Move webaudio to oilpan

Source/modules/webaudio/ScriptProcessorNode.h

 /*
  * Copyright (C) 2010, Google Inc. All rights reserved.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
  * are met:
  * 1.  Redistributions of source code must retain the above copyright
  *    notice, this list of conditions and the following disclaimer.
  * 2.  Redistributions in binary form must reproduce the above copyright
  *    notice, this list of conditions and the following disclaimer in the
@@ skipping 32 matching lines @@
 // This basic implementation supports no more than one input and output.
 // 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 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) OVERRIDE;
     virtual void initialize() OVERRIDE;
     virtual void uninitialize() OVERRIDE;
 
     size_t bufferSize() const { return m_bufferSize; }
 
     DEFINE_ATTRIBUTE_EVENT_LISTENER(audioprocess);
 
 
 private:
     virtual double tailTime() const OVERRIDE;
     virtual double latencyTime() const OVERRIDE;
 
     ScriptProcessorNode(AudioContext*, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels);
 
     static void fireProcessEventDispatch(void* userData);
     void fireProcessEvent();
 
     // Double buffering
     unsigned doubleBufferIndex() const { return m_doubleBufferIndex; }
     void swapBuffers() { m_doubleBufferIndex = 1 - m_doubleBufferIndex; }
     unsigned m_doubleBufferIndex;
     unsigned m_doubleBufferIndexForEvent;
-    Vector<RefPtr<AudioBuffer> > m_inputBuffers;
-    Vector<RefPtr<AudioBuffer> > m_outputBuffers;
+    WillBePersistentHeapVector<RefPtrWillBeMember<AudioBuffer> > m_inputBuffers;

[Mads Ager (chromium), 2014/03/27 11:06:49]

This is an AudioNode and is therefore in the heap. You should just use
WillBeHeapVector<RefPtrWillBeMember<AudioBuffer> > here and you should then
implement a trace method that traces m_inputBuffers and m_outputBuffers.

[keishi, 2014/04/03 06:53:19]

Done.

+    WillBePersistentHeapVector<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;
 };
 
 } // namespace WebCore
 
 #endif // ScriptProcessorNode_h