| Index: content/renderer/media/webrtc_local_audio_renderer.cc | 
| diff --git a/content/renderer/media/webrtc_local_audio_renderer.cc b/content/renderer/media/webrtc_local_audio_renderer.cc | 
| index addf9637a5ff68964a0cdfc471433d028de5981e..ac44a210be3a655320f58b3219838cef2eb2c077 100644 | 
| --- a/content/renderer/media/webrtc_local_audio_renderer.cc | 
| +++ b/content/renderer/media/webrtc_local_audio_renderer.cc | 
| @@ -153,7 +153,7 @@ void WebRtcLocalAudioRenderer::Stop() { | 
| // Stop the output audio stream, i.e, stop asking for data to render. | 
| // It is safer to call Stop() on the |sink_| to clean up the resources even | 
| // when the |sink_| is never started. | 
| -  if (sink_) { | 
| +  if (sink_.get()) { | 
| sink_->Stop(); | 
| sink_ = NULL; | 
| } | 
| @@ -316,7 +316,7 @@ void WebRtcLocalAudioRenderer::ReconfigureSink( | 
| loopback_fifo_.reset(new_fifo); | 
| } | 
|  | 
| -  if (!sink_) | 
| +  if (!sink_.get()) | 
| return;  // WebRtcLocalAudioRenderer has not yet been started. | 
|  | 
| // Stop |sink_| and re-create a new one to be initialized with different audio | 
|  |