Fix the race when the WebRtcAudioDeviceImpl goes away before capturers stop

content/renderer/media/webrtc/peer_connection_dependency_factory.cc

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
 
 #include <vector>
 
 #include "base/command_line.h"
 #include "base/strings/utf_string_conversions.h"
@@ skipping 429 matching lines @@
   StartLocalAudioTrack(audio_track.get());
 
   // Pass the ownership of the native local audio track to the blink track.
   blink::WebMediaStreamTrack writable_track = track;
   writable_track.setExtraData(audio_track.release());
 }
 
 void PeerConnectionDependencyFactory::StartLocalAudioTrack(
     WebRtcLocalAudioTrack* audio_track) {
   // Add the WebRtcAudioDevice as the sink to the local audio track.
-  // TODO(xians): Implement a PeerConnection sink adapter and remove this
-  // AddSink() call.
-  audio_track->AddSink(GetWebRtcAudioDevice());
+  // TODO(xians): Remove the following line of code after the APM in WebRTC is
+  // completely deprecated. See http://crbug/365672.
+  if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled())
+    audio_track->AddSink(GetWebRtcAudioDevice());
+
   // Start the audio track. This will hook the |audio_track| to the capturer
   // as the sink of the audio, and only start the source of the capturer if
   // it is the first audio track connecting to the capturer.
   audio_track->Start();
 }
 
 scoped_refptr<WebAudioCapturerSource>
 PeerConnectionDependencyFactory::CreateWebAudioSource(
     blink::WebMediaStreamSource* source) {
   DVLOG(1) << "PeerConnectionDependencyFactory::CreateWebAudioSource()";
@@ skipping 200 matching lines @@
 }
 
 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
   if (audio_device_)
     return;
 
   audio_device_ = new WebRtcAudioDeviceImpl();
 }
 
 }  // namespace content