Set rtcp_send_transport for AudioReceiveStreams. This was forgotten in https://codereview.webrtc.or…

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 1244 matching lines @@
 };
 
 class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
  public:
   WebRtcAudioReceiveStream(int ch,
                            uint32_t remote_ssrc,
                            uint32_t local_ssrc,
                            bool use_transport_cc,
                            const std::string& sync_group,
                            const std::vector<webrtc::RtpExtension>& extensions,
-                           webrtc::Call* call)
+                           webrtc::Call* call,
+                           webrtc::Transport* rtcp_send_transport)
       : call_(call), config_() {
     RTC_DCHECK_GE(ch, 0);
     RTC_DCHECK(call);
     config_.rtp.remote_ssrc = remote_ssrc;
     config_.rtp.local_ssrc = local_ssrc;
+    config_.rtcp_send_transport = rtcp_send_transport;
     config_.voe_channel_id = ch;
     config_.sync_group = sync_group;
     RecreateAudioReceiveStream(use_transport_cc, extensions);
   }
 
   ~WebRtcAudioReceiveStream() {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     call_->DestroyAudioReceiveStream(stream_);
   }
 
@@ skipping 811 matching lines @@
     // can obtain RTT from the send channel)
     engine()->voe()->base()->AssociateSendChannel(channel, send_channel);
     LOG(LS_INFO) << "VoiceEngine channel #" << channel
                  << " is associated with channel #" << send_channel << ".";
   }
 
   recv_streams_.insert(std::make_pair(
       ssrc, new WebRtcAudioReceiveStream(channel, ssrc, receiver_reports_ssrc_,
                                          recv_transport_cc_enabled_,
                                          sp.sync_label, recv_rtp_extensions_,
-                                         call_)));
+                                         call_, this)));
 
   SetNack(channel, send_codec_spec_.nack_enabled);
   SetPlayout(channel, playout_);
 
   return true;
 }
 
 bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
   TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::RemoveRecvStream");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
@@ skipping 446 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE