Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()

webrtc/call/call.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.
  */
 
@@ skipping 488 matching lines @@
   uint32_t send_bandwidth = 0;
   congestion_controller_->GetBitrateController()->AvailableBandwidth(
       &send_bandwidth);
   std::vector<unsigned int> ssrcs;
   uint32_t recv_bandwidth = 0;
   congestion_controller_->GetRemoteBitrateEstimator(false)->LatestEstimate(
       &ssrcs, &recv_bandwidth);
   stats.send_bandwidth_bps = send_bandwidth;
   stats.recv_bandwidth_bps = recv_bandwidth;
   stats.pacer_delay_ms = congestion_controller_->GetPacerQueuingDelayMs();
-  {
-    ReadLockScoped read_lock(*send_crit_);
-    // TODO(solenberg): Add audio send streams.
-    for (const auto& kv : video_send_ssrcs_) {
-      int rtt_ms = kv.second->GetRtt();
-      if (rtt_ms > 0)
-        stats.rtt_ms = rtt_ms;
-    }
-  }
+  stats.rtt_ms = call_stats_->rtcp_rtt_stats()->LastProcessedRtt();
   return stats;
 }
 
 void Call::SetBitrateConfig(
     const webrtc::Call::Config::BitrateConfig& bitrate_config) {
   TRACE_EVENT0("webrtc", "Call::SetBitrateConfig");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
   if (bitrate_config.max_bitrate_bps != -1)
     RTC_DCHECK_GT(bitrate_config.max_bitrate_bps, 0);
@@ skipping 209 matching lines @@
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc