| 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..743ae02254831c788967be363956b798ce34ae5b 100644
|
| --- a/webrtc/call/rtp_stream_receiver_controller.cc
|
| +++ b/webrtc/call/rtp_stream_receiver_controller.cc
|
| @@ -46,7 +46,7 @@ bool RtpStreamReceiverController::OnRtpPacket(const RtpPacketReceived& packet) {
|
| void RtpStreamReceiverController::AddSink(uint32_t ssrc,
|
| RtpPacketSinkInterface* sink) {
|
| rtc::CritScope cs(&lock_);
|
| - return demuxer_.AddSink(ssrc, sink);
|
| + demuxer_.AddSink(ssrc, sink); // TODO(eladalon): Return-value useful here?
|
| }
|
|
|
| size_t RtpStreamReceiverController::RemoveSink(
|
|
|