Avoid hitting the thread check when WebRtcAudioRenderer is going away.

content/renderer/media/webrtc_audio_device_impl.cc

 // Copyright 2013 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_audio_device_impl.h"
 
 #include "base/bind.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/win/windows_version.h"
@@ skipping 157 matching lines @@
   for (PlayoutDataSinkList::const_iterator it = playout_sinks_.begin();
        it != playout_sinks_.end(); ++it) {
     (*it)->OnPlayoutData(audio_bus, sample_rate, audio_delay_milliseconds);
   }
 }
 
 void WebRtcAudioDeviceImpl::RemoveAudioRenderer(WebRtcAudioRenderer* renderer) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK_EQ(renderer, renderer_);
   base::AutoLock auto_lock(lock_);
+  // Notify the playout sink of the change.
+  for (PlayoutDataSinkList::const_iterator it = playout_sinks_.begin();
+       it != playout_sinks_.end(); ++it) {
+    (*it)->OnPlayoutDataSourceChanged();
+  }
+
   renderer_ = NULL;
   playing_ = false;
 }
 
 int32_t WebRtcAudioDeviceImpl::RegisterAudioCallback(
     webrtc::AudioTransport* audio_callback) {
   DVLOG(1) << "WebRtcAudioDeviceImpl::RegisterAudioCallback()";
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK_EQ(audio_transport_callback_ == NULL, audio_callback != NULL);
   audio_transport_callback_ = audio_callback;
@@ skipping 291 matching lines @@
   // If there is no capturer or there are more than one open capture devices,
   // return false.
   if (capturers_.empty() || capturers_.size() > 1)
     return false;
 
   return GetDefaultCapturer()->GetPairedOutputParameters(
       session_id, output_sample_rate, output_frames_per_buffer);
 }
 
 }  // namespace content