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 d021ecab3ddc2c4cdc0ea72679e266bc5ad7968b..c1a90b02ffbaa06f1110199607b2c9b67a10b9f2 100644
--- a/webrtc/media/engine/fakewebrtccall.cc
+++ b/webrtc/media/engine/fakewebrtccall.cc
@@ -503,6 +503,23 @@ void FakeCall::SignalChannelNetworkState(webrtc::MediaType media,
   }
 }
 
+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) {