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

webrtc/video/end_to_end_tests.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include <algorithm>
@@ skipping 2103 matching lines @@
       ASSERT_EQ(1u, receive_configs->size());
       RtpRtcp::Configuration config;
       config.receiver_only = true;
       config.clock = clock_;
       config.outgoing_transport = receive_transport_;
       config.retransmission_rate_limiter = &retransmission_rate_limiter_;
       rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
       rtp_rtcp_->SetRemoteSSRC((*receive_configs)[0].rtp.remote_ssrc);
       rtp_rtcp_->SetSSRC((*receive_configs)[0].rtp.local_ssrc);
       rtp_rtcp_->SetREMBStatus(true);
-      rtp_rtcp_->SetSendingStatus(true);
       rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
     }
 
     void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
       sender_call_ = sender_call;
     }
 
     static void BitrateStatsPollingThread(void* obj) {
       static_cast<BweObserver*>(obj)->PollStats();
     }
@@ skipping 2045 matching lines @@
     std::unique_ptr<VideoEncoder> encoder_;
     std::unique_ptr<VideoDecoder> decoder_;
     rtc::CriticalSection crit_;
     int recorded_frames_ GUARDED_BY(crit_);
   } test(this);
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc