| Index: webrtc/api/rtpsender.cc
 | 
| diff --git a/webrtc/api/rtpsender.cc b/webrtc/api/rtpsender.cc
 | 
| index 1ca8fc1d0a8e16034772f567164367bcfecd7439..da5bce4fd7d0847e06390d1e0ac12d2fca8ee17c 100644
 | 
| --- a/webrtc/api/rtpsender.cc
 | 
| +++ b/webrtc/api/rtpsender.cc
 | 
| @@ -36,7 +36,7 @@ void LocalAudioSinkAdapter::OnData(const void* audio_data,
 | 
|    }
 | 
|  }
 | 
|  
 | 
| -void LocalAudioSinkAdapter::SetSink(cricket::AudioRenderer::Sink* sink) {
 | 
| +void LocalAudioSinkAdapter::SetSink(cricket::AudioSource::Sink* sink) {
 | 
|    rtc::CritScope lock(&lock_);
 | 
|    ASSERT(!sink || !sink_);
 | 
|    sink_ = sink;
 | 
| @@ -194,9 +194,9 @@ void AudioRtpSender::SetAudioSend() {
 | 
|    }
 | 
|  #endif
 | 
|  
 | 
| -  cricket::AudioRenderer* renderer = sink_adapter_.get();
 | 
| -  ASSERT(renderer != nullptr);
 | 
| -  provider_->SetAudioSend(ssrc_, track_->enabled(), options, renderer);
 | 
| +  cricket::AudioSource* source = sink_adapter_.get();
 | 
| +  ASSERT(source != nullptr);
 | 
| +  provider_->SetAudioSend(ssrc_, track_->enabled(), options, source);
 | 
|  }
 | 
|  
 | 
|  VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
 | 
| 
 |