Chromium Code Reviews| Index: webrtc/call/rtp_stream_receiver_controller.cc | 
| diff --git a/webrtc/call/rtp_stream_receiver_controller.cc b/webrtc/call/rtp_stream_receiver_controller.cc | 
| index a4b1e36ae262c1951479c36b57e07ea71a76041d..833ff0302d3db255cb3362a8c1f723961c69bc01 100644 | 
| --- a/webrtc/call/rtp_stream_receiver_controller.cc | 
| +++ b/webrtc/call/rtp_stream_receiver_controller.cc | 
| @@ -18,7 +18,7 @@ RtpStreamReceiverController::Receiver::Receiver( | 
| uint32_t ssrc, | 
| RtpPacketSinkInterface* sink) | 
| : controller_(controller), sink_(sink) { | 
| - controller_->AddSink(ssrc, sink_); | 
| + RTC_CHECK(controller_->AddSink(ssrc, sink_)); | 
| 
 
holmer
2017/07/03 09:06:14
Is this CHECK good given that this likely comes fr
 
eladalon
2017/07/03 11:25:35
Changed to a log. Do you think we should have a TO
 
 | 
| } | 
| RtpStreamReceiverController::Receiver::~Receiver() { | 
| @@ -43,7 +43,7 @@ bool RtpStreamReceiverController::OnRtpPacket(const RtpPacketReceived& packet) { | 
| return demuxer_.OnRtpPacket(packet); | 
| } | 
| -void RtpStreamReceiverController::AddSink(uint32_t ssrc, | 
| +bool RtpStreamReceiverController::AddSink(uint32_t ssrc, | 
| RtpPacketSinkInterface* sink) { | 
| rtc::CritScope cs(&lock_); | 
| return demuxer_.AddSink(ssrc, sink); |