Revert of Revert "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://…

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 618 matching lines @@
   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) {
     kv->second = network_route;
     LOG(LS_INFO) << "Network route changed on transport " << transport_name
                  << ": new local network id " << network_route.local_network_id
-                 << " new remote network id " << network_route.remote_network_id
-                 << " Reset bitrate to "
-                 << config_.bitrate_config.start_bitrate_bps << "bps";
-    congestion_controller_->ResetBweAndBitrates(
-        config_.bitrate_config.start_bitrate_bps,
-        config_.bitrate_config.min_bitrate_bps,
-        config_.bitrate_config.max_bitrate_bps);
+                 << " new remote network id "
+                 << network_route.remote_network_id;
+    // TODO(holmer): Update the BWE bitrates.
   }
 }
 
 void Call::UpdateAggregateNetworkState() {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
   bool have_audio = false;
   bool have_video = false;
   {
     ReadLockScoped read_lock(*send_crit_);
@@ skipping 201 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