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

Issue 1871993002: Only generate one CNAME per PeerConnection. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Renaming: cname -> rtcp_cname. Modified the peerconnectioninterface unit tests. 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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362 UMAObserver* uma_observer_; 362 UMAObserver* uma_observer_;
363 SignalingState signaling_state_; 363 SignalingState signaling_state_;
364 // TODO(bemasc): Remove ice_state_. 364 // TODO(bemasc): Remove ice_state_.
365 IceState ice_state_; 365 IceState ice_state_;
366 IceConnectionState ice_connection_state_; 366 IceConnectionState ice_connection_state_;
367 IceGatheringState ice_gathering_state_; 367 IceGatheringState ice_gathering_state_;
368 368
369 std::unique_ptr<cricket::PortAllocator> port_allocator_; 369 std::unique_ptr<cricket::PortAllocator> port_allocator_;
370 std::unique_ptr<MediaControllerInterface> media_controller_; 370 std::unique_ptr<MediaControllerInterface> media_controller_;
371 371
372 // One PeerConnection has only one RTCP CNAME.
373 // https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-26#section-4.9
374 std::string rtcp_cname_;
375
372 // Streams added via AddStream. 376 // Streams added via AddStream.
373 rtc::scoped_refptr<StreamCollection> local_streams_; 377 rtc::scoped_refptr<StreamCollection> local_streams_;
374 // Streams created as a result of SetRemoteDescription. 378 // Streams created as a result of SetRemoteDescription.
375 rtc::scoped_refptr<StreamCollection> remote_streams_; 379 rtc::scoped_refptr<StreamCollection> remote_streams_;
376 380
377 std::vector<std::unique_ptr<MediaStreamObserver>> stream_observers_; 381 std::vector<std::unique_ptr<MediaStreamObserver>> stream_observers_;
378 382
379 // These lists store track info seen in local/remote descriptions. 383 // These lists store track info seen in local/remote descriptions.
380 TrackInfos remote_audio_tracks_; 384 TrackInfos remote_audio_tracks_;
381 TrackInfos remote_video_tracks_; 385 TrackInfos remote_video_tracks_;
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397 // because its destruction fires signals (such as VoiceChannelDestroyed) 401 // because its destruction fires signals (such as VoiceChannelDestroyed)
398 // which will trigger some final actions in PeerConnection... 402 // which will trigger some final actions in PeerConnection...
399 std::unique_ptr<WebRtcSession> session_; 403 std::unique_ptr<WebRtcSession> session_;
400 // ... But stats_ depends on session_ so it should be destroyed even earlier. 404 // ... But stats_ depends on session_ so it should be destroyed even earlier.
401 std::unique_ptr<StatsCollector> stats_; 405 std::unique_ptr<StatsCollector> stats_;
402 }; 406 };
403 407
404 } // namespace webrtc 408 } // namespace webrtc
405 409
406 #endif // WEBRTC_API_PEERCONNECTION_H_ 410 #endif // WEBRTC_API_PEERCONNECTION_H_
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