Don't allocate any RTPSender object for a receive only RtpRtcp module

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
 
+#include <memory>
 #include <set>
 #include <utility>
 #include <vector>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/gtest_prod_util.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
@@ skipping 276 matching lines @@
       const std::vector<uint16_t>& nack_sequence_numbers) override;
   void OnReceivedRtcpReportBlocks(
       const ReportBlockList& report_blocks) override;
   void OnRequestSendReport() override;
 
   void SetVideoBitrateAllocation(const BitrateAllocation& bitrate) override;
 
  protected:
   bool UpdateRTCPReceiveInformationTimers();
 
-  RTPSender* rtp_sender() { return &rtp_sender_; }
-  const RTPSender* rtp_sender() const { return &rtp_sender_; }
+  RTPSender* rtp_sender() { return rtp_sender_.get(); }
+  const RTPSender* rtp_sender() const { return rtp_sender_.get(); }
 
   RTCPSender* rtcp_sender() { return &rtcp_sender_; }
   const RTCPSender* rtcp_sender() const { return &rtcp_sender_; }
 
   RTCPReceiver* rtcp_receiver() { return &rtcp_receiver_; }
   const RTCPReceiver* rtcp_receiver() const { return &rtcp_receiver_; }
 
   const Clock* clock() const { return clock_; }
 
  private:
   FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
   FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
   int64_t RtcpReportInterval();
   void SetRtcpReceiverSsrcs(uint32_t main_ssrc);
 
   void set_rtt_ms(int64_t rtt_ms);
   int64_t rtt_ms() const;
 
   bool TimeToSendFullNackList(int64_t now) const;
 
-  RTPSender rtp_sender_;
+  std::unique_ptr<RTPSender> rtp_sender_;
   RTCPSender rtcp_sender_;
   RTCPReceiver rtcp_receiver_;
 
   const Clock* const clock_;
 
   const bool audio_;
   int64_t last_process_time_;
   int64_t last_bitrate_process_time_;
   int64_t last_rtt_process_time_;
   uint16_t packet_overhead_;
@@ skipping 13 matching lines @@
   PacketLossStats receive_loss_stats_;
 
   // The processed RTT from RtcpRttStats.
   rtc::CriticalSection critical_section_rtt_;
   int64_t rtt_ms_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_