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

Issue 1315503003: Set the IceConnectionReceivingTimeout as a RTCConfiguration parameter. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Created 5 years, 3 months 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,
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240 // at the same time. 240 // at the same time.
241 IceServers servers; 241 IceServers servers;
242 // A localhost candidate is signaled whenever a candidate with the any 242 // A localhost candidate is signaled whenever a candidate with the any
243 // address is allocated. 243 // address is allocated.
244 bool enable_localhost_ice_candidate; 244 bool enable_localhost_ice_candidate;
245 BundlePolicy bundle_policy; 245 BundlePolicy bundle_policy;
246 RtcpMuxPolicy rtcp_mux_policy; 246 RtcpMuxPolicy rtcp_mux_policy;
247 TcpCandidatePolicy tcp_candidate_policy; 247 TcpCandidatePolicy tcp_candidate_policy;
248 int audio_jitter_buffer_max_packets; 248 int audio_jitter_buffer_max_packets;
249 bool audio_jitter_buffer_fast_accelerate; 249 bool audio_jitter_buffer_fast_accelerate;
250 // ICE connection receiving timeout value in milliseconds.
251 int ice_connection_receiving_timeout;
250 std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates; 252 std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
251 253
252 RTCConfiguration() 254 RTCConfiguration()
253 : type(kAll), 255 : type(kAll),
254 enable_localhost_ice_candidate(false), 256 enable_localhost_ice_candidate(false),
255 bundle_policy(kBundlePolicyBalanced), 257 bundle_policy(kBundlePolicyBalanced),
256 rtcp_mux_policy(kRtcpMuxPolicyNegotiate), 258 rtcp_mux_policy(kRtcpMuxPolicyNegotiate),
257 tcp_candidate_policy(kTcpCandidatePolicyEnabled), 259 tcp_candidate_policy(kTcpCandidatePolicyEnabled),
258 audio_jitter_buffer_max_packets(50), 260 audio_jitter_buffer_max_packets(50),
juberti1 2015/08/26 18:25:46 let's put these as constants in the RTCConfigurati
honghaiz3 2015/08/26 19:20:26 Done (for integer parameters).
259 audio_jitter_buffer_fast_accelerate(false) {} 261 audio_jitter_buffer_fast_accelerate(false),
262 ice_connection_receiving_timeout(2500) {}
260 }; 263 };
261 264
262 struct RTCOfferAnswerOptions { 265 struct RTCOfferAnswerOptions {
263 static const int kUndefined = -1; 266 static const int kUndefined = -1;
264 static const int kMaxOfferToReceiveMedia = 1; 267 static const int kMaxOfferToReceiveMedia = 1;
265 268
266 // The default value for constraint offerToReceiveX:true. 269 // The default value for constraint offerToReceiveX:true.
267 static const int kOfferToReceiveMediaTrue = 1; 270 static const int kOfferToReceiveMediaTrue = 1;
268 271
269 int offer_to_receive_video; 272 int offer_to_receive_video;
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351 // Sets the local session description. 354 // Sets the local session description.
352 // JsepInterface takes the ownership of |desc| even if it fails. 355 // JsepInterface takes the ownership of |desc| even if it fails.
353 // The |observer| callback will be called when done. 356 // The |observer| callback will be called when done.
354 virtual void SetLocalDescription(SetSessionDescriptionObserver* observer, 357 virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
355 SessionDescriptionInterface* desc) = 0; 358 SessionDescriptionInterface* desc) = 0;
356 // Sets the remote session description. 359 // Sets the remote session description.
357 // JsepInterface takes the ownership of |desc| even if it fails. 360 // JsepInterface takes the ownership of |desc| even if it fails.
358 // The |observer| callback will be called when done. 361 // The |observer| callback will be called when done.
359 virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer, 362 virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
360 SessionDescriptionInterface* desc) = 0; 363 SessionDescriptionInterface* desc) = 0;
361 // Sets the ICE connection receiving timeout value in milliseconds.
362 virtual void SetIceConnectionReceivingTimeout(int timeout_ms) {}
363 // Restarts or updates the ICE Agent process of gathering local candidates 364 // Restarts or updates the ICE Agent process of gathering local candidates
364 // and pinging remote candidates. 365 // and pinging remote candidates.
365 virtual bool UpdateIce(const IceServers& configuration, 366 virtual bool UpdateIce(const IceServers& configuration,
366 const MediaConstraintsInterface* constraints) = 0; 367 const MediaConstraintsInterface* constraints) = 0;
367 // Provides a remote candidate to the ICE Agent. 368 // Provides a remote candidate to the ICE Agent.
368 // A copy of the |candidate| will be created and added to the remote 369 // A copy of the |candidate| will be created and added to the remote
369 // description. So the caller of this method still has the ownership of the 370 // description. So the caller of this method still has the ownership of the
370 // |candidate|. 371 // |candidate|.
371 // TODO(ronghuawu): Consider to change this so that the AddIceCandidate will 372 // TODO(ronghuawu): Consider to change this so that the AddIceCandidate will
372 // take the ownership of the |candidate|. 373 // take the ownership of the |candidate|.
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601 CreatePeerConnectionFactory( 602 CreatePeerConnectionFactory(
602 rtc::Thread* worker_thread, 603 rtc::Thread* worker_thread,
603 rtc::Thread* signaling_thread, 604 rtc::Thread* signaling_thread,
604 AudioDeviceModule* default_adm, 605 AudioDeviceModule* default_adm,
605 cricket::WebRtcVideoEncoderFactory* encoder_factory, 606 cricket::WebRtcVideoEncoderFactory* encoder_factory,
606 cricket::WebRtcVideoDecoderFactory* decoder_factory); 607 cricket::WebRtcVideoDecoderFactory* decoder_factory);
607 608
608 } // namespace webrtc 609 } // namespace webrtc
609 610
610 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ 611 #endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
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