Fixing flaky test: WebRtcSessionTest.TestPacketOptionsAndOnPacketSent

webrtc/media/engine/fakewebrtccall.h

 /*
  *  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 159 matching lines @@
   webrtc::Call::Config GetConfig() const;
   const std::vector<FakeVideoSendStream*>& GetVideoSendStreams();
   const std::vector<FakeVideoReceiveStream*>& GetVideoReceiveStreams();
 
   const std::vector<FakeAudioSendStream*>& GetAudioSendStreams();
   const FakeAudioSendStream* GetAudioSendStream(uint32_t ssrc);
   const std::vector<FakeAudioReceiveStream*>& GetAudioReceiveStreams();
   const FakeAudioReceiveStream* GetAudioReceiveStream(uint32_t ssrc);
 
   rtc::SentPacket last_sent_packet() const { return last_sent_packet_; }
+
+  // This is useful if we care about the last media packet (with id populated)
+  // but not the last ICE packet (with -1 ID).
+  int last_sent_nonnegative_packet_id() const {
+    return last_sent_nonnegative_packet_id_;
+  }
+
   webrtc::NetworkState GetNetworkState(webrtc::MediaType media) 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 25 matching lines @@
   void OnNetworkRouteChanged(const std::string& transport_name,
                              const rtc::NetworkRoute& network_route) override {}
   void SignalChannelNetworkState(webrtc::MediaType media,
                                  webrtc::NetworkState state) override;
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
 
   webrtc::Call::Config config_;
   webrtc::NetworkState audio_network_state_;
   webrtc::NetworkState video_network_state_;
   rtc::SentPacket last_sent_packet_;
+  int last_sent_nonnegative_packet_id_ = -1;
   webrtc::Call::Stats stats_;
   std::vector<FakeVideoSendStream*> video_send_streams_;
   std::vector<FakeAudioSendStream*> audio_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_