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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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