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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 |
11 #ifndef WEBRTC_API_PEERCONNECTION_H_ | 11 #ifndef WEBRTC_API_PEERCONNECTION_H_ |
12 #define WEBRTC_API_PEERCONNECTION_H_ | 12 #define WEBRTC_API_PEERCONNECTION_H_ |
13 | 13 |
14 #include <string> | 14 #include <string> |
15 #include <map> | 15 #include <map> |
16 #include <vector> | 16 #include <vector> |
17 | 17 |
18 #include "webrtc/api/dtlsidentitystore.h" | 18 #include "webrtc/api/dtlsidentitystore.h" |
19 #include "webrtc/api/peerconnectionfactory.h" | 19 #include "webrtc/api/peerconnectionfactory.h" |
20 #include "webrtc/api/peerconnectioninterface.h" | 20 #include "webrtc/api/peerconnectioninterface.h" |
21 #include "webrtc/api/rtpreceiverinterface.h" | 21 #include "webrtc/api/rtpreceiverinterface.h" |
22 #include "webrtc/api/rtpsenderinterface.h" | 22 #include "webrtc/api/rtpsenderinterface.h" |
23 #include "webrtc/api/statscollector.h" | 23 #include "webrtc/api/statscollector.h" |
24 #include "webrtc/api/streamcollection.h" | 24 #include "webrtc/api/streamcollection.h" |
25 #include "webrtc/api/webrtcsession.h" | 25 #include "webrtc/api/webrtcsession.h" |
26 #include "webrtc/base/scoped_ptr.h" | 26 #include "webrtc/base/scoped_ptr.h" |
27 | 27 |
28 #ifdef USE_QUIC | |
pthatcher1
2016/03/30 20:34:48
I think our naming convention is HAVE_QUIC, but I
mikescarlett
2016/04/05 19:58:49
I renamed it HAVE_QUIC because HAVE_SCTP is used f
| |
29 #include "webrtc/api/quicdatachannel.h" | |
30 #include "webrtc/api/quictransport.h" | |
31 #endif // USE_QUIC | |
32 | |
28 namespace webrtc { | 33 namespace webrtc { |
29 | 34 |
30 class MediaStreamObserver; | 35 class MediaStreamObserver; |
31 class VideoRtpReceiver; | 36 class VideoRtpReceiver; |
32 | 37 |
33 // Populates |session_options| from |rtc_options|, and returns true if options | 38 // Populates |session_options| from |rtc_options|, and returns true if options |
34 // are valid. | 39 // are valid. |
35 // |session_options|->transport_options map entries must exist in order for | 40 // |session_options|->transport_options map entries must exist in order for |
36 // them to be populated from |rtc_options|. | 41 // them to be populated from |rtc_options|. |
37 bool ExtractMediaSessionOptions( | 42 bool ExtractMediaSessionOptions( |
(...skipping 342 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
380 TrackInfos remote_audio_tracks_; | 385 TrackInfos remote_audio_tracks_; |
381 TrackInfos remote_video_tracks_; | 386 TrackInfos remote_video_tracks_; |
382 TrackInfos local_audio_tracks_; | 387 TrackInfos local_audio_tracks_; |
383 TrackInfos local_video_tracks_; | 388 TrackInfos local_video_tracks_; |
384 | 389 |
385 SctpSidAllocator sid_allocator_; | 390 SctpSidAllocator sid_allocator_; |
386 // label -> DataChannel | 391 // label -> DataChannel |
387 std::map<std::string, rtc::scoped_refptr<DataChannel>> rtp_data_channels_; | 392 std::map<std::string, rtc::scoped_refptr<DataChannel>> rtp_data_channels_; |
388 std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_; | 393 std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_; |
389 std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_to_free_; | 394 std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_to_free_; |
395 #ifdef USE_QUIC | |
396 std::vector<rtc::scoped_refptr<QuicDataChannel>> quic_data_channels_; | |
397 rtc::scoped_ptr<QuicTransport> quic_transport_; | |
398 #endif // USE_QUIC | |
390 | 399 |
391 bool remote_peer_supports_msid_ = false; | 400 bool remote_peer_supports_msid_ = false; |
392 | 401 |
393 std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders_; | 402 std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders_; |
394 std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers_; | 403 std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers_; |
395 | 404 |
396 // The session_ scoped_ptr is declared at the bottom of PeerConnection | 405 // The session_ scoped_ptr is declared at the bottom of PeerConnection |
397 // because its destruction fires signals (such as VoiceChannelDestroyed) | 406 // because its destruction fires signals (such as VoiceChannelDestroyed) |
398 // which will trigger some final actions in PeerConnection... | 407 // which will trigger some final actions in PeerConnection... |
399 rtc::scoped_ptr<WebRtcSession> session_; | 408 rtc::scoped_ptr<WebRtcSession> session_; |
400 // ... But stats_ depends on session_ so it should be destroyed even earlier. | 409 // ... But stats_ depends on session_ so it should be destroyed even earlier. |
401 rtc::scoped_ptr<StatsCollector> stats_; | 410 rtc::scoped_ptr<StatsCollector> stats_; |
402 }; | 411 }; |
403 | 412 |
404 } // namespace webrtc | 413 } // namespace webrtc |
405 | 414 |
406 #endif // WEBRTC_API_PEERCONNECTION_H_ | 415 #endif // WEBRTC_API_PEERCONNECTION_H_ |
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