Index: content/renderer/media/webrtc_audio_device_impl.cc |
diff --git a/content/renderer/media/webrtc_audio_device_impl.cc b/content/renderer/media/webrtc_audio_device_impl.cc |
index 1339a6f4cb80161efb228271aaad2830eecec5ff..2e6eed649d5b7c4054565870da03792248bd45c0 100644 |
--- a/content/renderer/media/webrtc_audio_device_impl.cc |
+++ b/content/renderer/media/webrtc_audio_device_impl.cc |
@@ -202,7 +202,7 @@ void WebRtcAudioDeviceImpl::RenderData(media::AudioBus* audio_bus, |
void WebRtcAudioDeviceImpl::RemoveAudioRenderer(WebRtcAudioRenderer* renderer) { |
DCHECK(thread_checker_.CalledOnValidThread()); |
- DCHECK_EQ(renderer, renderer_); |
+ DCHECK_EQ(renderer, renderer_.get()); |
base::AutoLock auto_lock(lock_); |
// Notify the playout sink of the change. |
for (PlayoutDataSinkList::const_iterator it = playout_sinks_.begin(); |
@@ -401,7 +401,7 @@ int32_t WebRtcAudioDeviceImpl::MinMicrophoneVolume(uint32_t* min_volume) const { |
int32_t WebRtcAudioDeviceImpl::StereoPlayoutIsAvailable(bool* available) const { |
DCHECK(initialized_); |
- *available = renderer_ && renderer_->channels() == 2; |
+ *available = renderer_.get() && renderer_->channels() == 2; |
return 0; |
} |
@@ -444,7 +444,7 @@ int32_t WebRtcAudioDeviceImpl::RecordingSampleRate( |
int32_t WebRtcAudioDeviceImpl::PlayoutSampleRate( |
uint32_t* sample_rate) const { |
- *sample_rate = renderer_ ? renderer_->sample_rate() : 0; |
+ *sample_rate = renderer_.get() ? renderer_->sample_rate() : 0; |
return 0; |
} |