Avoids deadlock in WebRtcLocalAudioRenderer

content/renderer/media/webrtc_local_audio_renderer.cc

 // Copyright (c) 2012 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_local_audio_renderer.h"
 
 #include "base/debug/trace_event.h"
 #include "base/logging.h"
 #include "base/message_loop_proxy.h"
 #include "base/synchronization/lock.h"
@@ skipping 159 matching lines @@
   source_->Start();
   sink_->Start();
 
   last_render_time_ = base::Time::Now();
   playing_ = false;
 }
 
 void WebRtcLocalAudioRenderer::Stop() {
   DVLOG(1) << "WebRtcLocalAudioRenderer::Stop()";
   DCHECK(thread_checker_.CalledOnValidThread());
-  base::AutoLock auto_lock(thread_lock_);
 
   if (!sink_)
     return;
 
+  {
+    base::AutoLock auto_lock(thread_lock_);
+    playing_ = false;
+
+    if (loopback_fifo_.get() != NULL) {
+      loopback_fifo_->Clear();
+      loopback_fifo_.reset();
+    }
+  }
+
   // Stop the output audio stream, i.e, stop asking for data to render.
   sink_->Stop();
   sink_ = NULL;
 
-  if (loopback_fifo_.get() != NULL) {
-    loopback_fifo_->Clear();
-    loopback_fifo_.reset();
-  }
-
   // Ensure that the capturer stops feeding us with captured audio.
   // Note that, we do not stop the capturer here since it may still be used by
   // the WebRTC ADM.
   source_->RemoveCapturerSink(this);
   source_ = NULL;
 
   if (audio_track_) {
     audio_track_->UnregisterObserver(this);
     audio_track_ = NULL;
   }
@@ skipping 44 matching lines @@
   if (!sink_)
     return base::TimeDelta();
   return total_render_time();
 }
 
 bool WebRtcLocalAudioRenderer::IsLocalRenderer() const {
   return true;
 }
 
 }  // namespace content