Restoring behavior where PeerConnection tracks changes to MediaStreams.

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 24 matching lines @@
 #include "talk/app/webrtc/peerconnectioninterface.h"
 #include "talk/app/webrtc/rtpreceiverinterface.h"
 #include "talk/app/webrtc/rtpsenderinterface.h"
 #include "talk/app/webrtc/statscollector.h"
 #include "talk/app/webrtc/streamcollection.h"
 #include "talk/app/webrtc/webrtcsession.h"
 #include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 
+class MediaStreamObserver;
 class RemoteMediaStreamFactory;
 
 typedef std::vector<PortAllocatorFactoryInterface::StunConfiguration>
     StunConfigurations;
 typedef std::vector<PortAllocatorFactoryInterface::TurnConfiguration>
     TurnConfigurations;
 
 // Populates |session_options| from |rtc_options|, and returns true if options
 // are valid.
 bool ConvertRtcOptionsForOffer(
@@ skipping 155 matching lines @@
   void OnIceConnectionChange(IceConnectionState new_state) override;
   void OnIceGatheringChange(IceGatheringState new_state) override;
   void OnIceCandidate(const IceCandidateInterface* candidate) override;
   void OnIceComplete() override;
   void OnIceConnectionReceivingChange(bool receiving) override;
 
   // Signals from WebRtcSession.
   void OnSessionStateChange(WebRtcSession* session, WebRtcSession::State state);
   void ChangeSignalingState(SignalingState signaling_state);
 
+  // Signals from MediaStreamObserver.
+  void OnAudioTrackAdded(AudioTrackInterface* track,
+                         MediaStreamInterface* stream);
+  void OnAudioTrackRemoved(AudioTrackInterface* track,
+                           MediaStreamInterface* stream);
+  void OnVideoTrackAdded(VideoTrackInterface* track,
+                         MediaStreamInterface* stream);
+  void OnVideoTrackRemoved(VideoTrackInterface* track,
+                           MediaStreamInterface* stream);
+
   rtc::Thread* signaling_thread() const {
     return factory_->signaling_thread();
   }
 
   void PostSetSessionDescriptionFailure(SetSessionDescriptionObserver* observer,
                                         const std::string& error);
   void PostCreateSessionDescriptionFailure(
       CreateSessionDescriptionObserver* observer,
       const std::string& error);
 
@@ skipping 142 matching lines @@
   IceGatheringState ice_gathering_state_;
 
   rtc::scoped_ptr<cricket::PortAllocator> port_allocator_;
   rtc::scoped_ptr<MediaControllerInterface> media_controller_;
 
   // Streams added via AddStream.
   rtc::scoped_refptr<StreamCollection> local_streams_;
   // Streams created as a result of SetRemoteDescription.
   rtc::scoped_refptr<StreamCollection> remote_streams_;
 
+  std::vector<rtc::scoped_ptr<MediaStreamObserver>> stream_observers_;
+
   // These lists store track info seen in local/remote descriptions.
   TrackInfos remote_audio_tracks_;
   TrackInfos remote_video_tracks_;
   TrackInfos local_audio_tracks_;
   TrackInfos local_video_tracks_;
 
   SctpSidAllocator sid_allocator_;
   // label -> DataChannel
   std::map<std::string, rtc::scoped_refptr<DataChannel>> rtp_data_channels_;
   std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_;
 
   RemotePeerInfo remote_info_;
   rtc::scoped_ptr<RemoteMediaStreamFactory> remote_stream_factory_;
 
   std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders_;
   std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers_;
 
   // The session_ scoped_ptr is declared at the bottom of PeerConnection
   // 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_