Add signaling to support ICE renomination.

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 287 matching lines @@
     bool enable_rtp_data_channel = false;
     bool enable_quic = false;
     rtc::Optional<int> screencast_min_bitrate;
     rtc::Optional<bool> combined_audio_video_bwe;
     rtc::Optional<bool> enable_dtls_srtp;
     int ice_candidate_pool_size = 0;
     bool prune_turn_ports = false;
     // If set to true, this means the ICE transport should presume TURN-to-TURN
     // candidate pairs will succeed, even before a binding response is received.
     bool presume_writable_when_fully_relayed = false;
+
+    bool ice_renomination = false;

[Taylor Brandstetter, 2016/08/17 22:02:00]

nit: Could add comment explaining that this controls whether "renomination" is
added to the ICE options.

[honghaiz3, 2016/08/19 18:42:01]

Done.

   };
 
   struct RTCOfferAnswerOptions {
     static const int kUndefined = -1;
     static const int kMaxOfferToReceiveMedia = 1;
 
     // The default value for constraint offerToReceiveX:true.
     static const int kOfferToReceiveMediaTrue = 1;
 
-    int offer_to_receive_video;
-    int offer_to_receive_audio;
-    bool voice_activity_detection;
-    bool ice_restart;
-    bool use_rtp_mux;
+    int offer_to_receive_video = kUndefined;
+    int offer_to_receive_audio = kUndefined;
+    bool voice_activity_detection = true;
+    bool ice_restart = false;
+    bool use_rtp_mux = true;
 
-    RTCOfferAnswerOptions()
-        : offer_to_receive_video(kUndefined),
-          offer_to_receive_audio(kUndefined),
-          voice_activity_detection(true),
-          ice_restart(false),
-          use_rtp_mux(true) {}
+    RTCOfferAnswerOptions() = default;
 
     RTCOfferAnswerOptions(int offer_to_receive_video,
                           int offer_to_receive_audio,
                           bool voice_activity_detection,
                           bool ice_restart,
                           bool use_rtp_mux)
         : offer_to_receive_video(offer_to_receive_video),
           offer_to_receive_audio(offer_to_receive_audio),
           voice_activity_detection(voice_activity_detection),
           ice_restart(ice_restart),
@@ skipping 400 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_