Register FlexFEC SSRC to receive RTCP on sending side.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 217 matching lines @@
 void ModuleRtpRtcpImpl::SetRtxSsrc(uint32_t ssrc) {
   rtp_sender_->SetRtxSsrc(ssrc);
 }
 
 void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type,
                                               int associated_payload_type) {
   rtp_sender_->SetRtxPayloadType(payload_type, associated_payload_type);
 }
 
 rtc::Optional<uint32_t> ModuleRtpRtcpImpl::FlexfecSsrc() const {
-  return rtp_sender_->FlexfecSsrc();
+  if (rtp_sender_)
+    return rtp_sender_->FlexfecSsrc();
+  return rtc::Optional<uint32_t>();
 }
 
 int32_t ModuleRtpRtcpImpl::IncomingRtcpPacket(
     const uint8_t* rtcp_packet,
     const size_t length) {
   return rtcp_receiver_.IncomingPacket(rtcp_packet, length) ? 0 : -1;
 }
 
 int32_t ModuleRtpRtcpImpl::RegisterSendPayload(
     const CodecInst& voice_codec) {
@@ skipping 609 matching lines @@
     return RTCP_INTERVAL_AUDIO_MS;
   else
     return RTCP_INTERVAL_VIDEO_MS;
 }
 
 void ModuleRtpRtcpImpl::SetRtcpReceiverSsrcs(uint32_t main_ssrc) {
   std::set<uint32_t> ssrcs;
   ssrcs.insert(main_ssrc);
   if (RtxSendStatus() != kRtxOff)
     ssrcs.insert(rtp_sender_->RtxSsrc());
+  rtc::Optional<uint32_t> flexfec_ssrc = FlexfecSsrc();
+  if (flexfec_ssrc)
+    ssrcs.insert(*flexfec_ssrc);
   rtcp_receiver_.SetSsrcs(main_ssrc, ssrcs);
 }
 
 void ModuleRtpRtcpImpl::set_rtt_ms(int64_t rtt_ms) {
   rtc::CritScope cs(&critical_section_rtt_);
   rtt_ms_ = rtt_ms;
 }
 
 int64_t ModuleRtpRtcpImpl::rtt_ms() const {
   rtc::CritScope cs(&critical_section_rtt_);
   return rtt_ms_;
 }
 
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_->RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_->GetRtpStatisticsCallback();
 }
 
 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
     const BitrateAllocation& bitrate) {
   rtcp_sender_.SetVideoBitrateAllocation(bitrate);
 }
 }  // namespace webrtc