Remove BaseSession::SignalNewDescription. It was only used by GTP and now just clutters the code.

talk/app/webrtc/webrtcsession.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 240 matching lines @@
   void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
 
   // For unit test.
   bool waiting_for_identity() const;
 
   void set_metrics_observer(
       webrtc::MetricsObserverInterface* metrics_observer) {
     metrics_observer_ = metrics_observer;
   }
 
- protected:
-  // Don't fire a new description.  The only thing it's used for is to
-  // push new media descriptions to the BaseChannels.  But in
-  // WebRtcSession, we just push to the BaseChannels directly, so we
-  // don't need this (and it would cause the descriptions to be pushed
-  // down twice).
-  // TODO(pthatcher): Remove this method and signal completely from
-  // BaseSession once all the subclasses of BaseSession push to
-  // BaseChannels directly rather than relying on the signal, or once
-  // BaseChannel no longer listens to the event and requires
-  // descriptions to be pushed down.
-  virtual void SignalNewDescription() override {}
-
  private:
   // Indicates the type of SessionDescription in a call to SetLocalDescription
   // and SetRemoteDescription.
   enum Action {
     kOffer,
     kPrAnswer,
     kAnswer,
   };
 
   // Invokes ConnectChannels() on transport proxies, which initiates ice
@@ skipping 138 matching lines @@
   PeerConnectionInterface::BundlePolicy bundle_policy_;
 
   // Declares the RTCP mux policy for the WebRTCSession.
   PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
 };
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_WEBRTCSESSION_H_