Reland of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2993633002/ )

Reason for revert:
Relanding

Original issue's description:
> Revert of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #15 id:280001 of https://codereview.webrtc.org/2968693002/ )
>
> Reason for revert:
> Some internal tests keep failing after this change. Try to fix it by reverting it. Will reland it if this isn't the root cause.
>
> Original issue's description:
> > SSRC and RSID may only refer to one sink each in RtpDemuxer
> >
> > RTP demuxing should only match RTP packets with one sink.
> >
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/2968693002
> > Cr-Commit-Position: refs/heads/master@{#19233}
> > Committed: https://chromium.googlesource.com/external/webrtc/+/7b7e06fd23ac67d81f378b773bb631abb1d82116
>
> TBR=nisse@webrtc.org,danilchap@webrtc.org,perkj@webrtc.org,stefan@webrtc.org,holmer@google.com,deadbeef@webrtc.org,pthatcher@webrtc.org,steveanton@webrtc.org,eladalon@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2993633002
> Cr-Commit-Position: refs/heads/master@{#19239}
> Committed: https://chromium.googlesource.com/external/webrtc/+/59b603fbed5b069090f9084c8eeb82eff7bca30c

TBR=nisse@webrtc.org,danilchap@webrtc.org,perkj@webrtc.org,stefan@webrtc.org,holmer@google.com,deadbeef@webrtc.org,pthatcher@webrtc.org,steveanton@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2993053002
Cr-Commit-Position: refs/heads/master@{#19248}

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 1238665..94fa83b 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/rtc_base/logging.h"
 #include "webrtc/rtc_base/ptr_util.h"
 
 namespace webrtc {
@@ -18,7 +20,11 @@
     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 @@
   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);