Supporting a playbackTime attribute on AudioprocessingEvent.

Source/modules/webaudio/ScriptProcessorNode.cpp

 /*
  * 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
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@@ skipping 247 matching lines @@
     AudioBuffer* outputBuffer = m_outputBuffers[m_doubleBufferIndexForEvent].get();
     ASSERT(outputBuffer);
     if (!outputBuffer)
         return;
 
     // 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;
 
+        // 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) / context()->sampleRate();

[Raymond Toy, 2014/03/28 18:14:14]

Cast context()->sampleRate() to double because it's a float. Playback time will
be slightly off then.

[KhNo, 2014/03/29 00:24:46]

I missed it. Thank you.

+
         // Call the JavaScript event handler which will do the audio processing.
-        dispatchEvent(AudioProcessingEvent::create(inputBuffer, outputBuffer));
+        dispatchEvent(AudioProcessingEvent::create(inputBuffer, outputBuffer, playbackTime));
     }
 }
 
 double ScriptProcessorNode::tailTime() const
 {
     return std::numeric_limits<double>::infinity();
 }
 
 double ScriptProcessorNode::latencyTime() const
 {
     return std::numeric_limits<double>::infinity();
 }
 
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)