Set actual transport overhead in rtp_rtcp

webrtc/video/video_send_stream_tests.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.
  */
 #include <algorithm>  // max
@@ skipping 1241 matching lines @@
 
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       if (call_->GetStats().send_bandwidth_bps > kStartBitrateBps) {
         observation_complete_.Set();
       }
 
       return SEND_PACKET;
     }
 
     void PerformTest() override {
-      rtc::NetworkRoute new_route(true, 10, 20, -1);
+      rtc::NetworkRoute new_route(true, 10, 20, -1, 28);
       call_->OnNetworkRouteChanged("transport", new_route);
       Call::Config::BitrateConfig bitrate_config;
       bitrate_config.start_bitrate_bps = kStartBitrateBps;
       call_->SetBitrateConfig(bitrate_config);
       EXPECT_TRUE(Wait())
           << "Timed out while waiting for start bitrate to be exceeded.";
 
       bitrate_config.start_bitrate_bps = -1;
       bitrate_config.max_bitrate_bps = kNewMaxBitrateBps;
       call_->SetBitrateConfig(bitrate_config);
@@ skipping 1615 matching lines @@
         observation_complete_.Set();
       }
     }
   } test;
 
   RunBaseTest(&test);
 }
 #endif  // !defined(RTC_DISABLE_VP9)
 
 }  // namespace webrtc