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

Issue 1956453003: Relanding: Implement RTCConfiguration.iceCandidatePoolSize. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 7 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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286 bool prioritize_most_likely_ice_candidate_pairs; 286 bool prioritize_most_likely_ice_candidate_pairs;
287 struct cricket::MediaConfig media_config; 287 struct cricket::MediaConfig media_config;
288 // Flags corresponding to values set by constraint flags. 288 // Flags corresponding to values set by constraint flags.
289 // rtc::Optional flags can be "missing", in which case the webrtc 289 // rtc::Optional flags can be "missing", in which case the webrtc
290 // default applies. 290 // default applies.
291 bool disable_ipv6; 291 bool disable_ipv6;
292 bool enable_rtp_data_channel; 292 bool enable_rtp_data_channel;
293 rtc::Optional<int> screencast_min_bitrate; 293 rtc::Optional<int> screencast_min_bitrate;
294 rtc::Optional<bool> combined_audio_video_bwe; 294 rtc::Optional<bool> combined_audio_video_bwe;
295 rtc::Optional<bool> enable_dtls_srtp; 295 rtc::Optional<bool> enable_dtls_srtp;
296 int ice_candidate_pool_size = 0;
pthatcher1 2016/05/05 21:51:24 While you're in here, could you use this style fo
Taylor Brandstetter 2016/05/06 03:53:34 Done.
296 RTCConfiguration() 297 RTCConfiguration()
297 : type(kAll), 298 : type(kAll),
298 bundle_policy(kBundlePolicyBalanced), 299 bundle_policy(kBundlePolicyBalanced),
299 rtcp_mux_policy(kRtcpMuxPolicyNegotiate), 300 rtcp_mux_policy(kRtcpMuxPolicyNegotiate),
300 tcp_candidate_policy(kTcpCandidatePolicyEnabled), 301 tcp_candidate_policy(kTcpCandidatePolicyEnabled),
301 audio_jitter_buffer_max_packets(kAudioJitterBufferMaxPackets), 302 audio_jitter_buffer_max_packets(kAudioJitterBufferMaxPackets),
302 audio_jitter_buffer_fast_accelerate(false), 303 audio_jitter_buffer_fast_accelerate(false),
303 ice_connection_receiving_timeout(kUndefined), 304 ice_connection_receiving_timeout(kUndefined),
304 ice_backup_candidate_pair_ping_interval(kUndefined), 305 ice_backup_candidate_pair_ping_interval(kUndefined),
305 continual_gathering_policy(GATHER_ONCE), 306 continual_gathering_policy(GATHER_ONCE),
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695 CreatePeerConnectionFactory( 696 CreatePeerConnectionFactory(
696 rtc::Thread* worker_thread, 697 rtc::Thread* worker_thread,
697 rtc::Thread* signaling_thread, 698 rtc::Thread* signaling_thread,
698 AudioDeviceModule* default_adm, 699 AudioDeviceModule* default_adm,
699 cricket::WebRtcVideoEncoderFactory* encoder_factory, 700 cricket::WebRtcVideoEncoderFactory* encoder_factory,
700 cricket::WebRtcVideoDecoderFactory* decoder_factory); 701 cricket::WebRtcVideoDecoderFactory* decoder_factory);
701 702
702 } // namespace webrtc 703 } // namespace webrtc
703 704
704 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ 705 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_
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