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

webrtc/call/rtp_stream_receiver_controller.cc

 /*
  *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/call/rtp_stream_receiver_controller.h"
+
+#include "webrtc/rtc_base/logging.h"
 #include "webrtc/rtc_base/ptr_util.h"
 
 namespace webrtc {
 
 RtpStreamReceiverController::Receiver::Receiver(
     RtpStreamReceiverController* controller,
     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() {
   // Don't require return value > 0, since for RTX we currently may
   // have multiple Receiver objects with the same sink.
   // TODO(nisse): Consider adding a DCHECK when RtxReceiveStream is wired up.
   controller_->RemoveSink(sink_);
 }
 
 RtpStreamReceiverController::RtpStreamReceiverController() = default;
 RtpStreamReceiverController::~RtpStreamReceiverController() = default;
 
 std::unique_ptr<RtpStreamReceiverInterface>
 RtpStreamReceiverController::CreateReceiver(
     uint32_t ssrc,
     RtpPacketSinkInterface* sink) {
   return rtc::MakeUnique<Receiver>(this, ssrc, sink);
 }
 
 bool RtpStreamReceiverController::OnRtpPacket(const RtpPacketReceived& packet) {
   rtc::CritScope cs(&lock_);
   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);
 }
 
 size_t RtpStreamReceiverController::RemoveSink(
     const RtpPacketSinkInterface* sink) {
   rtc::CritScope cs(&lock_);
   return demuxer_.RemoveSink(sink);
 }
 
 }  // namespace webrtc