Remove unnecessary attribute "gain" in AudioBuffer and AudioBufferSourceNode interfaces

Source/modules/webaudio/AudioBufferSourceNode.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 61 matching lines @@
     // The old attribute is kept for backwards compatibility.
     bool loop() const { return m_isLooping; }
     void setLoop(bool looping) { m_isLooping = looping; }
 
     // Loop times in seconds.
     double loopStart() const { return m_loopStart; }
     double loopEnd() const { return m_loopEnd; }
     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.
     void setPannerNode(PannerNode*);
     void clearPannerNode();
 
     // If we are no longer playing, propogate silence ahead to downstream nodes.
     virtual bool propagatesSilence() const OVERRIDE;
 
     // AudioScheduledSourceNode
     virtual void finish() OVERRIDE;
 
 private:
     AudioBufferSourceNode(AudioContext*, float sampleRate);
 
     // Returns true on success.
     bool renderFromBuffer(AudioBus*, unsigned destinationFrameOffset, size_t numberOfFrames);
 
     // Render silence starting from "index" frame in AudioBus.
     inline bool renderSilenceAndFinishIfNotLooping(AudioBus*, unsigned index, size_t framesToProcess);
 
     // m_buffer holds the sample data which this node outputs.
     RefPtr<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;
+    // Used for the "playbackRate" attributes.
     RefPtr<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.
     bool m_isLooping;
 
     double m_loopStart;
     double m_loopEnd;
 
     // m_virtualReadIndex is a sample-frame index into our buffer representing the current playback position.
     // Since it's floating-point, it has sub-sample accuracy.
     double m_virtualReadIndex;
 
     // Granular playback
     bool m_isGrain;
     double m_grainOffset; // in seconds
     double m_grainDuration; // in seconds
 
     // totalPitchRate() returns the instantaneous pitch rate (non-time preserving).
     // 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;
 
     // This synchronizes process() with setBuffer() which can cause dynamic channel count changes.
     mutable Mutex m_processLock;
 };
 
 } // namespace WebCore
 
 #endif // AudioBufferSourceNode_h