Add flakiness check if there is no received packets in a certain period.

talk/app/webrtc/peerconnection.h

 /*
  * libjingle
  * Copyright 2012 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 131 matching lines @@
                                uint32 ssrc) override;
   void OnRemoveLocalVideoTrack(MediaStreamInterface* stream,
                                VideoTrackInterface* video_track) override;
   virtual void OnRemoveLocalStream(MediaStreamInterface* stream);
 
   // Implements IceObserver
   virtual void OnIceConnectionChange(IceConnectionState new_state);
   virtual void OnIceGatheringChange(IceGatheringState new_state);
   virtual void OnIceCandidate(const IceCandidateInterface* candidate);
   virtual void OnIceComplete();
+  void OnIceFlakinessChange(bool flaky) override;

[pthatcher1, 2015/06/24 23:42:52]

Can you make them all "override", then, to be consistent?

[honghaiz3, 2015/08/05 23:56:56]

Done.

 
   // Signals from WebRtcSession.
   void OnSessionStateChange(cricket::BaseSession* session,
                             cricket::BaseSession::State state);
   void ChangeSignalingState(SignalingState signaling_state);
 
   bool DoInitialize(IceTransportsType type,
                     const StunConfigurations& stun_config,
                     const TurnConfigurations& turn_config,
                     const MediaConstraintsInterface* constraints,
@@ skipping 30 matching lines @@
   rtc::scoped_ptr<cricket::PortAllocator> port_allocator_;
   rtc::scoped_ptr<WebRtcSession> session_;
   rtc::scoped_ptr<MediaStreamSignaling> mediastream_signaling_;
   rtc::scoped_ptr<MediaStreamHandlerContainer> stream_handler_container_;
   rtc::scoped_ptr<StatsCollector> stats_;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_PEERCONNECTION_H_