Index: content/renderer/media/webrtc_local_audio_track.cc |
diff --git a/content/renderer/media/webrtc_local_audio_track.cc b/content/renderer/media/webrtc_local_audio_track.cc |
index dc54c00450899d837384c0db94020e4fd3ffcbc5..704d551ed727b787a740c9b3473d25957f92b5bd 100644 |
--- a/content/renderer/media/webrtc_local_audio_track.cc |
+++ b/content/renderer/media/webrtc_local_audio_track.cc |
@@ -38,7 +38,7 @@ WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() { |
DVLOG(1) << "WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack()"; |
// Users might not call Stop() on the track. |
- if (capturer_) |
+ if (capturer_.get()) |
Stop(); |
} |
@@ -129,14 +129,14 @@ void WebRtcLocalAudioTrack::RemoveSink( |
void WebRtcLocalAudioTrack::Start() { |
DCHECK(thread_checker_.CalledOnValidThread()); |
DVLOG(1) << "WebRtcLocalAudioTrack::Start()"; |
- if (capturer_) |
+ if (capturer_.get()) |
capturer_->AddSink(this); |
} |
void WebRtcLocalAudioTrack::Stop() { |
DCHECK(thread_checker_.CalledOnValidThread()); |
DVLOG(1) << "WebRtcLocalAudioTrack::Stop()"; |
- if (capturer_) { |
+ if (capturer_.get()) { |
capturer_->RemoveSink(this); |
capturer_ = NULL; |
} |