Fix issue with failing to call AudioBufferSource.onended in some cases

Source/modules/webaudio/AudioBufferSourceNode.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
  *    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 471 matching lines @@
 double AudioBufferSourceHandler::computePlaybackRate()
 {
     double dopplerRate = 1.0;
     if (m_pannerNode)
         dopplerRate = m_pannerNode->dopplerRate();
 
     // Incorporate buffer's sample-rate versus AudioContext's sample-rate.
     // Normally it's not an issue because buffers are loaded at the
     // AudioContext's sample-rate, but we can handle it in any case.
     double sampleRateFactor = 1.0;
-    if (buffer())
-        sampleRateFactor = buffer()->sampleRate() / sampleRate();
+    if (buffer()) {
+        // Use doubles to compute this to full accuracy.
+        sampleRateFactor = buffer()->sampleRate() / static_cast<double>(sampleRate());
+    }
 
     // Use finalValue() to incorporate changes of AudioParamTimeline and
     // AudioSummingJunction from m_playbackRate AudioParam.
     double basePlaybackRate = m_playbackRate->finalValue();
 
     double finalPlaybackRate = dopplerRate * sampleRateFactor * basePlaybackRate;
 
     // Sanity check the total rate.  It's very important that the resampler not
     // get any bad rate values.
     finalPlaybackRate = clampTo(finalPlaybackRate, 0.0, MaxRate);
@@ skipping 30 matching lines @@
         m_pannerNode->breakConnection();
         m_pannerNode.clear();
     }
 }
 
 void AudioBufferSourceHandler::handleStoppableSourceNode()
 {
     // If the source node is not looping, and we have a buffer, we can determine when the
     // source would stop playing.
     if (!loop() && buffer() && isPlayingOrScheduled()) {
-        double stopTime = m_startTime + buffer()->duration();
+        // See crbug.com/478301. If a source node is started via start(), the source won't start at
+        // that time but one quantum (128 frames) later.  But we compute the stop time based on the
+        // start time and the duration, so we end up stopping one quantum too early.  Thus, add one

[hongchan, 2015/04/21 21:39:08]

I think 'one quantum early' reads better.

[Raymond Toy, 2015/04/21 22:00:05]

Done.

+        // quantum to the stop time.  Then add one more just for good measure; we don't need to be
+        // extra precise here.  We just need to stop the source sometime after it should have
+        // stopped if it hadn't already.
+        // FIXME: Expose the rendering quantom somehow instead of hardwiring a value here.

[hongchan, 2015/04/21 21:39:08]

quantom -> quantum

[Raymond Toy, 2015/04/21 22:00:05]

Done.

+        double extraStopTime = 256 / static_cast<double>(context()->sampleRate());

[hongchan, 2015/04/21 21:39:08]

Can we use an extra variable for '256' at the top like what we do for the
smoothing constant and etc.?

[Raymond Toy, 2015/04/21 22:00:05]

Done.

+        double stopTime = m_startTime + buffer()->duration() + extraStopTime;
+
         if (context()->currentTime() > stopTime) {
             // The context time has passed the time when the source nodes should have stopped
             // playing. Stop the node now and deref it. (But don't run the onEnded event because the
             // source never actually played.)
             finishWithoutOnEnded();
         }
     }
 }
 
 void AudioBufferSourceHandler::finish()
@@ skipping 88 matching lines @@
 }
 
 void AudioBufferSourceNode::start(double when, double grainOffset, double grainDuration, ExceptionState& exceptionState)
 {
     audioBufferSourceHandler().start(when, grainOffset, grainDuration, exceptionState);
 }
 
 } // namespace blink
 
 #endif // ENABLE(WEB_AUDIO)