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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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294 // Flags corresponding to values set by constraint flags. | 294 // Flags corresponding to values set by constraint flags. |
295 // rtc::Optional flags can be "missing", in which case the webrtc | 295 // rtc::Optional flags can be "missing", in which case the webrtc |
296 // default applies. | 296 // default applies. |
297 bool disable_ipv6 = false; | 297 bool disable_ipv6 = false; |
298 bool enable_rtp_data_channel = false; | 298 bool enable_rtp_data_channel = false; |
299 rtc::Optional<int> screencast_min_bitrate; | 299 rtc::Optional<int> screencast_min_bitrate; |
300 rtc::Optional<bool> combined_audio_video_bwe; | 300 rtc::Optional<bool> combined_audio_video_bwe; |
301 rtc::Optional<bool> enable_dtls_srtp; | 301 rtc::Optional<bool> enable_dtls_srtp; |
302 int ice_candidate_pool_size = 0; | 302 int ice_candidate_pool_size = 0; |
303 bool prune_turn_ports = false; | 303 bool prune_turn_ports = false; |
| 304 // If set to true, this means the ICE transport should presume TURN-to-TURN |
| 305 // candidate pairs will succeed, even before a binding response is received. |
| 306 bool presume_writable_when_fully_relayed = false; |
304 }; | 307 }; |
305 | 308 |
306 struct RTCOfferAnswerOptions { | 309 struct RTCOfferAnswerOptions { |
307 static const int kUndefined = -1; | 310 static const int kUndefined = -1; |
308 static const int kMaxOfferToReceiveMedia = 1; | 311 static const int kMaxOfferToReceiveMedia = 1; |
309 | 312 |
310 // The default value for constraint offerToReceiveX:true. | 313 // The default value for constraint offerToReceiveX:true. |
311 static const int kOfferToReceiveMediaTrue = 1; | 314 static const int kOfferToReceiveMediaTrue = 1; |
312 | 315 |
313 int offer_to_receive_video; | 316 int offer_to_receive_video; |
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722 cricket::WebRtcVideoEncoderFactory* encoder_factory, | 725 cricket::WebRtcVideoEncoderFactory* encoder_factory, |
723 cricket::WebRtcVideoDecoderFactory* decoder_factory) { | 726 cricket::WebRtcVideoDecoderFactory* decoder_factory) { |
724 return CreatePeerConnectionFactory( | 727 return CreatePeerConnectionFactory( |
725 worker_and_network_thread, worker_and_network_thread, signaling_thread, | 728 worker_and_network_thread, worker_and_network_thread, signaling_thread, |
726 default_adm, encoder_factory, decoder_factory); | 729 default_adm, encoder_factory, decoder_factory); |
727 } | 730 } |
728 | 731 |
729 } // namespace webrtc | 732 } // namespace webrtc |
730 | 733 |
731 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ | 734 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ |
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