| 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;
|
| }
|
|
|
|
|