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

webrtc/video/rtp_stream_receiver.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 55 matching lines @@
   configuration.transport_sequence_number_allocator =
       transport_sequence_number_allocator;
   configuration.send_bitrate_observer = nullptr;
   configuration.send_frame_count_observer = nullptr;
   configuration.send_side_delay_observer = nullptr;
   configuration.send_packet_observer = nullptr;
   configuration.bandwidth_callback = nullptr;
   configuration.transport_feedback_callback = nullptr;
 
   std::unique_ptr<RtpRtcp> rtp_rtcp(RtpRtcp::CreateRtpRtcp(configuration));
-  rtp_rtcp->SetSendingStatus(false);
-  rtp_rtcp->SetSendingMediaStatus(false);
   rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
 
   return rtp_rtcp;
 }
 
 static const int kPacketLogIntervalMs = 10000;
 
 RtpStreamReceiver::RtpStreamReceiver(
     Transport* transport,
     RtcpRttStats* rtt_stats,
@@ skipping 563 matching lines @@
     return;
 
   if (!sprop_decoder.DecodeSprop(sprop_base64_it->second.c_str()))
     return;
 
   tracker_.InsertSpsPpsNalus(sprop_decoder.sps_nalu(),
                              sprop_decoder.pps_nalu());
 }
 
 }  // namespace webrtc