Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(21)

Side by Side Diff: webrtc/api/peerconnectioninterface.h

Issue 2295493002: Add a switch to redetermine role when ICE restarts. (Closed)
Patch Set: Address comments Created 4 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 /* 1 /*
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
(...skipping 287 matching lines...) Expand 10 before | Expand all | Expand 10 after
298 bool enable_rtp_data_channel = false; 298 bool enable_rtp_data_channel = false;
299 bool enable_quic = false; 299 bool enable_quic = false;
300 rtc::Optional<int> screencast_min_bitrate; 300 rtc::Optional<int> screencast_min_bitrate;
301 rtc::Optional<bool> combined_audio_video_bwe; 301 rtc::Optional<bool> combined_audio_video_bwe;
302 rtc::Optional<bool> enable_dtls_srtp; 302 rtc::Optional<bool> enable_dtls_srtp;
303 int ice_candidate_pool_size = 0; 303 int ice_candidate_pool_size = 0;
304 bool prune_turn_ports = false; 304 bool prune_turn_ports = false;
305 // If set to true, this means the ICE transport should presume TURN-to-TURN 305 // If set to true, this means the ICE transport should presume TURN-to-TURN
306 // candidate pairs will succeed, even before a binding response is received. 306 // candidate pairs will succeed, even before a binding response is received.
307 bool presume_writable_when_fully_relayed = false; 307 bool presume_writable_when_fully_relayed = false;
308 // If true, ICE role is redetermined when peerconnection sets a local
309 // transport description that indicates an ICE restart.
310 // Default to true because that is the current behavior.
311 bool redetermine_role_on_ice_restart = true;
pthatcher1 2016/08/29 23:27:42 Can we make the default = false?
honghaiz3 2016/08/29 23:32:27 Done.
308 }; 312 };
309 313
310 struct RTCOfferAnswerOptions { 314 struct RTCOfferAnswerOptions {
311 static const int kUndefined = -1; 315 static const int kUndefined = -1;
312 static const int kMaxOfferToReceiveMedia = 1; 316 static const int kMaxOfferToReceiveMedia = 1;
313 317
314 // The default value for constraint offerToReceiveX:true. 318 // The default value for constraint offerToReceiveX:true.
315 static const int kOfferToReceiveMediaTrue = 1; 319 static const int kOfferToReceiveMediaTrue = 1;
316 320
317 int offer_to_receive_video; 321 int offer_to_receive_video;
(...skipping 421 matching lines...) Expand 10 before | Expand all | Expand 10 after
739 cricket::WebRtcVideoEncoderFactory* encoder_factory, 743 cricket::WebRtcVideoEncoderFactory* encoder_factory,
740 cricket::WebRtcVideoDecoderFactory* decoder_factory) { 744 cricket::WebRtcVideoDecoderFactory* decoder_factory) {
741 return CreatePeerConnectionFactory( 745 return CreatePeerConnectionFactory(
742 worker_and_network_thread, worker_and_network_thread, signaling_thread, 746 worker_and_network_thread, worker_and_network_thread, signaling_thread,
743 default_adm, encoder_factory, decoder_factory); 747 default_adm, encoder_factory, decoder_factory);
744 } 748 }
745 749
746 } // namespace webrtc 750 } // namespace webrtc
747 751
748 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ 752 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698