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 { |