Avoid AudioBufferSourceHandler data race.

Following r478084, the main thread may contend with the audio thread
on accessing AudioBufferSourceNodeHandler's mutable state. Coordinate
such access by introducing a Mutex over |min_playback_rate_|.

Using atomic ops would be the natural choice for handling this, but
steering clear of those over doubles (cf. https://crrev.com/1256053006)
until std::atomic<> is allowed.

R=haraken,hongchan
BUG=731568

Review-Url: https://codereview.chromium.org/2929283002
Cr-Commit-Position: refs/heads/master@{#478679}

third_party/WebKit/Source/modules/webaudio/AudioBufferSourceNode.cpp

@@ -590,12 +590,21 @@ double AudioBufferSourceHandler::ComputePlaybackRate() {
   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_;
 }
@@ -611,12 +620,13 @@ void AudioBufferSourceHandler::HandleStoppableSourceNode() {
   // 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;
 

third_party/WebKit/Source/modules/webaudio/AudioBufferSourceNode.h

@@ -168,8 +168,18 @@ class AudioBufferSourceHandler final : public AudioScheduledSourceHandler {
   // conversion factor, and the value of playbackRate and detune AudioParams.
   double ComputePlaybackRate();
 
+  double GetMinPlaybackRate();
+
   // The minimum playbackRate value ever used for this source.
   double min_playback_rate_;
+
+  // |min_playback_rate_| may be updated by the audio thread
+  // while the main thread checks if the node is in a stoppable
+  // state, hence access needs to be atomic.
+  //
+  // TODO: when the codebase adopts std::atomic<>, use it for
+  // |min_playback_rate_|.
+  Mutex min_playback_rate_mutex_;
 };
 
 class AudioBufferSourceNode final : public AudioScheduledSourceNode {