| Index: webrtc/api/rtpsender.cc | 
| diff --git a/webrtc/api/rtpsender.cc b/webrtc/api/rtpsender.cc | 
| index c5db92946d200ab0d643107cfa871fe4e6c519a6..822b7f457e47e96506f13303a002b614320c6624 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, | 
|  |