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

talk/media/webrtc/fakewebrtccall.h

 /*
  * 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 146 matching lines @@
   explicit FakeCall(const webrtc::Call::Config& config);
   ~FakeCall() override;
 
   webrtc::Call::Config GetConfig() const;
   const std::vector<FakeVideoSendStream*>& GetVideoSendStreams();
   const std::vector<FakeVideoReceiveStream*>& GetVideoReceiveStreams();
 
   const std::vector<FakeAudioReceiveStream*>& GetAudioReceiveStreams();
   const FakeAudioReceiveStream* GetAudioReceiveStream(uint32_t ssrc);
 
+  rtc::SentPacket last_sent_packet() const { return last_sent_packet_; }
   webrtc::NetworkState GetNetworkState() const;
   int GetNumCreatedSendStreams() const;
   int GetNumCreatedReceiveStreams() const;
   void SetStats(const webrtc::Call::Stats& stats);
 
  private:
   webrtc::AudioSendStream* CreateAudioSendStream(
       const webrtc::AudioSendStream::Config& config) override;
   void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
 
@@ skipping 16 matching lines @@
   DeliveryStatus DeliverPacket(webrtc::MediaType media_type,
                                const uint8_t* packet,
                                size_t length,
                                const webrtc::PacketTime& packet_time) override;
 
   webrtc::Call::Stats GetStats() const override;
 
   void SetBitrateConfig(
       const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
   void SignalNetworkState(webrtc::NetworkState state) override;
+  void OnSentPacket(const rtc::SentPacket& sent_packet) override;
 
   webrtc::Call::Config config_;
   webrtc::NetworkState network_state_;
+  rtc::SentPacket last_sent_packet_;
   webrtc::Call::Stats stats_;
   std::vector<FakeVideoSendStream*> video_send_streams_;
   std::vector<FakeVideoReceiveStream*> video_receive_streams_;
   std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
 
   int num_created_send_streams_;
   int num_created_receive_streams_;
 };
 
 }  // namespace cricket
 #endif  // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_