OLD | NEW |
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 299 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
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), |
(...skipping 400 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
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_ |
OLD | NEW |