Avoid AudioBufferSourceHandler data race.

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 572 matching lines @@
   // get any bad rate values.
   final_playback_rate = clampTo(final_playback_rate, 0.0, kMaxRate);
 
   bool is_playback_rate_valid =
       !std::isnan(final_playback_rate) && !std::isinf(final_playback_rate);
   DCHECK(is_playback_rate_valid);
 
   if (!is_playback_rate_valid)
     final_playback_rate = 1.0;
 
-  // Record the minimum playback rate for use by handleStoppableSourceNode.
-  min_playback_rate_ = std::min(final_playback_rate, min_playback_rate_);
+  // Record the minimum playback rate for use by HandleStoppableSourceNode.
+  if (final_playback_rate < min_playback_rate_) {
+    MutexLocker locker(min_playback_rate_mutex_);
+    min_playback_rate_ = final_playback_rate;
+  }
 
   return final_playback_rate;
 }
 
+double AudioBufferSourceHandler::GetMinPlaybackRate() {
+  DCHECK(IsMainThread());
+  MutexLocker locker(min_playback_rate_mutex_);
+  return min_playback_rate_;
+}
+
 bool AudioBufferSourceHandler::PropagatesSilence() const {
   return !IsPlayingOrScheduled() || HasFinished() || !buffer_;
 }
 
 void AudioBufferSourceHandler::HandleStoppableSourceNode() {
   // If the source node is not looping, and we have a buffer, we can determine
   // when the source would stop playing.  This is intended to handle the
   // (uncommon) scenario where start() has been called but is never connected to
   // the destination (directly or indirectly).  By stopping the node, the node
   // can be collected.  Otherwise, the node will never get collected, leaking
   // memory.
   //
   // If looping was ever done (m_didSetLooping = true), give up.  We can't
   // easily determine how long we looped so we don't know the actual duration
   // thus far, so don't try to do anything fancy.
+  double min_playback_rate = GetMinPlaybackRate();
   if (!DidSetLooping() && Buffer() && IsPlayingOrScheduled() &&
-      min_playback_rate_ > 0) {
+      min_playback_rate > 0) {
     // Adjust the duration to include the playback rate. Only need to account
     // for rate < 1 which makes the sound last longer.  For rate >= 1, the
     // source stops sooner, but that's ok.
-    double actual_duration = Buffer()->duration() / min_playback_rate_;
+    double actual_duration = Buffer()->duration() / min_playback_rate;
 
     double stop_time = start_time_ + actual_duration;
 
     // See crbug.com/478301. If a source node is started via start(), the source
     // may not 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 early.  Thus, add a little extra time; we just
     // need to stop the source sometime after it should have stopped if it
     // hadn't already.  We don't need to be super precise on when to stop.
     double extra_stop_time =
@@ skipping 126 matching lines @@
 
 void AudioBufferSourceNode::start(double when,
                                   double grain_offset,
                                   double grain_duration,
                                   ExceptionState& exception_state) {
   GetAudioBufferSourceHandler().Start(when, grain_offset, grain_duration,
                                       exception_state);
 }
 
 }  // namespace blink