SSRC and RSID may only refer to one sink each in RtpDemuxer

webrtc/call/rtp_stream_receiver_controller.cc

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..2097bacecfe08400a8270e2752fe4f4398941eef 100644
--- a/webrtc/call/rtp_stream_receiver_controller.cc
+++ b/webrtc/call/rtp_stream_receiver_controller.cc
@@ -9,6 +9,8 @@
  */
 
 #include "webrtc/call/rtp_stream_receiver_controller.h"
+
+#include "webrtc/base/logging.h"
 #include "webrtc/base/ptr_util.h"
 
 namespace webrtc {
@@ -18,7 +20,11 @@ RtpStreamReceiverController::Receiver::Receiver(
     uint32_t ssrc,
     RtpPacketSinkInterface* sink)
     : controller_(controller), sink_(sink) {
-  controller_->AddSink(ssrc, sink_);
+  const bool sink_added = controller_->AddSink(ssrc, sink_);
+  if (!sink_added) {
+    LOG(LS_ERROR) << "RtpStreamReceiverController::Receiver::Receiver: Sink "
+                  << "could not be added for SSRC=" << ssrc << ".";
+  }
 }
 
 RtpStreamReceiverController::Receiver::~Receiver() {
@@ -43,7 +49,7 @@ 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);