Fix for scenario where m-line is revived after being set to port 0.

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 218 matching lines @@
   virtual bool GetOptionsForOffer(
       const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
       cricket::MediaSessionOptions* session_options);
 
   // Returns a MediaSessionOptions struct with options decided by
   // |constraints|, the local MediaStreams and DataChannels.
   virtual bool GetOptionsForAnswer(
       const MediaConstraintsInterface* constraints,
       cricket::MediaSessionOptions* session_options);
 
+  // Remove all local and remote tracks of type |media_type|.
+  // Called when a media type is rejected (m-line set to port 0).
+  void RemoveTracks(cricket::MediaType media_type);
+
   // Makes sure a MediaStream Track is created for each StreamParam in
   // |streams|. |media_type| is the type of the |streams| and can be either
   // audio or video.
   // If a new MediaStream is created it is added to |new_streams|.
   void UpdateRemoteStreamsList(
       const std::vector<cricket::StreamParams>& streams,
       cricket::MediaType media_type,
       StreamCollection* new_streams);
 
   // Triggered when a remote track has been seen for the first time in a remote
@@ skipping 138 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_