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

Issue 2111663003: Revert of Add config to prune TURN ports (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 5 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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293 struct cricket::MediaConfig media_config; 293 struct cricket::MediaConfig media_config;
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;
304 }; 303 };
305 304
306 struct RTCOfferAnswerOptions { 305 struct RTCOfferAnswerOptions {
307 static const int kUndefined = -1; 306 static const int kUndefined = -1;
308 static const int kMaxOfferToReceiveMedia = 1; 307 static const int kMaxOfferToReceiveMedia = 1;
309 308
310 // The default value for constraint offerToReceiveX:true. 309 // The default value for constraint offerToReceiveX:true.
311 static const int kOfferToReceiveMediaTrue = 1; 310 static const int kOfferToReceiveMediaTrue = 1;
312 311
313 int offer_to_receive_video; 312 int offer_to_receive_video;
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722 cricket::WebRtcVideoEncoderFactory* encoder_factory, 721 cricket::WebRtcVideoEncoderFactory* encoder_factory,
723 cricket::WebRtcVideoDecoderFactory* decoder_factory) { 722 cricket::WebRtcVideoDecoderFactory* decoder_factory) {
724 return CreatePeerConnectionFactory( 723 return CreatePeerConnectionFactory(
725 worker_and_network_thread, worker_and_network_thread, signaling_thread, 724 worker_and_network_thread, worker_and_network_thread, signaling_thread,
726 default_adm, encoder_factory, decoder_factory); 725 default_adm, encoder_factory, decoder_factory);
727 } 726 }
728 727
729 } // namespace webrtc 728 } // namespace webrtc
730 729
731 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ 730 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_
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