Delete deprecated PeerConnection methods, and corresponding using declarations.

webrtc/api/peerconnectioninterface.h

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 420 matching lines @@
   // Add a new MediaStream to be sent on this PeerConnection.
   // Note that a SessionDescription negotiation is needed before the
   // remote peer can receive the stream.
   virtual bool AddStream(MediaStreamInterface* stream) = 0;
 
   // Remove a MediaStream from this PeerConnection.
   // Note that a SessionDescription negotiation is need before the
   // remote peer is notified.
   virtual void RemoveStream(MediaStreamInterface* stream) = 0;
 
-  // TODO(deadbeef): Make the following two methods pure virtual once
-  // implemented by all subclasses of PeerConnectionInterface.
   // Add a new MediaStreamTrack to be sent on this PeerConnection.
   // |streams| indicates which stream labels the track should be associated
   // with.
   virtual rtc::scoped_refptr<RtpSenderInterface> AddTrack(
       MediaStreamTrackInterface* track,
-      std::vector<MediaStreamInterface*> streams) {
-    return nullptr;
-  }
+      std::vector<MediaStreamInterface*> streams) = 0;
 
   // Remove an RtpSender from this PeerConnection.
   // Returns true on success.
-  virtual bool RemoveTrack(RtpSenderInterface* sender) {
-    return false;
-  }
+  virtual bool RemoveTrack(RtpSenderInterface* sender) = 0;
 
   // Returns pointer to the created DtmfSender on success.
   // Otherwise returns NULL.
   virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
       AudioTrackInterface* track) = 0;
 
   // TODO(deadbeef): Make these pure virtual once all subclasses implement them.
   // |kind| must be "audio" or "video".
   // |stream_id| is used to populate the msid attribute; if empty, one will
   // be generated automatically.
@@ skipping 182 matching lines @@
 
   // Triggered when the SignalingState changed.
   virtual void OnSignalingChange(
       PeerConnectionInterface::SignalingState new_state) = 0;
 
   // TODO(deadbeef): Once all subclasses override the scoped_refptr versions
   // of the below three methods, make them pure virtual and remove the raw
   // pointer version.
 
   // Triggered when media is received on a new stream from remote peer.
-  virtual void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) {}
-  // Deprecated; please use the version that uses a scoped_refptr.
-  virtual void OnAddStream(MediaStreamInterface* stream) {}

[nisse-webrtc, 2017/01/30 15:54:47]

This change breaks tests. Unclear why, I'll try adding proper override
declarations in subclasses, to see if that triggers any errors at compile time
rather than run time.

+  virtual void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) = 0;
 
   // Triggered when a remote peer close a stream.
-  virtual void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream) {
-  }
-  // Deprecated; please use the version that uses a scoped_refptr.
-  virtual void OnRemoveStream(MediaStreamInterface* stream) {}
+  virtual void OnRemoveStream(
+      rtc::scoped_refptr<MediaStreamInterface> stream) = 0;
 
   // Triggered when a remote peer opens a data channel.
   virtual void OnDataChannel(
-      rtc::scoped_refptr<DataChannelInterface> data_channel){};
-  // Deprecated; please use the version that uses a scoped_refptr.
-  virtual void OnDataChannel(DataChannelInterface* data_channel) {}
+      rtc::scoped_refptr<DataChannelInterface> data_channel) = 0;
 
   // Triggered when renegotiation is needed. For example, an ICE restart
   // has begun.
   virtual void OnRenegotiationNeeded() = 0;
 
   // Called any time the IceConnectionState changes.
   virtual void OnIceConnectionChange(
       PeerConnectionInterface::IceConnectionState new_state) = 0;
 
   // Called any time the IceGatheringState changes.
@@ skipping 198 matching lines @@
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory) {
   return CreatePeerConnectionFactory(
       worker_and_network_thread, worker_and_network_thread, signaling_thread,
       default_adm, encoder_factory, decoder_factory);
 }
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_PEERCONNECTIONINTERFACE_H_