Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.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 169 matching lines @@
     // Process RTT if we have received a receiver report and we haven't
     // processed RTT for at least |kRtpRtcpRttProcessTimeMs| milliseconds.
     if (rtcp_receiver_.LastReceivedReceiverReport() >
         last_rtt_process_time_ && process_rtt) {
       std::vector<RTCPReportBlock> receive_blocks;
       rtcp_receiver_.StatisticsReceived(&receive_blocks);
       int64_t max_rtt = 0;
       for (std::vector<RTCPReportBlock>::iterator it = receive_blocks.begin();
            it != receive_blocks.end(); ++it) {
         int64_t rtt = 0;
-        rtcp_receiver_.RTT(it->remoteSSRC, &rtt, NULL, NULL, NULL);
+        rtcp_receiver_.RTT(it->sender_ssrc, &rtt, NULL, NULL, NULL);
         max_rtt = (rtt > max_rtt) ? rtt : max_rtt;
       }
       // Report the rtt.
       if (rtt_stats_ && max_rtt != 0)
         rtt_stats_->OnRttUpdate(max_rtt);
     }
 
     // Verify receiver reports are delivered and the reported sequence number
     // is increasing.
     int64_t rtcp_interval = RtcpReportInterval();
@@ skipping 710 matching lines @@
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_->GetRtpStatisticsCallback();
 }
 
 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
     const BitrateAllocation& bitrate) {
   rtcp_sender_.SetVideoBitrateAllocation(bitrate);
 }
 }  // namespace webrtc