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Side by Side Diff: talk/app/webrtc/peerconnectioninterface.h

Issue 1455033004: Ping backup connection at a slower rate (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Merge with head, address comments Created 5 years ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2012 Google Inc. 3 * Copyright 2012 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
(...skipping 235 matching lines...) Expand 10 before | Expand all | Expand 10 after
246 // Chromium at the same time. 246 // Chromium at the same time.
247 IceTransportsType type; 247 IceTransportsType type;
248 // TODO(pthatcher): Rename this ice_servers, but update Chromium 248 // TODO(pthatcher): Rename this ice_servers, but update Chromium
249 // at the same time. 249 // at the same time.
250 IceServers servers; 250 IceServers servers;
251 BundlePolicy bundle_policy; 251 BundlePolicy bundle_policy;
252 RtcpMuxPolicy rtcp_mux_policy; 252 RtcpMuxPolicy rtcp_mux_policy;
253 TcpCandidatePolicy tcp_candidate_policy; 253 TcpCandidatePolicy tcp_candidate_policy;
254 int audio_jitter_buffer_max_packets; 254 int audio_jitter_buffer_max_packets;
255 bool audio_jitter_buffer_fast_accelerate; 255 bool audio_jitter_buffer_fast_accelerate;
256 int ice_connection_receiving_timeout; 256 int ice_connection_receiving_timeout; // ms
257 int ice_backup_connection_ping_interval; // ms
257 ContinualGatheringPolicy continual_gathering_policy; 258 ContinualGatheringPolicy continual_gathering_policy;
258 std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates; 259 std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
259 bool disable_prerenderer_smoothing; 260 bool disable_prerenderer_smoothing;
260 RTCConfiguration() 261 RTCConfiguration()
261 : type(kAll), 262 : type(kAll),
262 bundle_policy(kBundlePolicyBalanced), 263 bundle_policy(kBundlePolicyBalanced),
263 rtcp_mux_policy(kRtcpMuxPolicyNegotiate), 264 rtcp_mux_policy(kRtcpMuxPolicyNegotiate),
264 tcp_candidate_policy(kTcpCandidatePolicyEnabled), 265 tcp_candidate_policy(kTcpCandidatePolicyEnabled),
265 audio_jitter_buffer_max_packets(kAudioJitterBufferMaxPackets), 266 audio_jitter_buffer_max_packets(kAudioJitterBufferMaxPackets),
266 audio_jitter_buffer_fast_accelerate(false), 267 audio_jitter_buffer_fast_accelerate(false),
267 ice_connection_receiving_timeout(kUndefined), 268 ice_connection_receiving_timeout(kUndefined),
269 ice_backup_connection_ping_interval(kUndefined),
268 continual_gathering_policy(GATHER_ONCE), 270 continual_gathering_policy(GATHER_ONCE),
269 disable_prerenderer_smoothing(false) {} 271 disable_prerenderer_smoothing(false) {}
270 }; 272 };
271 273
272 struct RTCOfferAnswerOptions { 274 struct RTCOfferAnswerOptions {
273 static const int kUndefined = -1; 275 static const int kUndefined = -1;
274 static const int kMaxOfferToReceiveMedia = 1; 276 static const int kMaxOfferToReceiveMedia = 1;
275 277
276 // The default value for constraint offerToReceiveX:true. 278 // The default value for constraint offerToReceiveX:true.
277 static const int kOfferToReceiveMediaTrue = 1; 279 static const int kOfferToReceiveMediaTrue = 1;
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661 CreatePeerConnectionFactory( 663 CreatePeerConnectionFactory(
662 rtc::Thread* worker_thread, 664 rtc::Thread* worker_thread,
663 rtc::Thread* signaling_thread, 665 rtc::Thread* signaling_thread,
664 AudioDeviceModule* default_adm, 666 AudioDeviceModule* default_adm,
665 cricket::WebRtcVideoEncoderFactory* encoder_factory, 667 cricket::WebRtcVideoEncoderFactory* encoder_factory,
666 cricket::WebRtcVideoDecoderFactory* decoder_factory); 668 cricket::WebRtcVideoDecoderFactory* decoder_factory);
667 669
668 } // namespace webrtc 670 } // namespace webrtc
669 671
670 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ 672 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
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