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

Issue 1844803002: Modify PeerConnection for end-to-end QuicDataChannel usage (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 8 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
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
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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