Set actual transport overhead in rtp_rtcp

webrtc/media/engine/fakewebrtccall.cc

Index: webrtc/media/engine/fakewebrtccall.cc
diff --git a/webrtc/media/engine/fakewebrtccall.cc b/webrtc/media/engine/fakewebrtccall.cc
index 43e083fefb469a707190a9267d279814e108e94e..d2aa7bc1b4e2d3d994627a073d5acb26c115d15c 100644
--- a/webrtc/media/engine/fakewebrtccall.cc
+++ b/webrtc/media/engine/fakewebrtccall.cc
@@ -528,6 +528,22 @@ void FakeCall::SignalChannelNetworkState(webrtc::MediaType media,
   }
 }
 
+void FakeCall::OnTransportOverheadChanged(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) {