Wire up transport sequence number / send time callbacks to webrtc via libjingle.

talk/media/webrtc/fakewebrtccall.cc

 /*
  * libjingle
  * Copyright 2015 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 218 matching lines @@
 
 const FakeAudioReceiveStream* FakeCall::GetAudioReceiveStream(uint32_t ssrc) {
   for (const auto p : GetAudioReceiveStreams()) {
     if (p->GetConfig().rtp.remote_ssrc == ssrc) {
       return p;
     }
   }
   return nullptr;
 }
 
+webrtc::SentPacket FakeCall::last_packet_sent() const {
+  return last_packet_sent_;
+}
+
 webrtc::NetworkState FakeCall::GetNetworkState() const {
   return network_state_;
 }
 
 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream(
     const webrtc::AudioSendStream::Config& config) {
   return nullptr;
 }
 
 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
@@ skipping 111 matching lines @@
 }
 
 void FakeCall::SetBitrateConfig(
     const webrtc::Call::Config::BitrateConfig& bitrate_config) {
   config_.bitrate_config = bitrate_config;
 }
 
 void FakeCall::SignalNetworkState(webrtc::NetworkState state) {
   network_state_ = state;
 }
+
+void FakeCall::OnSentPacket(const webrtc::SentPacket& packet_sent) {
+  last_packet_sent_ = packet_sent;
+}
 }  // namespace cricket