| 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 {
|
|
|