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( |