Signal exceptions if start/stop is called out-of-order or too many times.

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 43 matching lines @@
 
     // setBuffer() is called on the main thread. This is the buffer we use for playback.
     void setBuffer(AudioBuffer*, ExceptionState&);
     AudioBuffer* buffer() { return m_buffer.get(); }
 
     // numberOfChannels() returns the number of output channels.  This value equals the number of channels from the buffer.
     // If a new buffer is set with a different number of channels, then this value will dynamically change.
     unsigned numberOfChannels();
 
     // Play-state
-    void start(double when = 0);
-    void start(double when, double grainOffset);
-    void start(double when, double grainOffset, double grainDuration);
+    void start(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);
+    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.
     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;
 
     // AudioScheduledSourceNode
     virtual void finish() OVERRIDE;
 
 private:
     AudioBufferSourceNode(AudioContext*, float sampleRate);
 
-    void startPlaying(bool isGrain, double when, double grainOffset, double grainDuration);
+    void startPlaying(bool isGrain, double when, double grainOffset, double grainDuration, ExceptionState&);
 
     // 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;
 
@@ skipping 32 matching lines @@
     // 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