Set actual transport overhead in rtp_rtcp

webrtc/media/engine/fakewebrtccall.cc

 /*
  *  Copyright (c) 2015 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 485 matching lines @@
     case webrtc::MediaType::VIDEO:
       video_network_state_ = state;
       break;
     case webrtc::MediaType::DATA:
     case webrtc::MediaType::ANY:
       ADD_FAILURE()
           << "SignalChannelNetworkState called with unknown parameter.";
   }
 }
 
+void FakeCall::SignalTransportOverheadChange(
+    webrtc::MediaType media,
+    int transport_overhead_per_packet) {
+  switch (media) {
+    case webrtc::MediaType::AUDIO:
+      audio_transport_overhead_ = transport_overhead_per_packet;
+      break;
+    case webrtc::MediaType::VIDEO:
+      video_transport_overhead_ = transport_overhead_per_packet;
+      break;
+    case webrtc::MediaType::DATA:
+    case webrtc::MediaType::ANY:
+      ADD_FAILURE()
+          << "SignalChannelNetworkState called with unknown parameter.";
+  }
+}
+
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
   last_sent_packet_ = sent_packet;
   if (sent_packet.packet_id >= 0) {
     last_sent_nonnegative_packet_id_ = sent_packet.packet_id;
   }
 }
 
 }  // namespace cricket