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

Issue 2224563004: Add signaling to support ICE renomination. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Merge Created 4 years, 3 months ago
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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
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310 bool enable_rtp_data_channel = false; 310 bool enable_rtp_data_channel = false;
311 bool enable_quic = false; 311 bool enable_quic = false;
312 rtc::Optional<int> screencast_min_bitrate; 312 rtc::Optional<int> screencast_min_bitrate;
313 rtc::Optional<bool> combined_audio_video_bwe; 313 rtc::Optional<bool> combined_audio_video_bwe;
314 rtc::Optional<bool> enable_dtls_srtp; 314 rtc::Optional<bool> enable_dtls_srtp;
315 int ice_candidate_pool_size = 0; 315 int ice_candidate_pool_size = 0;
316 bool prune_turn_ports = false; 316 bool prune_turn_ports = false;
317 // If set to true, this means the ICE transport should presume TURN-to-TURN 317 // If set to true, this means the ICE transport should presume TURN-to-TURN
318 // candidate pairs will succeed, even before a binding response is received. 318 // candidate pairs will succeed, even before a binding response is received.
319 bool presume_writable_when_fully_relayed = false; 319 bool presume_writable_when_fully_relayed = false;
320 // If true, "renomination" will be added to the ice options in the transport
321 // description.
322 bool enable_ice_renomination = false;
320 // If true, ICE role is redetermined when peerconnection sets a local 323 // If true, ICE role is redetermined when peerconnection sets a local
321 // transport description that indicates an ICE restart. 324 // transport description that indicates an ICE restart.
322 bool redetermine_role_on_ice_restart = true; 325 bool redetermine_role_on_ice_restart = true;
323 }; 326 };
324 327
325 struct RTCOfferAnswerOptions { 328 struct RTCOfferAnswerOptions {
326 static const int kUndefined = -1; 329 static const int kUndefined = -1;
327 static const int kMaxOfferToReceiveMedia = 1; 330 static const int kMaxOfferToReceiveMedia = 1;
328 331
329 // The default value for constraint offerToReceiveX:true. 332 // The default value for constraint offerToReceiveX:true.
330 static const int kOfferToReceiveMediaTrue = 1; 333 static const int kOfferToReceiveMediaTrue = 1;
331 334
332 int offer_to_receive_video; 335 int offer_to_receive_video = kUndefined;
333 int offer_to_receive_audio; 336 int offer_to_receive_audio = kUndefined;
334 bool voice_activity_detection; 337 bool voice_activity_detection = true;
335 bool ice_restart; 338 bool ice_restart = false;
336 bool use_rtp_mux; 339 bool use_rtp_mux = true;
337 340
338 RTCOfferAnswerOptions() 341 RTCOfferAnswerOptions() = default;
339 : offer_to_receive_video(kUndefined),
340 offer_to_receive_audio(kUndefined),
341 voice_activity_detection(true),
342 ice_restart(false),
343 use_rtp_mux(true) {}
344 342
345 RTCOfferAnswerOptions(int offer_to_receive_video, 343 RTCOfferAnswerOptions(int offer_to_receive_video,
346 int offer_to_receive_audio, 344 int offer_to_receive_audio,
347 bool voice_activity_detection, 345 bool voice_activity_detection,
348 bool ice_restart, 346 bool ice_restart,
349 bool use_rtp_mux) 347 bool use_rtp_mux)
350 : offer_to_receive_video(offer_to_receive_video), 348 : offer_to_receive_video(offer_to_receive_video),
351 offer_to_receive_audio(offer_to_receive_audio), 349 offer_to_receive_audio(offer_to_receive_audio),
352 voice_activity_detection(voice_activity_detection), 350 voice_activity_detection(voice_activity_detection),
353 ice_restart(ice_restart), 351 ice_restart(ice_restart),
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754 cricket::WebRtcVideoEncoderFactory* encoder_factory, 752 cricket::WebRtcVideoEncoderFactory* encoder_factory,
755 cricket::WebRtcVideoDecoderFactory* decoder_factory) { 753 cricket::WebRtcVideoDecoderFactory* decoder_factory) {
756 return CreatePeerConnectionFactory( 754 return CreatePeerConnectionFactory(
757 worker_and_network_thread, worker_and_network_thread, signaling_thread, 755 worker_and_network_thread, worker_and_network_thread, signaling_thread,
758 default_adm, encoder_factory, decoder_factory); 756 default_adm, encoder_factory, decoder_factory);
759 } 757 }
760 758
761 } // namespace webrtc 759 } // namespace webrtc
762 760
763 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ 761 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_
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