| Index: webrtc/api/webrtcsession.cc
 | 
| diff --git a/webrtc/api/webrtcsession.cc b/webrtc/api/webrtcsession.cc
 | 
| index 28d2f63ef7ace44f618b2e79840f7d34fb4bbe93..1cf7924fb77f3e062a9fc907df4783cfdd10589c 100644
 | 
| --- a/webrtc/api/webrtcsession.cc
 | 
| +++ b/webrtc/api/webrtcsession.cc
 | 
| @@ -1241,7 +1241,7 @@ void WebRtcSession::SetRawAudioSink(uint32_t ssrc,
 | 
|    if (!voice_channel_)
 | 
|      return;
 | 
|  
 | 
| -  voice_channel_->SetRawAudioSink(ssrc, rtc::ScopedToUnique(std::move(sink)));
 | 
| +  voice_channel_->SetRawAudioSink(ssrc, std::move(sink));
 | 
|  }
 | 
|  
 | 
|  RtpParameters WebRtcSession::GetAudioRtpParameters(uint32_t ssrc) const {
 | 
| 
 |