Adding "first packet received" notification to PeerConnectionObserver.

talk/app/webrtc/webrtcsession.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 291 matching lines @@
 
   bool IceRestartPending() const;
 
   void ResetIceRestartLatch();
 
   // Called when an RTCCertificate is generated or retrieved by
   // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
   void OnCertificateReady(
       const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
   void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
+  void OnChannelFirstPacketReceived(cricket::BaseChannel*);
 
   // For unit test.
   bool waiting_for_certificate_for_testing() const;
   const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
 
   void set_metrics_observer(
       webrtc::MetricsObserverInterface* metrics_observer) {
     metrics_observer_ = metrics_observer;
   }
 
   // Called when voice_channel_, video_channel_ and data_channel_ are created
   // and destroyed. As a result of, for example, setting a new description.
   sigslot::signal0<> SignalVoiceChannelCreated;
   sigslot::signal0<> SignalVoiceChannelDestroyed;
   sigslot::signal0<> SignalVideoChannelCreated;
   sigslot::signal0<> SignalVideoChannelDestroyed;
   sigslot::signal0<> SignalDataChannelCreated;
   sigslot::signal0<> SignalDataChannelDestroyed;
   // Called when the whole session is destroyed.
   sigslot::signal0<> SignalDestroyed;
 
   // Called when a valid data channel OPEN message is received.
   // std::string represents the data channel label.
   sigslot::signal2<const std::string&, const InternalDataChannelInit&>
       SignalDataChannelOpenMessage;
 
+  // Called when the first RTP packet is received.
+  sigslot::signal0<> SignalFirstMediaPacketReceived;
+
  private:
   // Indicates the type of SessionDescription in a call to SetLocalDescription
   // and SetRemoteDescription.
   enum Action {
     kOffer,
     kPrAnswer,
     kAnswer,
   };
 
   // Log session state.
@@ skipping 163 matching lines @@
   cricket::AudioOptions audio_options_;
   cricket::VideoOptions video_options_;
   MetricsObserverInterface* metrics_observer_;
 
   // Declares the bundle policy for the WebRTCSession.
   PeerConnectionInterface::BundlePolicy bundle_policy_;
 
   // Declares the RTCP mux policy for the WebRTCSession.
   PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
 
+  bool has_received_media_packet_ = false;
+
   RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
 };
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_WEBRTCSESSION_H_