Support for AudioContextOptions latencyHint.

third_party/WebKit/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 543 matching lines @@
 }
 
 double AudioBufferSourceHandler::computePlaybackRate() {
   // Incorporate buffer's sample-rate versus BaseAudioContext's sample-rate.
   // Normally it's not an issue because buffers are loaded at the
   // BaseAudioContext's sample-rate, but we can handle it in any case.
   double sampleRateFactor = 1.0;
   if (buffer()) {
     // Use doubles to compute this to full accuracy.
     sampleRateFactor =
-        buffer()->sampleRate() / static_cast<double>(sampleRate());
+        buffer()->sampleRate() / static_cast<double>(context()->sampleRate());
   }
 
   // Use finalValue() to incorporate changes of AudioParamTimeline and
   // AudioSummingJunction from m_playbackRate AudioParam.
   double basePlaybackRate = m_playbackRate->finalValue();
 
   double finalPlaybackRate = sampleRateFactor * basePlaybackRate;
 
   // Take the detune value into account for the final playback rate.
   finalPlaybackRate *= pow(2, m_detune->finalValue() / 1200);
@@ skipping 179 matching lines @@
 
 void AudioBufferSourceNode::start(double when,
                                   double grainOffset,
                                   double grainDuration,
                                   ExceptionState& exceptionState) {
   audioBufferSourceHandler().start(when, grainOffset, grainDuration,
                                    exceptionState);
 }
 
 }  // namespace blink