitterNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/ChannelSplitterNode.cpp
index 4009ba49373..18a5e1a6848 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/ChannelSplitterNode.cpp
+++ b/chromium/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 0;
+        return nullptr;
 
-    return adoptRef(new ChannelSplitterNode(context, sampleRate, numberOfOutputs));
+    return adoptRefWillBeNoop(new ChannelSplitterNode(context, sampleRate, numberOfOutputs));
 }
 
 ChannelSplitterNode::ChannelSplitterNode(AudioContext* context, float sampleRate, unsigned numberOfOutputs)
@@ -79,10 +79,6 @@ void ChannelSplitterNode::process(size_t framesToProcess)
     }
 }
 
-void ChannelSplitterNode::reset()
-{
-}
-
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/ChannelSplitterNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/ChannelSplitterNode.h
index c3c4a668e6a..32bdbd78833 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/ChannelSplitterNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/ChannelSplitterNode.h
@@ -32,13 +32,12 @@ namespace WebCore {
 
 class AudioContext;
 
-class ChannelSplitterNode : public AudioNode {
+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);
-    virtual void reset();
+    virtual void process(size_t framesToProcess) OVERRIDE;
 
 private:
     virtual double tailTime() const OVERRIDE { return 0; }
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/ConvolverNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/ConvolverNode.cpp
index cef9bca9c78..a0eac4de379 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/ConvolverNode.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/ConvolverNode.cpp
@@ -28,6 +28,8 @@
 
 #include "modules/webaudio/ConvolverNode.h"
 
+#include "bindings/v8/ExceptionState.h"
+#include "core/dom/ExceptionCode.h"
 #include "platform/audio/Reverb.h"
 #include "modules/webaudio/AudioBuffer.h"
 #include "modules/webaudio/AudioContext.h"
@@ -90,13 +92,6 @@ void ConvolverNode::process(size_t framesToProcess)
     }
 }
 
-void ConvolverNode::reset()
-{
-    MutexLocker locker(m_processLock);
-    if (m_reverb.get())
-        m_reverb->reset();
-}
-
 void ConvolverNode::initialize()
 {
     if (isInitialized())
@@ -114,13 +109,21 @@ void ConvolverNode::uninitialize()
     AudioNode::uninitialize();
 }
 
-void ConvolverNode::setBuffer(AudioBuffer* buffer)
+void ConvolverNode::setBuffer(AudioBuffer* buffer, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
 
     if (!buffer)
         return;
 
+    if (buffer->sampleRate() != context()->sampleRate()) {
+        exceptionState.throwDOMException(
+            NotSupportedError,
+            "The buffer sample rate of " + String::number(buffer->sampleRate())
+            + " does not match the context rate of " + String::number(context()->sampleRate())
+            + " Hz.");
+    }
+
     unsigned numberOfChannels = buffer->numberOfChannels();
     size_t bufferLength = buffer->length();
 
@@ -176,6 +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)
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/ConvolverNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/ConvolverNode.h
index e5e33e62691..02ad70b3a8f 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/ConvolverNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/ConvolverNode.h
@@ -33,30 +33,32 @@
 namespace WebCore {
 
 class AudioBuffer;
+class ExceptionState;
 class Reverb;
 
-class ConvolverNode : public AudioNode {
+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();
 
     // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void reset();
-    virtual void initialize();
-    virtual void uninitialize();
+    virtual void process(size_t framesToProcess) OVERRIDE;
+    virtual void initialize() OVERRIDE;
+    virtual void uninitialize() OVERRIDE;
 
     // Impulse responses
-    void setBuffer(AudioBuffer*);
+    void setBuffer(AudioBuffer*, ExceptionState&);
     AudioBuffer* buffer();
 
     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;
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/ConvolverNode.idl b/chromium/third_party/WebKit/Source/modules/webaudio/ConvolverNode.idl
index f0339170133..fe0544b33eb 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/ConvolverNode.idl
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/ConvolverNode.idl
@@ -26,6 +26,6 @@
 [
     Conditional=WEB_AUDIO
 ] interface ConvolverNode : AudioNode {
-    attribute AudioBuffer buffer;
+    [RaisesException=Setter] attribute AudioBuffer buffer;
     attribute boolean normalize;
 };
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/DOMWindowWebAudio.h b/chromium/third_party/WebKit/Source/modules/webaudio/DOMWindowWebAudio.h
new file mode 100644
index 00000000000..03f01d5c617
--- /dev/null
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/DOMWindowWebAudio.h
@@ -0,0 +1,7 @@
+// Copyright (c) 2014 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+// TODO: This is a dummy header file required by the generated binding code.
+// This file should be removed after fixing the code generator.
+
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/DefaultAudioDestinationNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/DefaultAudioDestinationNode.cpp
index a6b93bf28d8..50f1df3254d 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/DefaultAudioDestinationNode.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/DefaultAudioDestinationNode.cpp
@@ -34,8 +34,6 @@
 #include "platform/Logging.h"
 #include "wtf/MainThread.h"
 
-const unsigned EnabledInputChannels = 2;
-
 namespace WebCore {
 
 DefaultAudioDestinationNode::DefaultAudioDestinationNode(AudioContext* context)
@@ -83,22 +81,6 @@ void DefaultAudioDestinationNode::createDestination()
     m_destination = AudioDestination::create(*this, m_inputDeviceId, m_numberOfInputChannels, channelCount(), hardwareSampleRate);
 }
 
-void DefaultAudioDestinationNode::enableInput(const String& inputDeviceId)
-{
-    ASSERT(isMainThread());
-    if (m_numberOfInputChannels != EnabledInputChannels) {
-        m_numberOfInputChannels = EnabledInputChannels;
-        m_inputDeviceId = inputDeviceId;
-
-        if (isInitialized()) {
-            // Re-create destination.
-            m_destination->stop();
-            createDestination();
-            m_destination->start();
-        }
-    }
-}
-
 void DefaultAudioDestinationNode::startRendering()
 {
     ASSERT(isInitialized());
@@ -122,12 +104,7 @@ void DefaultAudioDestinationNode::setChannelCount(unsigned long channelCount, Ex
     if (!maxChannelCount() || channelCount > maxChannelCount()) {
         exceptionState.throwDOMException(
             IndexSizeError,
-            ExceptionMessages::failedToSet(
-                "channelCount",
-                "AudioDestinationNode",
-                "channel count (" + String::number(channelCount)
-                + ") must be between 1 and "
-                + String::number(maxChannelCount()) + "."));
+            ExceptionMessages::indexOutsideRange<unsigned>("channel count", channelCount, 1, ExceptionMessages::InclusiveBound, maxChannelCount(), ExceptionMessages::InclusiveBound));
         return;
     }
 
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/DefaultAudioDestinationNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/DefaultAudioDestinationNode.h
index 8c0506d221e..409d31c4c50 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/DefaultAudioDestinationNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/DefaultAudioDestinationNode.h
@@ -34,11 +34,11 @@ namespace WebCore {
 class AudioContext;
 class ExceptionState;
 
-class DefaultAudioDestinationNode : public AudioDestinationNode {
+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();
@@ -49,7 +49,6 @@ public:
     virtual void setChannelCount(unsigned long, ExceptionState&) OVERRIDE;
 
     // AudioDestinationNode
-    virtual void enableInput(const String& inputDeviceId) OVERRIDE;
     virtual void startRendering() OVERRIDE;
     virtual unsigned long maxChannelCount() const OVERRIDE;
 
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/DelayDSPKernel.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/DelayDSPKernel.cpp
index 05b8cc566fd..2301325f95b 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/DelayDSPKernel.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/DelayDSPKernel.cpp
@@ -29,6 +29,7 @@
 #include "modules/webaudio/DelayDSPKernel.h"
 
 #include "platform/audio/AudioUtilities.h"
+#include "wtf/MathExtras.h"
 #include <algorithm>
 
 using namespace std;
@@ -45,8 +46,8 @@ DelayDSPKernel::DelayDSPKernel(DelayProcessor* processor)
         return;
 
     m_maxDelayTime = processor->maxDelayTime();
-    ASSERT(m_maxDelayTime >= 0);
-    if (m_maxDelayTime < 0)
+    ASSERT(m_maxDelayTime >= 0 && !std::isnan(m_maxDelayTime));
+    if (m_maxDelayTime < 0 || std::isnan(m_maxDelayTime))
         return;
 
     m_buffer.allocate(bufferLengthForDelay(m_maxDelayTime, processor->sampleRate()));
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/DelayDSPKernel.h b/chromium/third_party/WebKit/Source/modules/webaudio/DelayDSPKernel.h
index 320350ebb5c..a0fbd661015 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/DelayDSPKernel.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/DelayDSPKernel.h
@@ -32,7 +32,7 @@ namespace WebCore {
 
 class DelayProcessor;
 
-class DelayDSPKernel : public AudioDelayDSPKernel {
+class DelayDSPKernel FINAL : public AudioDelayDSPKernel {
 public:
     explicit DelayDSPKernel(DelayProcessor*);
 
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/DelayNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/DelayNode.cpp
index 7a8817ee82e..55c6337635e 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/DelayNode.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/DelayNode.cpp
@@ -31,6 +31,7 @@
 #include "bindings/v8/ExceptionMessages.h"
 #include "bindings/v8/ExceptionState.h"
 #include "core/dom/ExceptionCode.h"
+#include "wtf/MathExtras.h"
 
 namespace WebCore {
 
@@ -40,7 +41,7 @@ DelayNode::DelayNode(AudioContext* context, float sampleRate, double maxDelayTim
     : AudioBasicProcessorNode(context, sampleRate)
 {
     ScriptWrappable::init(this);
-    if (maxDelayTime <= 0 || maxDelayTime >= maximumAllowedDelayTime) {
+    if (maxDelayTime <= 0 || maxDelayTime >= maximumAllowedDelayTime || std::isnan(maxDelayTime)) {
         exceptionState.throwDOMException(
             NotSupportedError,
             "max delay time (" + String::number(maxDelayTime)
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/DelayNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/DelayNode.h
index 14ee7e4deab..8324d73d2e9 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/DelayNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/DelayNode.h
@@ -34,11 +34,11 @@ namespace WebCore {
 class AudioParam;
 class ExceptionState;
 
-class DelayNode : public AudioBasicProcessorNode {
+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();
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/DelayProcessor.h b/chromium/third_party/WebKit/Source/modules/webaudio/DelayProcessor.h
index 779aa7682e6..63b8af26cb7 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/DelayProcessor.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/DelayProcessor.h
@@ -34,19 +34,19 @@ namespace WebCore {
 
 class AudioDSPKernel;
 
-class DelayProcessor : public AudioDSPKernelProcessor {
+class DelayProcessor FINAL : public AudioDSPKernelProcessor {
 public:
     DelayProcessor(AudioContext*, float sampleRate, unsigned numberOfChannels, double maxDelayTime);
     virtual ~DelayProcessor();
 
-    virtual PassOwnPtr<AudioDSPKernel> createKernel();
+    virtual PassOwnPtr<AudioDSPKernel> createKernel() OVERRIDE;
 
     AudioParam* delayTime() const { return m_delayTime.get(); }
 
     double maxDelayTime() { return m_maxDelayTime; }
 private:
 
-    RefPtr<AudioParam> m_delayTime;
+    RefPtrWillBePersistent<AudioParam> m_delayTime;
     double m_maxDelayTime;
 };
 
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/DynamicsCompressorNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/DynamicsCompressorNode.cpp
index 6700897e469..476e8db5dab 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/DynamicsCompressorNode.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/DynamicsCompressorNode.cpp
@@ -85,11 +85,6 @@ void DynamicsCompressorNode::process(size_t framesToProcess)
     m_reduction->setValue(reduction);
 }
 
-void DynamicsCompressorNode::reset()
-{
-    m_dynamicsCompressor->reset();
-}
-
 void DynamicsCompressorNode::initialize()
 {
     if (isInitialized())
@@ -118,6 +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)
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/DynamicsCompressorNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/DynamicsCompressorNode.h
index 413c0b08e26..068c6696fdb 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/DynamicsCompressorNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/DynamicsCompressorNode.h
@@ -33,20 +33,19 @@ namespace WebCore {
 
 class DynamicsCompressor;
 
-class DynamicsCompressorNode : public AudioNode {
+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();
 
     // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void reset();
-    virtual void initialize();
-    virtual void uninitialize();
+    virtual void process(size_t framesToProcess) OVERRIDE;
+    virtual void initialize() OVERRIDE;
+    virtual void uninitialize() OVERRIDE;
 
     // Static compression curve parameters.
     AudioParam* threshold() { return m_threshold.get(); }
@@ -58,6 +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;
@@ -65,12 +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
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/GainNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/GainNode.cpp
index 22a4c8377bc..03156476ef3 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/GainNode.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/GainNode.cpp
@@ -79,12 +79,6 @@ void GainNode::process(size_t framesToProcess)
     }
 }
 
-void GainNode::reset()
-{
-    // Snap directly to desired gain.
-    m_lastGain = gain()->value();
-}
-
 // FIXME: this can go away when we do mixing with gain directly in summing junction of AudioNodeInput
 //
 // As soon as we know the channel count of our input, we can lazily initialize.
@@ -114,6 +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)
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/GainNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/GainNode.h
index 361f581a040..a0886b918e8 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/GainNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/GainNode.h
@@ -37,23 +37,24 @@ class AudioContext;
 // GainNode is an AudioNode with one input and one output which applies a gain (volume) change to the audio signal.
 // De-zippering (smoothing) is applied when the gain value is changed dynamically.
 
-class GainNode : public AudioNode {
+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
-    virtual void process(size_t framesToProcess);
-    virtual void reset();
+    virtual void process(size_t framesToProcess) OVERRIDE;
 
     // Called in the main thread when the number of channels for the input may have changed.
-    virtual void checkNumberOfChannelsForInput(AudioNodeInput*);
+    virtual void checkNumberOfChannelsForInput(AudioNodeInput*) OVERRIDE;
 
     // 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; }
@@ -61,7 +62,7 @@ private:
     GainNode(AudioContext*, float sampleRate);
 
     float m_lastGain; // for de-zippering
-    RefPtr<AudioParam> m_gain;
+    RefPtrWillBeMember<AudioParam> m_gain;
 
     AudioFloatArray m_sampleAccurateGainValues;
 };
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/MediaElementAudioSourceNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/MediaElementAudioSourceNode.cpp
index 50b574ab9cd..ac3f7443b93 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/MediaElementAudioSourceNode.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/MediaElementAudioSourceNode.cpp
@@ -28,6 +28,7 @@
 
 #include "modules/webaudio/MediaElementAudioSourceNode.h"
 
+#include "core/html/HTMLMediaElement.h"
 #include "modules/webaudio/AudioContext.h"
 #include "modules/webaudio/AudioNodeOutput.h"
 #include "platform/Logging.h"
@@ -40,9 +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)
@@ -62,7 +63,9 @@ MediaElementAudioSourceNode::MediaElementAudioSourceNode(AudioContext* context,
 
 MediaElementAudioSourceNode::~MediaElementAudioSourceNode()
 {
+#if !ENABLE(OILPAN)
     m_mediaElement->setAudioSourceNode(0);
+#endif
     uninitialize();
 }
 
@@ -135,10 +138,6 @@ void MediaElementAudioSourceNode::process(size_t numberOfFrames)
     }
 }
 
-void MediaElementAudioSourceNode::reset()
-{
-}
-
 void MediaElementAudioSourceNode::lock()
 {
     ref();
@@ -151,6 +150,13 @@ void MediaElementAudioSourceNode::unlock()
     deref();
 }
 
+void MediaElementAudioSourceNode::trace(Visitor* visitor)
+{
+    visitor->trace(m_mediaElement);
+    AudioSourceNode::trace(visitor);
+    AudioSourceProviderClient::trace(visitor);
+}
+
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/MediaElementAudioSourceNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/MediaElementAudioSourceNode.h
index 27ae39272e3..c5888027f3c 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/MediaElementAudioSourceNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/MediaElementAudioSourceNode.h
@@ -27,7 +27,6 @@
 
 #if ENABLE(WEB_AUDIO)
 
-#include "core/html/HTMLMediaElement.h"
 #include "platform/audio/AudioSourceProviderClient.h"
 #include "platform/audio/MultiChannelResampler.h"
 #include "modules/webaudio/AudioSourceNode.h"
@@ -38,24 +37,27 @@
 namespace WebCore {
 
 class AudioContext;
+class HTMLMediaElement;
 
-class MediaElementAudioSourceNode : public AudioSourceNode, public AudioSourceProviderClient {
+class MediaElementAudioSourceNode FINAL : public AudioSourceNode, public AudioSourceProviderClient {
+    WILL_BE_USING_GARBAGE_COLLECTED_MIXIN(MediaElementAudioSourceNode);
 public:
-    static PassRefPtr<MediaElementAudioSourceNode> create(AudioContext*, HTMLMediaElement*);
+    static PassRefPtrWillBeRawPtr<MediaElementAudioSourceNode> create(AudioContext*, HTMLMediaElement*);
 
     virtual ~MediaElementAudioSourceNode();
 
     HTMLMediaElement* mediaElement() { return m_mediaElement.get(); }
 
     // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void reset();
+    virtual void process(size_t framesToProcess) OVERRIDE;
 
     // AudioSourceProviderClient
-    virtual void setFormat(size_t numberOfChannels, float sampleRate);
+    virtual void setFormat(size_t numberOfChannels, float sampleRate) OVERRIDE;
 
-    void lock();
-    void unlock();
+    virtual void lock() OVERRIDE;
+    virtual void unlock() OVERRIDE;
+
+    virtual void trace(Visitor*) OVERRIDE;
 
 private:
     MediaElementAudioSourceNode(AudioContext*, HTMLMediaElement*);
@@ -63,7 +65,7 @@ private:
     // As an audio source, we will never propagate silence.
     virtual bool propagatesSilence() const OVERRIDE { return false; }
 
-    RefPtr<HTMLMediaElement> m_mediaElement;
+    RefPtrWillBeMember<HTMLMediaElement> m_mediaElement;
     Mutex m_processLock;
 
     unsigned m_sourceNumberOfChannels;
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/MediaStreamAudioDestinationNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/MediaStreamAudioDestinationNode.cpp
index 34d883a0d42..0e26912194d 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/MediaStreamAudioDestinationNode.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/MediaStreamAudioDestinationNode.cpp
@@ -28,18 +28,18 @@
 
 #include "modules/webaudio/MediaStreamAudioDestinationNode.h"
 
-#include "core/platform/mediastream/MediaStreamCenter.h"
-#include "core/platform/mediastream/RTCPeerConnectionHandler.h"
 #include "modules/webaudio/AudioContext.h"
 #include "modules/webaudio/AudioNodeInput.h"
 #include "platform/UUID.h"
+#include "platform/mediastream/MediaStreamCenter.h"
+#include "public/platform/WebRTCPeerConnectionHandler.h"
 #include "wtf/Locker.h"
 
 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)
@@ -54,18 +54,13 @@ MediaStreamAudioDestinationNode::MediaStreamAudioDestinationNode(AudioContext* c
     audioSources.append(m_source);
     MediaStreamSourceVector videoSources;
     m_stream = MediaStream::create(context->executionContext(), MediaStreamDescriptor::create(audioSources, videoSources));
-    MediaStreamCenter::instance().didCreateMediaStream(m_stream->descriptor());
+    MediaStreamCenter::instance().didCreateMediaStreamAndTracks(m_stream->descriptor());
 
     m_source->setAudioFormat(numberOfChannels, context->sampleRate());
 
     initialize();
 }
 
-MediaStreamSource* MediaStreamAudioDestinationNode::mediaStreamSource()
-{
-    return m_source.get();
-}
-
 MediaStreamAudioDestinationNode::~MediaStreamAudioDestinationNode()
 {
     uninitialize();
@@ -77,10 +72,6 @@ void MediaStreamAudioDestinationNode::process(size_t numberOfFrames)
     m_source->consumeAudio(m_mixBus.get(), numberOfFrames);
 }
 
-void MediaStreamAudioDestinationNode::reset()
-{
-}
-
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/MediaStreamAudioDestinationNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/MediaStreamAudioDestinationNode.h
index 7ac3862f7c0..ef8f39340d8 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/MediaStreamAudioDestinationNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/MediaStreamAudioDestinationNode.h
@@ -37,19 +37,16 @@ namespace WebCore {
 
 class AudioContext;
 
-class MediaStreamAudioDestinationNode : public AudioBasicInspectorNode {
+class MediaStreamAudioDestinationNode FINAL : public AudioBasicInspectorNode {
 public:
-    static PassRefPtr<MediaStreamAudioDestinationNode> create(AudioContext*, size_t numberOfChannels);
+    static PassRefPtrWillBeRawPtr<MediaStreamAudioDestinationNode> create(AudioContext*, size_t numberOfChannels);
 
     virtual ~MediaStreamAudioDestinationNode();
 
     MediaStream* stream() { return m_stream.get(); }
 
     // AudioNode.
-    virtual void process(size_t framesToProcess);
-    virtual void reset();
-
-    MediaStreamSource* mediaStreamSource();
+    virtual void process(size_t framesToProcess) OVERRIDE;
 
 private:
     MediaStreamAudioDestinationNode(AudioContext*, size_t numberOfChannels);
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/MediaStreamAudioSourceNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/MediaStreamAudioSourceNode.cpp
index 58b91678423..2664e68c5e9 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/MediaStreamAudioSourceNode.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/MediaStreamAudioSourceNode.cpp
@@ -35,12 +35,12 @@
 
 namespace WebCore {
 
-PassRefPtr<MediaStreamAudioSourceNode> MediaStreamAudioSourceNode::create(AudioContext* context, MediaStream* mediaStream, MediaStreamTrack* audioTrack, 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, AudioSourceProvider* audioSourceProvider)
+MediaStreamAudioSourceNode::MediaStreamAudioSourceNode(AudioContext* context, MediaStream* mediaStream, MediaStreamTrack* audioTrack, PassOwnPtr<AudioSourceProvider> audioSourceProvider)
     : AudioSourceNode(context, context->sampleRate())
     , m_mediaStream(mediaStream)
     , m_audioTrack(audioTrack)
@@ -113,8 +113,12 @@ void MediaStreamAudioSourceNode::process(size_t numberOfFrames)
     }
 }
 
-void MediaStreamAudioSourceNode::reset()
+void MediaStreamAudioSourceNode::trace(Visitor* visitor)
 {
+    visitor->trace(m_mediaStream);
+    visitor->trace(m_audioTrack);
+    AudioSourceNode::trace(visitor);
+    AudioSourceProviderClient::trace(visitor);
 }
 
 } // namespace WebCore
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/MediaStreamAudioSourceNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/MediaStreamAudioSourceNode.h
index 625690bc5e1..42f3f43f2fd 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/MediaStreamAudioSourceNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/MediaStreamAudioSourceNode.h
@@ -39,32 +39,34 @@ namespace WebCore {
 
 class AudioContext;
 
-class MediaStreamAudioSourceNode : public AudioSourceNode, public AudioSourceProviderClient {
+class MediaStreamAudioSourceNode FINAL : public AudioSourceNode, public AudioSourceProviderClient {
+    WILL_BE_USING_GARBAGE_COLLECTED_MIXIN(MediaStreamAudioSourceNode);
 public:
-    static PassRefPtr<MediaStreamAudioSourceNode> create(AudioContext*, MediaStream*, MediaStreamTrack*, AudioSourceProvider*);
+    static PassRefPtrWillBeRawPtr<MediaStreamAudioSourceNode> create(AudioContext*, MediaStream*, MediaStreamTrack*, PassOwnPtr<AudioSourceProvider>);
 
     virtual ~MediaStreamAudioSourceNode();
 
     MediaStream* mediaStream() { return m_mediaStream.get(); }
 
     // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void reset();
+    virtual void process(size_t framesToProcess) OVERRIDE;
 
     // AudioSourceProviderClient
-    virtual void setFormat(size_t numberOfChannels, float sampleRate);
+    virtual void setFormat(size_t numberOfChannels, float sampleRate) OVERRIDE;
 
-    AudioSourceProvider* audioSourceProvider() const { return m_audioSourceProvider; }
+    AudioSourceProvider* audioSourceProvider() const { return m_audioSourceProvider.get(); }
+
+    virtual void trace(Visitor*) OVERRIDE;
 
 private:
-    MediaStreamAudioSourceNode(AudioContext*, MediaStream*, MediaStreamTrack*, AudioSourceProvider*);
+    MediaStreamAudioSourceNode(AudioContext*, MediaStream*, MediaStreamTrack*, PassOwnPtr<AudioSourceProvider>);
 
     // As an audio source, we will never propagate silence.
     virtual bool propagatesSilence() const OVERRIDE { return false; }
 
-    RefPtr<MediaStream> m_mediaStream;
-    RefPtr<MediaStreamTrack> m_audioTrack;
-    AudioSourceProvider* m_audioSourceProvider;
+    RefPtrWillBeMember<MediaStream> m_mediaStream;
+    RefPtrWillBeMember<MediaStreamTrack> m_audioTrack;
+    OwnPtr<AudioSourceProvider> m_audioSourceProvider;
 
     Mutex m_processLock;
 
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/OWNERS b/chromium/third_party/WebKit/Source/modules/webaudio/OWNERS
index d591a3eaa09..3cd6baeb1da 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/OWNERS
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/OWNERS
@@ -1 +1,2 @@
 kbr@chromium.org
+rtoy@chromium.org
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioCompletionEvent.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioCompletionEvent.cpp
index 45f73e94e73..a2a72d1e27b 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioCompletionEvent.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioCompletionEvent.cpp
@@ -28,18 +28,16 @@
 
 #include "modules/webaudio/OfflineAudioCompletionEvent.h"
 
-#include "core/events/ThreadLocalEventNames.h"
-
 namespace WebCore {
 
-PassRefPtr<OfflineAudioCompletionEvent> OfflineAudioCompletionEvent::create()
+PassRefPtrWillBeRawPtr<OfflineAudioCompletionEvent> OfflineAudioCompletionEvent::create()
 {
-    return adoptRef(new OfflineAudioCompletionEvent);
+    return adoptRefWillBeNoop(new OfflineAudioCompletionEvent);
 }
 
-PassRefPtr<OfflineAudioCompletionEvent> OfflineAudioCompletionEvent::create(PassRefPtr<AudioBuffer> renderedBuffer)
+PassRefPtrWillBeRawPtr<OfflineAudioCompletionEvent> OfflineAudioCompletionEvent::create(PassRefPtrWillBeRawPtr<AudioBuffer> renderedBuffer)
 {
-    return adoptRef(new OfflineAudioCompletionEvent(renderedBuffer));
+    return adoptRefWillBeNoop(new OfflineAudioCompletionEvent(renderedBuffer));
 }
 
 OfflineAudioCompletionEvent::OfflineAudioCompletionEvent()
@@ -47,7 +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 +61,12 @@ const AtomicString& OfflineAudioCompletionEvent::interfaceName() const
     return EventNames::OfflineAudioCompletionEvent;
 }
 
+void OfflineAudioCompletionEvent::trace(Visitor* visitor)
+{
+    visitor->trace(m_renderedBuffer);
+    Event::trace(visitor);
+}
+
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioCompletionEvent.h b/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioCompletionEvent.h
index c994c628a71..da5b8582a1d 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioCompletionEvent.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioCompletionEvent.h
@@ -25,7 +25,7 @@
 #ifndef OfflineAudioCompletionEvent_h
 #define OfflineAudioCompletionEvent_h
 
-#include "core/events/Event.h"
+#include "modules/EventModules.h"
 #include "modules/webaudio/AudioBuffer.h"
 #include "wtf/PassRefPtr.h"
 #include "wtf/RefPtr.h"
@@ -34,22 +34,24 @@ namespace WebCore {
 
 class AudioBuffer;
 
-class OfflineAudioCompletionEvent : public Event {
+class OfflineAudioCompletionEvent FINAL : public Event {
 public:
-    static PassRefPtr<OfflineAudioCompletionEvent> create();
-    static PassRefPtr<OfflineAudioCompletionEvent> create(PassRefPtr<AudioBuffer> renderedBuffer);
+    static PassRefPtrWillBeRawPtr<OfflineAudioCompletionEvent> create();
+    static PassRefPtrWillBeRawPtr<OfflineAudioCompletionEvent> create(PassRefPtrWillBeRawPtr<AudioBuffer> renderedBuffer);
 
     virtual ~OfflineAudioCompletionEvent();
 
     AudioBuffer* renderedBuffer() { return m_renderedBuffer.get(); }
 
-    virtual const AtomicString& interfaceName() const;
+    virtual const AtomicString& interfaceName() const OVERRIDE;
+
+    virtual void trace(Visitor*) OVERRIDE;
 
 private:
     OfflineAudioCompletionEvent();
-    explicit OfflineAudioCompletionEvent(PassRefPtr<AudioBuffer> renderedBuffer);
+    explicit OfflineAudioCompletionEvent(PassRefPtrWillBeRawPtr<AudioBuffer> renderedBuffer);
 
-    RefPtr<AudioBuffer> m_renderedBuffer;
+    RefPtrWillBeMember<AudioBuffer> m_renderedBuffer;
 };
 
 } // namespace WebCore
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioContext.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioContext.cpp
index 4cf7947dfe1..bed82c16d9a 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioContext.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioContext.cpp
@@ -36,34 +36,52 @@
 
 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()) {
         exceptionState.throwDOMException(
             NotSupportedError,
             "Workers are not supported.");
-        return 0;
+        return nullptr;
     }
 
     Document* document = toDocument(context);
 
     if (!numberOfFrames) {
         exceptionState.throwDOMException(SyntaxError, "number of frames cannot be zero.");
-        return 0;
+        return nullptr;
     }
 
-    if (numberOfChannels > 10) {
-        exceptionState.throwDOMException(SyntaxError, "number of channels (" + String::number(numberOfChannels) + ") exceeds maximum (10).");
-        return 0;
+    if (numberOfChannels > AudioContext::maxNumberOfChannels()) {
+        exceptionState.throwDOMException(
+            IndexSizeError,
+            ExceptionMessages::indexOutsideRange<unsigned>(
+                "number of channels",
+                numberOfChannels,
+                0,
+                ExceptionMessages::InclusiveBound,
+                AudioContext::maxNumberOfChannels(),
+                ExceptionMessages::InclusiveBound));
+        return nullptr;
     }
 
     if (!isSampleRateRangeGood(sampleRate)) {
         exceptionState.throwDOMException(SyntaxError, "sample rate (" + String::number(sampleRate) + ") must be in the range 44100-96000 Hz.");
-        return 0;
+        return nullptr;
+    }
+
+    RefPtrWillBeRawPtr<OfflineAudioContext> audioContext(adoptRefWillBeThreadSafeRefCountedGarbageCollected(new OfflineAudioContext(document, numberOfChannels, numberOfFrames, sampleRate)));
+
+    if (!audioContext->destination()) {
+        exceptionState.throwDOMException(
+            NotSupportedError,
+            "OfflineAudioContext(" + String::number(numberOfChannels)
+            + ", " + String::number(numberOfFrames)
+            + ", " + String::number(sampleRate)
+            + ")");
     }
 
-    RefPtr<OfflineAudioContext> audioContext(adoptRef(new OfflineAudioContext(document, numberOfChannels, numberOfFrames, sampleRate)));
     audioContext->suspendIfNeeded();
     return audioContext.release();
 }
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioContext.h b/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioContext.h
index 4cd6928da3e..6434069a4c9 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioContext.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioContext.h
@@ -31,9 +31,9 @@ namespace WebCore {
 
 class ExceptionState;
 
-class OfflineAudioContext : public AudioContext {
+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();
 
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioContext.idl b/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioContext.idl
index 5f5f8ac3ee4..5d5ec0c357d 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioContext.idl
+++ b/chromium/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,
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioDestinationNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioDestinationNode.cpp
index 055458216a6..4bf2fc0a5bb 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioDestinationNode.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioDestinationNode.cpp
@@ -96,6 +96,11 @@ void OfflineAudioDestinationNode::offlineRender()
     if (!m_renderBus.get())
         return;
 
+    bool isAudioContextInitialized = context()->isInitialized();
+    ASSERT(isAudioContextInitialized);
+    if (!isAudioContextInitialized)
+        return;
+
     bool channelsMatch = m_renderBus->numberOfChannels() == m_renderTarget->numberOfChannels();
     ASSERT(channelsMatch);
     if (!channelsMatch)
@@ -106,15 +111,6 @@ void OfflineAudioDestinationNode::offlineRender()
     if (!isRenderBusAllocated)
         return;
 
-    // Synchronize with HRTFDatabaseLoader.
-    // The database must be loaded before we can proceed.
-    HRTFDatabaseLoader* loader = context()->hrtfDatabaseLoader();
-    ASSERT(loader);
-    if (!loader)
-        return;
-
-    loader->waitForLoaderThreadCompletion();
-
     // Break up the render target into smaller "render quantize" sized pieces.
     // Render until we're finished.
     size_t framesToProcess = m_renderTarget->length();
@@ -157,6 +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)
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioDestinationNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioDestinationNode.h
index a2f2a615417..c2ee3ddb26a 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioDestinationNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/OfflineAudioDestinationNode.h
@@ -36,11 +36,11 @@ namespace WebCore {
 class AudioBus;
 class AudioContext;
 
-class OfflineAudioDestinationNode : public AudioDestinationNode {
+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();
@@ -50,10 +50,11 @@ public:
     virtual void uninitialize() OVERRIDE;
 
     // AudioDestinationNode
-    virtual void enableInput(const String&) OVERRIDE { }
     virtual void startRendering() OVERRIDE;
 
-    virtual float sampleRate()  const { return m_renderTarget->sampleRate(); }
+    virtual float sampleRate()  const OVERRIDE { return m_renderTarget->sampleRate(); }
+
+    virtual void trace(Visitor*) OVERRIDE;
 
 private:
     class OfflineRenderingTask;
@@ -62,7 +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;
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/OscillatorNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/OscillatorNode.cpp
index 9790a2eef25..7a6be3d0ed4 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/OscillatorNode.cpp
+++ b/chromium/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)
@@ -107,8 +107,6 @@ void OscillatorNode::setType(const String& type)
         setType(SAWTOOTH);
     else if (type == "triangle")
         setType(TRIANGLE);
-    else
-        ASSERT_NOT_REACHED();
 }
 
 bool OscillatorNode::setType(unsigned type)
@@ -118,22 +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;
     }
@@ -328,11 +326,6 @@ void OscillatorNode::process(size_t framesToProcess)
     outputBus->clearSilentFlag();
 }
 
-void OscillatorNode::reset()
-{
-    m_virtualReadIndex = 0;
-}
-
 void OscillatorNode::setPeriodicWave(PeriodicWave* periodicWave)
 {
     ASSERT(isMainThread());
@@ -348,6 +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)
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/OscillatorNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/OscillatorNode.h
index a4d25390644..be18d7d94ff 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/OscillatorNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/OscillatorNode.h
@@ -40,7 +40,7 @@ class PeriodicWave;
 
 // OscillatorNode is an audio generator of periodic waveforms.
 
-class OscillatorNode : public AudioScheduledSourceNode {
+class OscillatorNode FINAL : public AudioScheduledSourceNode {
 public:
     // The waveform type.
     // These must be defined as in the .idl file.
@@ -52,17 +52,15 @@ public:
         CUSTOM = 4
     };
 
-    static PassRefPtr<OscillatorNode> create(AudioContext*, float sampleRate);
+    static PassRefPtrWillBeRawPtr<OscillatorNode> create(AudioContext*, float sampleRate);
 
     virtual ~OscillatorNode();
 
     // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void reset();
+    virtual void process(size_t framesToProcess) OVERRIDE;
 
     String type() const;
 
-    bool setType(unsigned); // Returns true on success.
     void setType(const String&);
 
     AudioParam* frequency() { return m_frequency.get(); }
@@ -70,9 +68,13 @@ public:
 
     void setPeriodicWave(PeriodicWave*);
 
+    virtual void trace(Visitor*) OVERRIDE;
+
 private:
     OscillatorNode(AudioContext*, float sampleRate);
 
+    bool setType(unsigned); // Returns true on success.
+
     // Returns true if there are sample-accurate timeline parameter changes.
     bool calculateSampleAccuratePhaseIncrements(size_t framesToProcess);
 
@@ -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
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/OscillatorNode.idl b/chromium/third_party/WebKit/Source/modules/webaudio/OscillatorNode.idl
index 7cbe29dbd30..823d20ed521 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/OscillatorNode.idl
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/OscillatorNode.idl
@@ -22,27 +22,20 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
+enum OscillatorType {
+    "sine",
+    "square",
+    "sawtooth",
+    "triangle",
+    "custom"
+};
+    
 // OscillatorNode is an audio generator of periodic waveforms.
 [
     Conditional=WEB_AUDIO
 ] interface OscillatorNode : AudioSourceNode {
 
-    // Type constants.
-    const unsigned short SINE = 0;
-    const unsigned short SQUARE = 1;
-    const unsigned short SAWTOOTH = 2;
-    const unsigned short TRIANGLE = 3;
-    const unsigned short CUSTOM = 4;
-
-    [Custom=Setter] attribute DOMString type;
-
-    // Playback state constants.
-    const unsigned short UNSCHEDULED_STATE = 0;
-    const unsigned short SCHEDULED_STATE = 1;
-    const unsigned short PLAYING_STATE = 2;
-    const unsigned short FINISHED_STATE = 3;
-
-    readonly attribute unsigned short playbackState;
+    attribute OscillatorType type;
 
     readonly attribute AudioParam frequency; // in Hertz
     readonly attribute AudioParam detune; // in Cents
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/PannerNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/PannerNode.cpp
index fcebe1273dc..029c73cf16c 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/PannerNode.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/PannerNode.cpp
@@ -49,9 +49,24 @@ static void fixNANs(double &x)
 PannerNode::PannerNode(AudioContext* context, float sampleRate)
     : AudioNode(context, sampleRate)
     , m_panningModel(Panner::PanningModelHRTF)
+    , m_distanceModel(DistanceEffect::ModelInverse)
+    , m_position(0, 0, 0)
+    , m_orientation(1, 0, 0)
+    , m_velocity(0, 0, 0)
+    , m_isAzimuthElevationDirty(true)
+    , m_isDistanceConeGainDirty(true)
+    , m_isDopplerRateDirty(true)
     , m_lastGain(-1.0)
+    , m_cachedAzimuth(0)
+    , m_cachedElevation(0)
+    , m_cachedDistanceConeGain(1.0f)
+    , m_cachedDopplerRate(1)
     , m_connectionCount(0)
 {
+    // Load the HRTF database asynchronously so we don't block the Javascript thread while creating the HRTF database.
+    // The HRTF panner will return zeroes until the database is loaded.
+    m_hrtfDatabaseLoader = HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(context->sampleRate());
+
     ScriptWrappable::init(this);
     addInput(adoptPtr(new AudioNodeInput(this)));
     addOutput(adoptPtr(new AudioNodeOutput(this, 2)));
@@ -61,13 +76,6 @@ PannerNode::PannerNode(AudioContext* context, float sampleRate)
     m_channelCountMode = ClampedMax;
     m_channelInterpretation = AudioBus::Speakers;
 
-    m_distanceGain = AudioParam::create(context, "distanceGain", 1.0, 0.0, 1.0);
-    m_coneGain = AudioParam::create(context, "coneGain", 1.0, 0.0, 1.0);
-
-    m_position = FloatPoint3D(0, 0, 0);
-    m_orientation = FloatPoint3D(1, 0, 0);
-    m_velocity = FloatPoint3D(0, 0, 0);
-
     setNodeType(NodeTypePanner);
 
     initialize();
@@ -85,8 +93,12 @@ void PannerNode::pullInputs(size_t framesToProcess)
     if (m_connectionCount != context()->connectionCount()) {
         m_connectionCount = context()->connectionCount();
 
-        // Recursively go through all nodes connected to us.
-        notifyAudioSourcesConnectedToNode(this);
+        // A map for keeping track if we have visited a node or not. This prevents feedback loops
+        // from recursing infinitely. See crbug.com/331446.
+        HashMap<AudioNode*, bool> visitedNodes;
+
+        // Recursively go through all nodes connected to us
+        notifyAudioSourcesConnectedToNode(this, visitedNodes);
     }
 
     AudioNode::pullInputs(framesToProcess);
@@ -102,23 +114,35 @@ void PannerNode::process(size_t framesToProcess)
     }
 
     AudioBus* source = input(0)->bus();
-
     if (!source) {
         destination->zero();
         return;
     }
 
     // The audio thread can't block on this lock, so we call tryLock() instead.
-    MutexTryLocker tryLocker(m_pannerLock);
-    if (tryLocker.locked()) {
+    MutexTryLocker tryLocker(m_processLock);
+    MutexTryLocker tryListenerLocker(listener()->listenerLock());
+
+    if (tryLocker.locked() && tryListenerLocker.locked()) {
+        // HRTFDatabase should be loaded before proceeding for offline audio context when the panning model is HRTF.
+        if (m_panningModel == Panner::PanningModelHRTF && !m_hrtfDatabaseLoader->isLoaded()) {
+            if (context()->isOfflineContext()) {
+                m_hrtfDatabaseLoader->waitForLoaderThreadCompletion();
+            } else {
+                destination->zero();
+                return;
+            }
+        }
+
         // Apply the panning effect.
         double azimuth;
         double elevation;
-        getAzimuthElevation(&azimuth, &elevation);
+        azimuthElevation(&azimuth, &elevation);
+
         m_panner->pan(azimuth, elevation, source, destination, framesToProcess);
 
         // Get the distance and cone gain.
-        double totalGain = distanceConeGain();
+        float totalGain = distanceConeGain();
 
         // Snap to desired gain at the beginning.
         if (m_lastGain == -1.0)
@@ -127,24 +151,19 @@ void PannerNode::process(size_t framesToProcess)
         // Apply gain in-place with de-zippering.
         destination->copyWithGainFrom(*destination, &m_lastGain, totalGain);
     } else {
-        // Too bad - The tryLock() failed. We must be in the middle of changing the panner.
+        // Too bad - The tryLock() failed.
+        // We must be in the middle of changing the properties of the panner or the listener.
         destination->zero();
     }
 }
 
-void PannerNode::reset()
-{
-    m_lastGain = -1.0; // force to snap to initial gain
-    if (m_panner.get())
-        m_panner->reset();
-}
-
 void PannerNode::initialize()
 {
     if (isInitialized())
         return;
 
-    m_panner = Panner::create(m_panningModel, sampleRate(), context()->hrtfDatabaseLoader());
+    m_panner = Panner::create(m_panningModel, sampleRate(), m_hrtfDatabaseLoader.get());
+    listener()->addPanner(this);
 
     AudioNode::initialize();
 }
@@ -155,6 +174,8 @@ void PannerNode::uninitialize()
         return;
 
     m_panner.clear();
+    listener()->removePanner(this);
+
     AudioNode::uninitialize();
 }
 
@@ -166,12 +187,10 @@ AudioListener* PannerNode::listener()
 String PannerNode::panningModel() const
 {
     switch (m_panningModel) {
-    case EQUALPOWER:
+    case Panner::PanningModelEqualPower:
         return "equalpower";
-    case HRTF:
+    case Panner::PanningModelHRTF:
         return "HRTF";
-    case SOUNDFIELD:
-        return "soundfield";
     default:
         ASSERT_NOT_REACHED();
         return "HRTF";
@@ -181,34 +200,26 @@ String PannerNode::panningModel() const
 void PannerNode::setPanningModel(const String& model)
 {
     if (model == "equalpower")
-        setPanningModel(EQUALPOWER);
+        setPanningModel(Panner::PanningModelEqualPower);
     else if (model == "HRTF")
-        setPanningModel(HRTF);
-    else if (model == "soundfield")
-        setPanningModel(SOUNDFIELD);
-    else
-        ASSERT_NOT_REACHED();
+        setPanningModel(Panner::PanningModelHRTF);
 }
 
 bool PannerNode::setPanningModel(unsigned model)
 {
     switch (model) {
-    case EQUALPOWER:
-    case HRTF:
+    case Panner::PanningModelEqualPower:
+    case Panner::PanningModelHRTF:
         if (!m_panner.get() || model != m_panningModel) {
             // This synchronizes with process().
-            MutexLocker processLocker(m_pannerLock);
-
-            OwnPtr<Panner> newPanner = Panner::create(model, sampleRate(), context()->hrtfDatabaseLoader());
+            MutexLocker processLocker(m_processLock);
+            OwnPtr<Panner> newPanner = Panner::create(model, sampleRate(), m_hrtfDatabaseLoader.get());
             m_panner = newPanner.release();
             m_panningModel = model;
         }
         break;
-    case SOUNDFIELD:
-        // FIXME: Implement sound field model. See // https://bugs.webkit.org/show_bug.cgi?id=77367.
-        context()->executionContext()->addConsoleMessage(JSMessageSource, WarningMessageLevel, "'soundfield' panning model not implemented.");
-        break;
     default:
+        ASSERT_NOT_REACHED();
         return false;
     }
 
@@ -238,8 +249,6 @@ void PannerNode::setDistanceModel(const String& model)
         setDistanceModel(DistanceEffect::ModelInverse);
     else if (model == "exponential")
         setDistanceModel(DistanceEffect::ModelExponential);
-    else
-        ASSERT_NOT_REACHED();
 }
 
 bool PannerNode::setDistanceModel(unsigned model)
@@ -248,32 +257,135 @@ bool PannerNode::setDistanceModel(unsigned model)
     case DistanceEffect::ModelLinear:
     case DistanceEffect::ModelInverse:
     case DistanceEffect::ModelExponential:
-        m_distanceEffect.setModel(static_cast<DistanceEffect::ModelType>(model), true);
+        if (model != m_distanceModel) {
+            // This synchronizes with process().
+            MutexLocker processLocker(m_processLock);
+            m_distanceEffect.setModel(static_cast<DistanceEffect::ModelType>(model), true);
+            m_distanceModel = model;
+        }
         break;
     default:
+        ASSERT_NOT_REACHED();
         return false;
     }
 
     return true;
 }
 
-void PannerNode::getAzimuthElevation(double* outAzimuth, double* outElevation)
+void PannerNode::setRefDistance(double distance)
+{
+    if (refDistance() == distance)
+        return;
+
+    // This synchronizes with process().
+    MutexLocker processLocker(m_processLock);
+    m_distanceEffect.setRefDistance(distance);
+    markPannerAsDirty(PannerNode::DistanceConeGainDirty);
+}
+
+void PannerNode::setMaxDistance(double distance)
+{
+    if (maxDistance() == distance)
+        return;
+
+    // This synchronizes with process().
+    MutexLocker processLocker(m_processLock);
+    m_distanceEffect.setMaxDistance(distance);
+    markPannerAsDirty(PannerNode::DistanceConeGainDirty);
+}
+
+void PannerNode::setRolloffFactor(double factor)
+{
+    if (rolloffFactor() == factor)
+        return;
+
+    // This synchronizes with process().
+    MutexLocker processLocker(m_processLock);
+    m_distanceEffect.setRolloffFactor(factor);
+    markPannerAsDirty(PannerNode::DistanceConeGainDirty);
+}
+
+void PannerNode::setConeInnerAngle(double angle)
+{
+    if (coneInnerAngle() == angle)
+        return;
+
+    // This synchronizes with process().
+    MutexLocker processLocker(m_processLock);
+    m_coneEffect.setInnerAngle(angle);
+    markPannerAsDirty(PannerNode::DistanceConeGainDirty);
+}
+
+void PannerNode::setConeOuterAngle(double angle)
 {
-    // FIXME: we should cache azimuth and elevation (if possible), so we only re-calculate if a change has been made.
+    if (coneOuterAngle() == angle)
+        return;
+
+    // This synchronizes with process().
+    MutexLocker processLocker(m_processLock);
+    m_coneEffect.setOuterAngle(angle);
+    markPannerAsDirty(PannerNode::DistanceConeGainDirty);
+}
 
+void PannerNode::setConeOuterGain(double angle)
+{
+    if (coneOuterGain() == angle)
+        return;
+
+    // This synchronizes with process().
+    MutexLocker processLocker(m_processLock);
+    m_coneEffect.setOuterGain(angle);
+    markPannerAsDirty(PannerNode::DistanceConeGainDirty);
+}
+
+void PannerNode::setPosition(float x, float y, float z)
+{
+    FloatPoint3D position = FloatPoint3D(x, y, z);
+
+    if (m_position == position)
+        return;
+
+    // This synchronizes with process().
+    MutexLocker processLocker(m_processLock);
+    m_position = position;
+    markPannerAsDirty(PannerNode::AzimuthElevationDirty | PannerNode::DistanceConeGainDirty | PannerNode::DopplerRateDirty);
+}
+
+void PannerNode::setOrientation(float x, float y, float z)
+{
+    FloatPoint3D orientation = FloatPoint3D(x, y, z);
+
+    if (m_orientation == orientation)
+        return;
+
+    // This synchronizes with process().
+    MutexLocker processLocker(m_processLock);
+    m_orientation = orientation;
+    markPannerAsDirty(PannerNode::DistanceConeGainDirty);
+}
+
+void PannerNode::setVelocity(float x, float y, float z)
+{
+    FloatPoint3D velocity = FloatPoint3D(x, y, z);
+
+    if (m_velocity == velocity)
+        return;
+
+    // This synchronizes with process().
+    MutexLocker processLocker(m_processLock);
+    m_velocity = velocity;
+    markPannerAsDirty(PannerNode::DopplerRateDirty);
+}
+
+void PannerNode::calculateAzimuthElevation(double* outAzimuth, double* outElevation)
+{
     double azimuth = 0.0;
 
     // Calculate the source-listener vector
     FloatPoint3D listenerPosition = listener()->position();
     FloatPoint3D sourceListener = m_position - listenerPosition;
 
-    if (sourceListener.isZero()) {
-        // degenerate case if source and listener are at the same point
-        *outAzimuth = 0.0;
-        *outElevation = 0.0;
-        return;
-    }
-
+    // normalize() does nothing if the length of |sourceListener| is zero.
     sourceListener.normalize();
 
     // Align axes
@@ -321,11 +433,9 @@ void PannerNode::getAzimuthElevation(double* outAzimuth, double* outElevation)
         *outElevation = elevation;
 }
 
-float PannerNode::dopplerRate()
+double PannerNode::calculateDopplerRate()
 {
     double dopplerShift = 1.0;
-
-    // FIXME: optimize for case when neither source nor listener has changed...
     double dopplerFactor = listener()->dopplerFactor();
 
     if (dopplerFactor > 0.0) {
@@ -345,48 +455,97 @@ float PannerNode::dopplerRate()
 
             double sourceListenerMagnitude = sourceToListener.length();
 
-            double listenerProjection = sourceToListener.dot(listenerVelocity) / sourceListenerMagnitude;
-            double sourceProjection = sourceToListener.dot(sourceVelocity) / sourceListenerMagnitude;
+            if (!sourceListenerMagnitude) {
+                // Source and listener are at the same position. Skip the computation of the doppler
+                // shift, and just return the cached value.
+                dopplerShift = m_cachedDopplerRate;
+            } else {
+                double listenerProjection = sourceToListener.dot(listenerVelocity) / sourceListenerMagnitude;
+                double sourceProjection = sourceToListener.dot(sourceVelocity) / sourceListenerMagnitude;
+
+                listenerProjection = -listenerProjection;
+                sourceProjection = -sourceProjection;
+
+                double scaledSpeedOfSound = speedOfSound / dopplerFactor;
+                listenerProjection = min(listenerProjection, scaledSpeedOfSound);
+                sourceProjection = min(sourceProjection, scaledSpeedOfSound);
+
+                dopplerShift = ((speedOfSound - dopplerFactor * listenerProjection) / (speedOfSound - dopplerFactor * sourceProjection));
+                fixNANs(dopplerShift); // avoid illegal values
+
+                // Limit the pitch shifting to 4 octaves up and 3 octaves down.
+                if (dopplerShift > 16.0)
+                    dopplerShift = 16.0;
+                else if (dopplerShift < 0.125)
+                    dopplerShift = 0.125;
+            }
+        }
+    }
 
-            listenerProjection = -listenerProjection;
-            sourceProjection = -sourceProjection;
+    return dopplerShift;
+}
 
-            double scaledSpeedOfSound = speedOfSound / dopplerFactor;
-            listenerProjection = min(listenerProjection, scaledSpeedOfSound);
-            sourceProjection = min(sourceProjection, scaledSpeedOfSound);
+float PannerNode::calculateDistanceConeGain()
+{
+    FloatPoint3D listenerPosition = listener()->position();
 
-            dopplerShift = ((speedOfSound - dopplerFactor * listenerProjection) / (speedOfSound - dopplerFactor * sourceProjection));
-            fixNANs(dopplerShift); // avoid illegal values
+    double listenerDistance = m_position.distanceTo(listenerPosition);
+    double distanceGain = m_distanceEffect.gain(listenerDistance);
+    double coneGain = m_coneEffect.gain(m_position, m_orientation, listenerPosition);
 
-            // Limit the pitch shifting to 4 octaves up and 3 octaves down.
-            if (dopplerShift > 16.0)
-                dopplerShift = 16.0;
-            else if (dopplerShift < 0.125)
-                dopplerShift = 0.125;
-        }
+    return float(distanceGain * coneGain);
+}
+
+void PannerNode::azimuthElevation(double* outAzimuth, double* outElevation)
+{
+    ASSERT(context()->isAudioThread());
+
+    if (isAzimuthElevationDirty()) {
+        calculateAzimuthElevation(&m_cachedAzimuth, &m_cachedElevation);
+        m_isAzimuthElevationDirty = false;
     }
 
-    return static_cast<float>(dopplerShift);
+    *outAzimuth = m_cachedAzimuth;
+    *outElevation = m_cachedElevation;
+}
+
+double PannerNode::dopplerRate()
+{
+    ASSERT(context()->isAudioThread());
+
+    if (isDopplerRateDirty()) {
+        m_cachedDopplerRate = calculateDopplerRate();
+        m_isDopplerRateDirty = false;
+    }
+
+    return m_cachedDopplerRate;
 }
 
 float PannerNode::distanceConeGain()
 {
-    FloatPoint3D listenerPosition = listener()->position();
+    ASSERT(context()->isAudioThread());
 
-    double listenerDistance = m_position.distanceTo(listenerPosition);
-    double distanceGain = m_distanceEffect.gain(listenerDistance);
+    if (isDistanceConeGainDirty()) {
+        m_cachedDistanceConeGain = calculateDistanceConeGain();
+        m_isDistanceConeGainDirty = false;
+    }
 
-    m_distanceGain->setValue(static_cast<float>(distanceGain));
+    return m_cachedDistanceConeGain;
+}
 
-    // FIXME: could optimize by caching coneGain
-    double coneGain = m_coneEffect.gain(m_position, m_orientation, listenerPosition);
+void PannerNode::markPannerAsDirty(unsigned dirty)
+{
+    if (dirty & PannerNode::AzimuthElevationDirty)
+        m_isAzimuthElevationDirty = true;
 
-    m_coneGain->setValue(static_cast<float>(coneGain));
+    if (dirty & PannerNode::DistanceConeGainDirty)
+        m_isDistanceConeGainDirty = true;
 
-    return float(distanceGain * coneGain);
+    if (dirty & PannerNode::DopplerRateDirty)
+        m_isDopplerRateDirty = true;
 }
 
-void PannerNode::notifyAudioSourcesConnectedToNode(AudioNode* node)
+void PannerNode::notifyAudioSourcesConnectedToNode(AudioNode* node, HashMap<AudioNode*, bool>& visitedNodes)
 {
     ASSERT(node);
     if (!node)
@@ -405,7 +564,14 @@ void PannerNode::notifyAudioSourcesConnectedToNode(AudioNode* node)
             for (unsigned j = 0; j < input->numberOfRenderingConnections(); ++j) {
                 AudioNodeOutput* connectedOutput = input->renderingOutput(j);
                 AudioNode* connectedNode = connectedOutput->node();
-                notifyAudioSourcesConnectedToNode(connectedNode); // recurse
+                HashMap<AudioNode*, bool>::iterator iterator = visitedNodes.find(connectedNode);
+
+                // If we've seen this node already, we don't need to process it again. Otherwise,
+                // mark it as visited and recurse through the node looking for sources.
+                if (iterator == visitedNodes.end()) {
+                    visitedNodes.set(connectedNode, true);
+                    notifyAudioSourcesConnectedToNode(connectedNode, visitedNodes); // recurse
+                }
             }
         }
     }
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/PannerNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/PannerNode.h
index 358eccabea2..840982b387f 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/PannerNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/PannerNode.h
@@ -28,11 +28,12 @@
 #include "platform/audio/AudioBus.h"
 #include "platform/audio/Cone.h"
 #include "platform/audio/Distance.h"
+#include "platform/audio/HRTFDatabaseLoader.h"
 #include "platform/audio/Panner.h"
 #include "modules/webaudio/AudioListener.h"
 #include "modules/webaudio/AudioNode.h"
-#include "modules/webaudio/AudioParam.h"
 #include "platform/geometry/FloatPoint3D.h"
+#include "wtf/HashMap.h"
 #include "wtf/OwnPtr.h"
 
 namespace WebCore {
@@ -44,87 +45,64 @@ namespace WebCore {
 // A cone effect will attenuate the gain as the orientation moves away from the listener.
 // All of these effects follow the OpenAL specification very closely.
 
-class PannerNode : public AudioNode {
+class PannerNode FINAL : public AudioNode {
 public:
-    // These must be defined as in the .idl file and must match those in the Panner class.
+    // These enums are used to distinguish what cached values of panner are dirty.
     enum {
-        EQUALPOWER = 0,
-        HRTF = 1,
-        SOUNDFIELD = 2,
+        AzimuthElevationDirty = 0x1,
+        DistanceConeGainDirty = 0x2,
+        DopplerRateDirty = 0x4,
     };
 
-    // These must be defined as in the .idl file and must match those
-    // in the DistanceEffect class.
-    enum {
-        LINEAR_DISTANCE = 0,
-        INVERSE_DISTANCE = 1,
-        EXPONENTIAL_DISTANCE = 2,
-    };
-
-    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();
 
     // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void pullInputs(size_t framesToProcess);
-    virtual void reset();
-    virtual void initialize();
-    virtual void uninitialize();
-
-    // Listener
-    AudioListener* listener();
+    virtual void process(size_t framesToProcess) OVERRIDE;
+    virtual void pullInputs(size_t framesToProcess) OVERRIDE;
+    virtual void initialize() OVERRIDE;
+    virtual void uninitialize() OVERRIDE;
 
     // Panning model
     String panningModel() const;
-    bool setPanningModel(unsigned); // Returns true on success.
     void setPanningModel(const String&);
 
-    // Position
-    FloatPoint3D position() const { return m_position; }
-    void setPosition(float x, float y, float z) { m_position = FloatPoint3D(x, y, z); }
-
-    // Orientation
-    FloatPoint3D orientation() const { return m_position; }
-    void setOrientation(float x, float y, float z) { m_orientation = FloatPoint3D(x, y, z); }
-
-    // Velocity
-    FloatPoint3D velocity() const { return m_velocity; }
-    void setVelocity(float x, float y, float z) { m_velocity = FloatPoint3D(x, y, z); }
+    // Position, orientation and velocity
+    void setPosition(float x, float y, float z);
+    void setOrientation(float x, float y, float z);
+    void setVelocity(float x, float y, float z);
 
     // Distance parameters
     String distanceModel() const;
-    bool setDistanceModel(unsigned); // Returns true on success.
     void setDistanceModel(const String&);
 
     double refDistance() { return m_distanceEffect.refDistance(); }
-    void setRefDistance(double refDistance) { m_distanceEffect.setRefDistance(refDistance); }
+    void setRefDistance(double);
 
     double maxDistance() { return m_distanceEffect.maxDistance(); }
-    void setMaxDistance(double maxDistance) { m_distanceEffect.setMaxDistance(maxDistance); }
+    void setMaxDistance(double);
 
     double rolloffFactor() { return m_distanceEffect.rolloffFactor(); }
-    void setRolloffFactor(double rolloffFactor) { m_distanceEffect.setRolloffFactor(rolloffFactor); }
+    void setRolloffFactor(double);
 
     // Sound cones - angles in degrees
     double coneInnerAngle() const { return m_coneEffect.innerAngle(); }
-    void setConeInnerAngle(double angle) { m_coneEffect.setInnerAngle(angle); }
+    void setConeInnerAngle(double);
 
     double coneOuterAngle() const { return m_coneEffect.outerAngle(); }
-    void setConeOuterAngle(double angle) { m_coneEffect.setOuterAngle(angle); }
+    void setConeOuterAngle(double);
 
     double coneOuterGain() const { return m_coneEffect.outerGain(); }
-    void setConeOuterGain(double angle) { m_coneEffect.setOuterGain(angle); }
+    void setConeOuterGain(double);
 
-    void getAzimuthElevation(double* outAzimuth, double* outElevation);
-    float dopplerRate();
+    void markPannerAsDirty(unsigned);
 
-    // Accessors for dynamically calculated gain values.
-    AudioParam* distanceGain() { return m_distanceGain.get(); }
-    AudioParam* coneGain() { return m_coneGain.get(); }
+    // It must be called on audio thread, currently called only process() in AudioBufferSourceNode.
+    double dopplerRate();
 
     virtual double tailTime() const OVERRIDE { return m_panner ? m_panner->tailTime() : 0; }
     virtual double latencyTime() const OVERRIDE { return m_panner ? m_panner->latencyTime() : 0; }
@@ -132,31 +110,58 @@ public:
 private:
     PannerNode(AudioContext*, float sampleRate);
 
-    // Returns the combined distance and cone gain attenuation.
+    // AudioContext's listener
+    AudioListener* listener();
+
+    bool setPanningModel(unsigned); // Returns true on success.
+    bool setDistanceModel(unsigned); // Returns true on success.
+
+    void calculateAzimuthElevation(double* outAzimuth, double* outElevation);
+    float calculateDistanceConeGain(); // Returns the combined distance and cone gain attenuation.
+    double calculateDopplerRate();
+
+    void azimuthElevation(double* outAzimuth, double* outElevation);
     float distanceConeGain();
 
+    bool isAzimuthElevationDirty() const { return m_isAzimuthElevationDirty; }
+    bool isDistanceConeGainDirty() const { return m_isDistanceConeGainDirty; }
+    bool isDopplerRateDirty() const { return m_isDopplerRateDirty; }
+
     // Notifies any AudioBufferSourceNodes connected to us either directly or indirectly about our existence.
     // This is in order to handle the pitch change necessary for the doppler shift.
-    void notifyAudioSourcesConnectedToNode(AudioNode*);
+    void notifyAudioSourcesConnectedToNode(AudioNode*, HashMap<AudioNode*, bool> &visitedNodes);
 
     OwnPtr<Panner> m_panner;
     unsigned m_panningModel;
+    unsigned m_distanceModel;
 
+    // Current source location information
     FloatPoint3D m_position;
     FloatPoint3D m_orientation;
     FloatPoint3D m_velocity;
 
+    bool m_isAzimuthElevationDirty;
+    bool m_isDistanceConeGainDirty;
+    bool m_isDopplerRateDirty;
+
     // Gain
-    RefPtr<AudioParam> m_distanceGain;
-    RefPtr<AudioParam> m_coneGain;
     DistanceEffect m_distanceEffect;
     ConeEffect m_coneEffect;
     float m_lastGain;
 
+    // Cached values
+    double m_cachedAzimuth;
+    double m_cachedElevation;
+    float m_cachedDistanceConeGain;
+    double m_cachedDopplerRate;
+
+    RefPtr<HRTFDatabaseLoader> m_hrtfDatabaseLoader;
+
+    // AudioContext's connection count
     unsigned m_connectionCount;
 
-    // Synchronize process() and setPanningModel() which can change the panner.
-    mutable Mutex m_pannerLock;
+    // Synchronize process() with setting of the panning model, source's location information, listener, distance parameters and sound cones.
+    mutable Mutex m_processLock;
 };
 
 } // namespace WebCore
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/PannerNode.idl b/chromium/third_party/WebKit/Source/modules/webaudio/PannerNode.idl
index 455740817ba..cbd1fc8f797 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/PannerNode.idl
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/PannerNode.idl
@@ -22,22 +22,23 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
+enum PanningModelType {
+    "equalpower",
+    "HRTF"
+};
+
+enum DistanceModelType {
+    "linear",
+    "inverse",
+    "exponential"
+};
+
 [
     NoInterfaceObject,
     Conditional=WEB_AUDIO
 ] interface PannerNode : AudioNode {
-    // Panning model
-    const unsigned short EQUALPOWER = 0;
-    const unsigned short HRTF = 1;
-    const unsigned short SOUNDFIELD = 2;
-
-    // Distance model
-    const unsigned short LINEAR_DISTANCE = 0;
-    const unsigned short INVERSE_DISTANCE = 1;
-    const unsigned short EXPONENTIAL_DISTANCE = 2;
-
     // Default model for stereo is HRTF
-    [Custom=Setter] attribute DOMString panningModel;
+    attribute PanningModelType panningModel;
 
     // Uses a 3D cartesian coordinate system
     void setPosition(float x, float y, float z);
@@ -45,7 +46,7 @@
     void setVelocity(float x, float y, float z);
 
     // Distance model
-    [Custom=Setter] attribute DOMString distanceModel;
+    attribute DistanceModelType distanceModel;
 
     attribute double refDistance;
     attribute double maxDistance;
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/PeriodicWave.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/PeriodicWave.cpp
index 28b7a24dc19..3724e5fce51 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/PeriodicWave.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/PeriodicWave.cpp
@@ -45,43 +45,43 @@ 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;
     }
-    return 0;
+    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;
 }
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/PeriodicWave.h b/chromium/third_party/WebKit/Source/modules/webaudio/PeriodicWave.h
index d10b3e2c90f..1e62820f3ff 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/PeriodicWave.h
+++ b/chromium/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
@@ -62,7 +62,8 @@ public:
     float rateScale() const { return m_rateScale; }
 
     unsigned periodicWaveSize() const { return m_periodicWaveSize; }
-    float sampleRate() const { return m_sampleRate; }
+
+    void trace(Visitor*) { }
 
 private:
     explicit PeriodicWave(float sampleRate);
@@ -81,8 +82,6 @@ private:
 
     float m_rateScale;
 
-    unsigned numberOfRanges() const { return m_numberOfRanges; }
-
     // Maximum possible number of partials (before culling).
     unsigned maxNumberOfPartials() const;
 
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/PeriodicWave.idl b/chromium/third_party/WebKit/Source/modules/webaudio/PeriodicWave.idl
index 40a04b127b0..3dccbc7b665 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/PeriodicWave.idl
+++ b/chromium/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 {
 
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/RealtimeAnalyser.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/RealtimeAnalyser.cpp
index d470a267079..b768a243917 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/RealtimeAnalyser.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/RealtimeAnalyser.cpp
@@ -30,7 +30,6 @@
 
 #include "platform/audio/AudioBus.h"
 #include "platform/audio/AudioUtilities.h"
-#include "platform/audio/FFTFrame.h"
 #include "platform/audio/VectorMath.h"
 
 #include <algorithm>
@@ -67,13 +66,6 @@ RealtimeAnalyser::RealtimeAnalyser()
     m_analysisFrame = adoptPtr(new FFTFrame(DefaultFFTSize));
 }
 
-void RealtimeAnalyser::reset()
-{
-    m_writeIndex = 0;
-    m_inputBuffer.zero();
-    m_magnitudeBuffer.zero();
-}
-
 bool RealtimeAnalyser::setFftSize(size_t size)
 {
     ASSERT(isMainThread());
@@ -145,7 +137,7 @@ void applyWindow(float* p, size_t n)
 
     for (unsigned i = 0; i < n; ++i) {
         double x = static_cast<double>(i) / static_cast<double>(n);
-        double window = a0 - a1 * cos(2 * piDouble * x) + a2 * cos(4 * piDouble * x);
+        double window = a0 - a1 * cos(twoPiDouble * x) + a2 * cos(twoPiDouble * 2.0 * x);
         p[i] *= float(window);
     }
 }
@@ -185,7 +177,7 @@ void RealtimeAnalyser::doFFTAnalysis()
     imagP[0] = 0;
 
     // Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT scaling factor).
-    const double magnitudeScale = 1.0 / DefaultFFTSize;
+    const double magnitudeScale = 1.0 / fftSize;
 
     // A value of 0 does no averaging with the previous result.  Larger values produce slower, but smoother changes.
     double k = m_smoothingTimeConstant;
@@ -264,6 +256,35 @@ void RealtimeAnalyser::getByteFrequencyData(Uint8Array* destinationArray)
     }
 }
 
+void RealtimeAnalyser::getFloatTimeDomainData(Float32Array* destinationArray)
+{
+    ASSERT(isMainThread());
+
+    if (!destinationArray)
+        return;
+
+    unsigned fftSize = this->fftSize();
+    size_t len = min(fftSize, destinationArray->length());
+    if (len > 0) {
+        bool isInputBufferGood = m_inputBuffer.size() == InputBufferSize && m_inputBuffer.size() > fftSize;
+        ASSERT(isInputBufferGood);
+        if (!isInputBufferGood)
+            return;
+
+        float* inputBuffer = m_inputBuffer.data();
+        float* destination = destinationArray->data();
+
+        unsigned writeIndex = m_writeIndex;
+
+        for (unsigned i = 0; i < len; ++i) {
+            // Buffer access is protected due to modulo operation.
+            float value = inputBuffer[(i + writeIndex - fftSize + InputBufferSize) % InputBufferSize];
+
+            destination[i] = value;
+        }
+    }
+}
+
 void RealtimeAnalyser::getByteTimeDomainData(Uint8Array* destinationArray)
 {
     ASSERT(isMainThread());
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/RealtimeAnalyser.h b/chromium/third_party/WebKit/Source/modules/webaudio/RealtimeAnalyser.h
index 5ce798f0703..87e7fda037a 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/RealtimeAnalyser.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/RealtimeAnalyser.h
@@ -26,6 +26,7 @@
 #define RealtimeAnalyser_h
 
 #include "platform/audio/AudioArray.h"
+#include "platform/audio/FFTFrame.h"
 #include "wtf/Forward.h"
 #include "wtf/Noncopyable.h"
 #include "wtf/OwnPtr.h"
@@ -33,31 +34,29 @@
 namespace WebCore {
 
 class AudioBus;
-class FFTFrame;
 
 class RealtimeAnalyser FINAL {
     WTF_MAKE_NONCOPYABLE(RealtimeAnalyser);
 public:
     RealtimeAnalyser();
 
-    void reset();
-
     size_t fftSize() const { return m_fftSize; }
     bool setFftSize(size_t);
 
     unsigned frequencyBinCount() const { return m_fftSize / 2; }
 
-    void setMinDecibels(float k) { m_minDecibels = k; }
-    float minDecibels() const { return static_cast<float>(m_minDecibels); }
+    void setMinDecibels(double k) { m_minDecibels = k; }
+    double minDecibels() const { return m_minDecibels; }
 
-    void setMaxDecibels(float k) { m_maxDecibels = k; }
-    float maxDecibels() const { return static_cast<float>(m_maxDecibels); }
+    void setMaxDecibels(double k) { m_maxDecibels = k; }
+    double maxDecibels() const { return m_maxDecibels; }
 
-    void setSmoothingTimeConstant(float k) { m_smoothingTimeConstant = k; }
-    float smoothingTimeConstant() const { return static_cast<float>(m_smoothingTimeConstant); }
+    void setSmoothingTimeConstant(double k) { m_smoothingTimeConstant = k; }
+    double smoothingTimeConstant() const { return m_smoothingTimeConstant; }
 
     void getFloatFrequencyData(Float32Array*);
     void getByteFrequencyData(Uint8Array*);
+    void getFloatTimeDomainData(Float32Array*);
     void getByteTimeDomainData(Uint8Array*);
 
     // The audio thread writes input data here.
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/ScriptProcessorNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/ScriptProcessorNode.cpp
index d3f14c884f2..dbbcfdd595f 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/ScriptProcessorNode.cpp
+++ b/chromium/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) {
@@ -72,19 +72,19 @@ PassRefPtr<ScriptProcessorNode> ScriptProcessorNode::create(AudioContext* contex
     case 16384:
         break;
     default:
-        return 0;
+        return nullptr;
     }
 
     if (!numberOfInputChannels && !numberOfOutputChannels)
-        return 0;
+        return nullptr;
 
     if (numberOfInputChannels > AudioContext::maxNumberOfChannels())
-        return 0;
+        return nullptr;
 
     if (numberOfOutputChannels > AudioContext::maxNumberOfChannels())
-        return 0;
+        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)
@@ -93,7 +93,6 @@ ScriptProcessorNode::ScriptProcessorNode(AudioContext* context, float sampleRate
     , m_doubleBufferIndexForEvent(0)
     , m_bufferSize(bufferSize)
     , m_bufferReadWriteIndex(0)
-    , m_isRequestOutstanding(false)
     , m_numberOfInputChannels(numberOfInputChannels)
     , m_numberOfOutputChannels(numberOfOutputChannels)
     , m_internalInputBus(AudioBus::create(numberOfInputChannels, AudioNode::ProcessingSizeInFrames, false))
@@ -128,8 +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) : 0;
-        RefPtr<AudioBuffer> outputBuffer = m_numberOfOutputChannels ? AudioBuffer::create(m_numberOfOutputChannels, bufferSize(), sampleRate) : 0;
+        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);
@@ -214,7 +213,9 @@ void ScriptProcessorNode::process(size_t framesToProcess)
     if (!m_bufferReadWriteIndex) {
         // Avoid building up requests on the main thread to fire process events when they're not being handled.
         // This could be a problem if the main thread is very busy doing other things and is being held up handling previous requests.
-        if (m_isRequestOutstanding) {
+        // The audio thread can't block on this lock, so we call tryLock() instead.
+        MutexTryLocker tryLocker(m_processEventLock);
+        if (!tryLocker.locked()) {
             // We're late in handling the previous request. The main thread must be very busy.
             // The best we can do is clear out the buffer ourself here.
             outputBuffer->zero();
@@ -224,7 +225,6 @@ void ScriptProcessorNode::process(size_t framesToProcess)
 
             // Fire the event on the main thread, not this one (which is the realtime audio thread).
             m_doubleBufferIndexForEvent = m_doubleBufferIndex;
-            m_isRequestOutstanding = true;
             callOnMainThread(fireProcessEventDispatch, this);
         }
 
@@ -247,7 +247,7 @@ void ScriptProcessorNode::fireProcessEventDispatch(void* userData)
 
 void ScriptProcessorNode::fireProcessEvent()
 {
-    ASSERT(isMainThread() && m_isRequestOutstanding);
+    ASSERT(isMainThread());
 
     bool isIndexGood = m_doubleBufferIndexForEvent < 2;
     ASSERT(isIndexGood);
@@ -262,22 +262,15 @@ void ScriptProcessorNode::fireProcessEvent()
 
     // Avoid firing the event if the document has already gone away.
     if (context()->executionContext()) {
-        // Let the audio thread know we've gotten to the point where it's OK for it to make another request.
-        m_isRequestOutstanding = false;
+        // This synchronizes with process().
+        MutexLocker processLocker(m_processEventLock);
 
-        // Call the JavaScript event handler which will do the audio processing.
-        dispatchEvent(AudioProcessingEvent::create(inputBuffer, outputBuffer));
-    }
-}
-
-void ScriptProcessorNode::reset()
-{
-    m_bufferReadWriteIndex = 0;
-    m_doubleBufferIndex = 0;
+        // Calculate a playbackTime with the buffersize which needs to be processed each time onaudioprocess is called.
+        // The outputBuffer being passed to JS will be played after exhuasting previous outputBuffer by double-buffering.
+        double playbackTime = (context()->currentSampleFrame() + m_bufferSize) / static_cast<double>(context()->sampleRate());
 
-    for (unsigned i = 0; i < 2; ++i) {
-        m_inputBuffers[i]->zero();
-        m_outputBuffers[i]->zero();
+        // Call the JavaScript event handler which will do the audio processing.
+        dispatchEvent(AudioProcessingEvent::create(inputBuffer, outputBuffer, playbackTime));
     }
 }
 
@@ -291,6 +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)
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/ScriptProcessorNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/ScriptProcessorNode.h
index 3e1aa7a5535..edbe7423101 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/ScriptProcessorNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/ScriptProcessorNode.h
@@ -44,26 +44,26 @@ class AudioProcessingEvent;
 // The "onaudioprocess" attribute is an event listener which will get called periodically with an AudioProcessingEvent which has
 // AudioBuffers for each input and output.
 
-class ScriptProcessorNode : public AudioNode {
+class ScriptProcessorNode FINAL : public AudioNode {
 public:
     // bufferSize must be one of the following values: 256, 512, 1024, 2048, 4096, 8192, 16384.
     // 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();
 
     // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void reset();
-    virtual void initialize();
-    virtual void uninitialize();
+    virtual void process(size_t framesToProcess) OVERRIDE;
+    virtual void initialize() OVERRIDE;
+    virtual void uninitialize() OVERRIDE;
 
     size_t bufferSize() const { return m_bufferSize; }
 
     DEFINE_ATTRIBUTE_EVENT_LISTENER(audioprocess);
 
+    void trace(Visitor*);
 
 private:
     virtual double tailTime() const OVERRIDE;
@@ -79,17 +79,19 @@ 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;
-    volatile bool m_isRequestOutstanding;
 
     unsigned m_numberOfInputChannels;
     unsigned m_numberOfOutputChannels;
 
     RefPtr<AudioBus> m_internalInputBus;
+
+    // Synchronize process() with fireProcessEvent().
+    mutable Mutex m_processEventLock;
 };
 
 } // namespace WebCore
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperDSPKernel.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperDSPKernel.cpp
index 6134abe7571..e2339c43b29 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperDSPKernel.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperDSPKernel.cpp
@@ -100,24 +100,30 @@ void WaveShaperDSPKernel::processCurve(const float* source, float* destination,
     for (unsigned i = 0; i < framesToProcess; ++i) {
         const float input = source[i];
 
-        // Calculate a virtual index based on input -1 -> +1 with 0 being at the center of the curve data.
-        // Then linearly interpolate between the two points in the curve.
-        double virtualIndex = 0.5 * (input + 1) * curveLength;
-        int index1 = static_cast<int>(virtualIndex);
-        int index2 = index1 + 1;
-        double interpolationFactor = virtualIndex - index1;
-
-        // Clip index to the input range of the curve.
-        // This takes care of input outside of nominal range -1 -> +1
-        index1 = max(index1, 0);
-        index1 = min(index1, curveLength - 1);
-        index2 = max(index2, 0);
-        index2 = min(index2, curveLength - 1);
-
-        double value1 = curveData[index1];
-        double value2 = curveData[index2];
-
-        double output = (1.0 - interpolationFactor) * value1 + interpolationFactor * value2;
+        // Calculate a virtual index based on input -1 -> +1 with -1 being curve[0], +1 being
+        // curve[curveLength - 1], and 0 being at the center of the curve data. Then linearly
+        // interpolate between the two points in the curve.
+        double virtualIndex = 0.5 * (input + 1) * (curveLength - 1);
+        double output;
+
+        if (virtualIndex < 0) {
+            // input < -1, so use curve[0]
+            output = curveData[0];
+        } else if (virtualIndex >= curveLength - 1) {
+            // input >= 1, so use last curve value
+            output = curveData[curveLength - 1];
+        } else {
+            // The general case where -1 <= input < 1, where 0 <= virtualIndex < curveLength - 1,
+            // so interpolate between the nearest samples on the curve.
+            unsigned index1 = static_cast<unsigned>(virtualIndex);
+            unsigned index2 = index1 + 1;
+            double interpolationFactor = virtualIndex - index1;
+
+            double value1 = curveData[index1];
+            double value2 = curveData[index2];
+
+            output = (1.0 - interpolationFactor) * value1 + interpolationFactor * value2;
+        }
         destination[i] = output;
     }
 }
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperDSPKernel.h b/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperDSPKernel.h
index 78d7bfe85c4..54017a18c8f 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperDSPKernel.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperDSPKernel.h
@@ -38,13 +38,13 @@ class WaveShaperProcessor;
 
 // WaveShaperDSPKernel is an AudioDSPKernel and is responsible for non-linear distortion on one channel.
 
-class WaveShaperDSPKernel : public AudioDSPKernel {
+class WaveShaperDSPKernel FINAL : public AudioDSPKernel {
 public:
     explicit WaveShaperDSPKernel(WaveShaperProcessor*);
 
     // AudioDSPKernel
-    virtual void process(const float* source, float* dest, size_t framesToProcess);
-    virtual void reset();
+    virtual void process(const float* source, float* dest, size_t framesToProcess) OVERRIDE;
+    virtual void reset() OVERRIDE;
     virtual double tailTime() const OVERRIDE { return 0; }
     virtual double latencyTime() const OVERRIDE;
 
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperNode.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperNode.cpp
index 3e84aaa304b..0474826663e 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperNode.cpp
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperNode.cpp
@@ -72,12 +72,7 @@ void WaveShaperNode::setOversample(const String& type, ExceptionState& exception
     } else if (type == "4x") {
         waveShaperProcessor()->setOversample(WaveShaperProcessor::OverSample4x);
     } else {
-        exceptionState.throwDOMException(
-            InvalidStateError,
-            ExceptionMessages::failedToSet(
-                "oversample",
-                "WaveShaperNode",
-                "invalid oversample '" + type + "': must be 'none', '2x', or '4x'."));
+        ASSERT_NOT_REACHED();
     }
 }
 
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperNode.h b/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperNode.h
index 8855630bb4d..159a39fb012 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperNode.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperNode.h
@@ -34,11 +34,11 @@ namespace WebCore {
 
 class ExceptionState;
 
-class WaveShaperNode : public AudioBasicProcessorNode {
+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.
@@ -48,8 +48,6 @@ public:
     void setOversample(const String& , ExceptionState&);
     String oversample() const;
 
-    double latency() const { return latencyTime(); }
-
 private:
     explicit WaveShaperNode(AudioContext*);
 
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperNode.idl b/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperNode.idl
index e1ee72b6326..e17ad17ee4a 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperNode.idl
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperNode.idl
@@ -22,9 +22,15 @@
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
+enum OverSampleType {
+    "none",
+    "2x",
+    "4x"
+};
+
 [
     Conditional=WEB_AUDIO
 ] interface WaveShaperNode : AudioNode {
     attribute Float32Array curve;
-    [RaisesException=Setter] attribute DOMString oversample;
+    [RaisesException=Setter] attribute OverSampleType oversample;
 };
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperProcessor.h b/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperProcessor.h
index 79c826496ff..e69f0a616d0 100644
--- a/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperProcessor.h
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/WaveShaperProcessor.h
@@ -36,7 +36,7 @@ namespace WebCore {
 
 // WaveShaperProcessor is an AudioDSPKernelProcessor which uses WaveShaperDSPKernel objects to implement non-linear distortion effects.
 
-class WaveShaperProcessor : public AudioDSPKernelProcessor {
+class WaveShaperProcessor FINAL : public AudioDSPKernelProcessor {
 public:
     enum OverSampleType {
         OverSampleNone,
@@ -48,9 +48,9 @@ public:
 
     virtual ~WaveShaperProcessor();
 
-    virtual PassOwnPtr<AudioDSPKernel> createKernel();
+    virtual PassOwnPtr<AudioDSPKernel> createKernel() OVERRIDE;
 
-    virtual void process(const AudioBus* source, AudioBus* destination, size_t framesToProcess);
+    virtual void process(const AudioBus* source, AudioBus* destination, size_t framesToProcess) OVERRIDE;
 
     void setCurve(Float32Array*);
     Float32Array* curve() { return m_curve.get(); }
@@ -63,9 +63,6 @@ private:
     RefPtr<Float32Array> m_curve;
 
     OverSampleType m_oversample;
-
-    // This synchronizes process() with setCurve().
-    mutable Mutex m_processLock;
 };
 
 } // namespace WebCore
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/WindowWebAudio.idl b/chromium/third_party/WebKit/Source/modules/webaudio/WindowWebAudio.idl
new file mode 100644
index 00000000000..75788e12b42
--- /dev/null
+++ b/chromium/third_party/WebKit/Source/modules/webaudio/WindowWebAudio.idl
@@ -0,0 +1,12 @@
+// Copyright (c) 2014 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+[
+    ImplementedAs=DOMWindowWebAudio,
+] partial interface Window {
+    [Conditional=WEB_AUDIO, RuntimeEnabled=WebAudio] attribute AudioContextConstructor AudioContext;
+    [Conditional=WEB_AUDIO, RuntimeEnabled=WebAudio] attribute OfflineAudioContextConstructor OfflineAudioContext;
+    [Conditional=WEB_AUDIO, RuntimeEnabled=WebAudio] attribute AudioContextConstructor webkitAudioContext;
+    [Conditional=WEB_AUDIO, RuntimeEnabled=WebAudio] attribute OfflineAudioContextConstructor webkitOfflineAudioContext;
+};
author	Jocelyn Turcotte <jocelyn.turcotte@digia.com>	2014-08-08 14:30:41 +0200
committer	Jocelyn Turcotte <jocelyn.turcotte@digia.com>	2014-08-12 13:49:54 +0200
commit	ab0a50979b9eb4dfa3320eff7e187e41efedf7a9 (patch)
tree	498dfb8a97ff3361a9f7486863a52bb4e26bb898 /chromium/third_party/WebKit/Source/modules/webaudio
parent	4ce69f7403811819800e7c5ae1318b2647e778d1 (diff)