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

talk/app/webrtc/rtpsenderinterface.h

 /*
  * libjingle
  * Copyright 2015 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 17 matching lines @@
 // This file contains interfaces for RtpSenders
 // http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
 
 #ifndef TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_
 #define TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_
 
 #include <string>
 
 #include "talk/app/webrtc/proxy.h"
 #include "talk/app/webrtc/mediastreaminterface.h"
+#include "talk/session/media/mediasession.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 
 namespace webrtc {
 
 class RtpSenderInterface : public rtc::RefCountInterface {
  public:
   // Returns true if successful in setting the track.
   // Fails if an audio track is set on a video RtpSender, or vice-versa.
   virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;
   virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
 
+  // Audio or video?
+  virtual cricket::MediaType media_type() const = 0;
+
   // Not to be confused with "mid", this is a field we can temporarily use
   // to uniquely identify a receiver until we implement Unified Plan SDP.
   virtual std::string id() const = 0;
 
   virtual void Stop() = 0;
 
+  virtual void Reconfigure() = 0;
+
  protected:
   virtual ~RtpSenderInterface() {}
 };
 
 // Define proxy for RtpSenderInterface.
 BEGIN_PROXY_MAP(RtpSender)
 PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
 PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
+PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
 PROXY_CONSTMETHOD0(std::string, id)
 PROXY_METHOD0(void, Stop)
+PROXY_METHOD0(void, Reconfigure)
 END_PROXY()
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_