Move Ownership of RtpModules to VideoSendStream.

webrtc/modules/rtp_rtcp/source/rtcp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index 95bfeeea1f6ba6c564109abda64174f3ffa42188..7579a36edcbb46c04d9963e6800e297cc5306461 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -871,11 +871,13 @@ void RTCPSender::PrepareReport(const std::set<RTCPPacketType>& packetTypes,
         random_.Rand(minIntervalMs * 1 / 2, minIntervalMs * 3 / 2);
     next_time_to_send_rtcp_ = clock_->TimeInMilliseconds() + timeToNext;
 
-    StatisticianMap statisticians =
-        receive_statistics_->GetActiveStatisticians();
-    RTC_DCHECK(report_blocks_.empty());
-    for (auto& it : statisticians) {
-      AddReportBlock(feedback_state, it.first, it.second);
+    if (receive_statistics_) {
+      StatisticianMap statisticians =
+          receive_statistics_->GetActiveStatisticians();
+      RTC_DCHECK(report_blocks_.empty());
+      for (auto& it : statisticians) {
+        AddReportBlock(feedback_state, it.first, it.second);
+      }
     }
   }
 }