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

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 277 matching lines @@
   rtc::CritScope lock(&critical_section_rtcp_sender_);
   last_rtp_timestamp_ = rtp_timestamp;
   if (capture_time_ms < 0) {
     // We don't currently get a capture time from VoiceEngine.
     last_frame_capture_time_ms_ = clock_->TimeInMilliseconds();
   } else {
     last_frame_capture_time_ms_ = capture_time_ms;
   }
 }
 
+uint32_t RTCPSender::SSRC() const {
+  rtc::CritScope lock(&critical_section_rtcp_sender_);
+  return ssrc_;
+}
+
 void RTCPSender::SetSSRC(uint32_t ssrc) {
   rtc::CritScope lock(&critical_section_rtcp_sender_);
 
   if (ssrc_ != 0) {
     // not first SetSSRC, probably due to a collision
     // schedule a new RTCP report
     // make sure that we send a RTP packet
     next_time_to_send_rtcp_ = clock_->TimeInMilliseconds() + 100;
   }
   ssrc_ = ssrc;
@@ skipping 694 matching lines @@
     max_packet_size = max_packet_size_;
   }
 
   RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
   uint8_t buffer[IP_PACKET_SIZE];
   return packet.BuildExternalBuffer(buffer, max_packet_size, &sender) &&
          !sender.send_failure_;
 }
 
 }  // namespace webrtc