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

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 652cfb0294906fd68095e28ab0ee42e495e3ee71..e42943f426304d945095bf5041abe08975f8ab1e 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -506,15 +506,7 @@ Call::Stats Call::GetStats() const {
   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;
 }