Adding the ability to create an RtpSender without a track.

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 75 matching lines @@
   rtc::scoped_refptr<StreamCollectionInterface> local_streams() override;
   rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override;
   bool AddStream(MediaStreamInterface* local_stream) override;
   void RemoveStream(MediaStreamInterface* local_stream) override;
 
   virtual WebRtcSession* session() { return session_.get(); }
 
   rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
       AudioTrackInterface* track) override;
 
+  rtc::scoped_refptr<RtpSenderInterface> CreateSender(
+      const std::string& kind) override;
+
   std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
       const override;
   std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
       const override;
 
   rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
       const std::string& label,
       const DataChannelInit* config) override;
   bool GetStats(StatsObserver* observer,
                 webrtc::MediaStreamTrackInterface* track,
@@ skipping 76 matching lines @@
   void CreateAudioReceiver(MediaStreamInterface* stream,
                            AudioTrackInterface* audio_track,
                            uint32_t ssrc);
   void CreateVideoReceiver(MediaStreamInterface* stream,
                            VideoTrackInterface* video_track,
                            uint32_t ssrc);
   void DestroyAudioReceiver(MediaStreamInterface* stream,
                             AudioTrackInterface* audio_track);
   void DestroyVideoReceiver(MediaStreamInterface* stream,
                             VideoTrackInterface* video_track);
-  void CreateAudioSender(MediaStreamInterface* stream,
-                         AudioTrackInterface* audio_track,
-                         uint32_t ssrc);
-  void CreateVideoSender(MediaStreamInterface* stream,
-                         VideoTrackInterface* video_track,
-                         uint32_t ssrc);
   void DestroyAudioSender(MediaStreamInterface* stream,
                           AudioTrackInterface* audio_track,
                           uint32_t ssrc);
   void DestroyVideoSender(MediaStreamInterface* stream,
                           VideoTrackInterface* video_track);
 
   // Implements IceObserver
   void OnIceConnectionChange(IceConnectionState new_state) override;
   void OnIceGatheringChange(IceGatheringState new_state) override;
   void OnIceCandidate(const IceCandidateInterface* candidate) override;
@@ skipping 115 matching lines @@
   // Notifications from WebRtcSession relating to BaseChannels.
   void OnVoiceChannelDestroyed();
   void OnVideoChannelDestroyed();
   void OnDataChannelCreated();
   void OnDataChannelDestroyed();
   // Called when the cricket::DataChannel receives a message indicating that a
   // webrtc::DataChannel should be opened.
   void OnDataChannelOpenMessage(const std::string& label,
                                 const InternalDataChannelInit& config);
 
+  RtpSenderInterface* FindSenderById(const std::string& id);
+
   std::vector<rtc::scoped_refptr<RtpSenderInterface>>::iterator
   FindSenderForTrack(MediaStreamTrackInterface* track);
   std::vector<rtc::scoped_refptr<RtpReceiverInterface>>::iterator
   FindReceiverForTrack(MediaStreamTrackInterface* track);
 
   TrackInfos* GetRemoteTracks(cricket::MediaType media_type);
   TrackInfos* GetLocalTracks(cricket::MediaType media_type);
   const TrackInfo* FindTrackInfo(const TrackInfos& infos,
                                  const std::string& stream_label,
                                  const std::string track_id) const;
@@ skipping 46 matching lines @@
   // because its destruction fires signals (such as VoiceChannelDestroyed)
   // which will trigger some final actions in PeerConnection...
   rtc::scoped_ptr<WebRtcSession> session_;
   // ... But stats_ depends on session_ so it should be destroyed even earlier.
   rtc::scoped_ptr<StatsCollector> stats_;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_PEERCONNECTION_H_