Fixing scenario where track is rejected and later un-rejected.

talk/app/webrtc/mediastreamsignaling.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 309 matching lines @@
   // MediaStream and triggers MediaStreamSignaling::OnRemoveRemoteAudioTrack or
   // MediaStreamSignaling::OnRemoveRemoteVideoTrack.
   void OnRemoteTrackRemoved(const std::string& stream_label,
                             const std::string& track_id,
                             cricket::MediaType media_type);
 
   // Set the MediaStreamTrackInterface::TrackState to |kEnded| on all remote
   // tracks of type |media_type|.
   void RejectRemoteTracks(cricket::MediaType media_type);
 
+  // Set the MediaStreamTrackInterface::TrackState to |kLive| on all remote
+  // tracks of type |media_type|.
+  // Needed in case the local description caused the track to be rejected,
+  // then later accepted, without being destroyed.
+  void AcceptRemoteTracks(cricket::MediaType media_type);
+
   // Finds remote MediaStreams without any tracks and removes them from
   // |remote_streams_| and notifies the observer that the MediaStream no longer
   // exist.
   void UpdateEndedRemoteMediaStreams();
   void MaybeCreateDefaultStream();
   TrackInfos* GetRemoteTracks(cricket::MediaType type);
 
   // Returns a map of currently negotiated LocalTrackInfo of type |type|.
   TrackInfos* GetLocalTracks(cricket::MediaType type);
   bool FindLocalTrack(const std::string& track_id, cricket::MediaType type);
@@ skipping 57 matching lines @@
   int last_allocated_sctp_even_sid_;
   int last_allocated_sctp_odd_sid_;
 
   RtpDataChannels rtp_data_channels_;
   SctpDataChannels sctp_data_channels_;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_MEDIASTREAMSIGNALING_H_