Move Ownership of RtpModules to VideoSendStream.

webrtc/modules/rtp_rtcp/source/rtcp_sender.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 853 matching lines @@
       }
       if (minIntervalMs > RTCP_INTERVAL_VIDEO_MS)
         minIntervalMs = RTCP_INTERVAL_VIDEO_MS;
     }
     // The interval between RTCP packets is varied randomly over the
     // range [1/2,3/2] times the calculated interval.
     uint32_t timeToNext =
         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);
+      }
     }
   }
 }
 
 bool RTCPSender::AddReportBlock(const FeedbackState& feedback_state,
                                 uint32_t ssrc,
                                 StreamStatistician* statistician) {
   // Do we have receive statistics to send?
   RtcpStatistics stats;
   if (!statistician->GetStatistics(&stats, true))
@@ skipping 149 matching lines @@
     // but we can't because of an incorrect warning (C4822) in MVS 2013.
   } sender(transport_, event_log_);
 
   RTC_DCHECK_LE(max_payload_length_, static_cast<size_t>(IP_PACKET_SIZE));
   uint8_t buffer[IP_PACKET_SIZE];
   return packet.BuildExternalBuffer(buffer, max_payload_length_, &sender) &&
          !sender.send_failure_;
 }
 
 }  // namespace webrtc