Set actual transport overhead in rtp_rtcp

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 660 matching lines @@
                                  const rtc::NetworkRoute& network_route) {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   // Check if the network route is connected.
   if (!network_route.connected) {
     LOG(LS_INFO) << "Transport " << transport_name << " is disconnected";
     // TODO(honghaiz): Perhaps handle this in SignalChannelNetworkState and
     // consider merging these two methods.
     return;
   }
 
+  {
+    ReadLockScoped read_lock(*send_crit_);
+    for (auto& kv : audio_send_ssrcs_) {
+      kv.second->SetTransportOverhead(
+          network_route.transport_overhead_per_packet);
+    }
+
+    for (auto& kv : video_send_ssrcs_) {
+      kv.second->SetTransportOverhead(
+          network_route.transport_overhead_per_packet);
+    }
+  }
+
   // Check whether the network route has changed on each transport.
   auto result =
       network_routes_.insert(std::make_pair(transport_name, network_route));
   auto kv = result.first;
   bool inserted = result.second;
   if (inserted) {
     // No need to reset BWE if this is the first time the network connects.
     return;
   }
   if (kv->second != network_route) {
@@ skipping 261 matching lines @@
   uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
   ReadLockScoped read_lock(*receive_crit_);
   auto it = video_receive_ssrcs_.find(ssrc);
   if (it == video_receive_ssrcs_.end())
     return false;
   return it->second->OnRecoveredPacket(packet, length);
 }
 
 }  // namespace internal
 }  // namespace webrtc