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

Issue 1765423005: Change VideoRtpReceiver to create remote VideoTrack and VideoTrackSource. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixed up the comments. Created 4 years, 9 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 15
16 #include "webrtc/api/dtlsidentitystore.h" 16 #include "webrtc/api/dtlsidentitystore.h"
17 #include "webrtc/api/peerconnectionfactory.h" 17 #include "webrtc/api/peerconnectionfactory.h"
18 #include "webrtc/api/peerconnectioninterface.h" 18 #include "webrtc/api/peerconnectioninterface.h"
19 #include "webrtc/api/rtpreceiverinterface.h" 19 #include "webrtc/api/rtpreceiverinterface.h"
20 #include "webrtc/api/rtpsenderinterface.h" 20 #include "webrtc/api/rtpsenderinterface.h"
21 #include "webrtc/api/statscollector.h" 21 #include "webrtc/api/statscollector.h"
22 #include "webrtc/api/streamcollection.h" 22 #include "webrtc/api/streamcollection.h"
23 #include "webrtc/api/webrtcsession.h" 23 #include "webrtc/api/webrtcsession.h"
24 #include "webrtc/base/scoped_ptr.h" 24 #include "webrtc/base/scoped_ptr.h"
25 25
26 namespace webrtc { 26 namespace webrtc {
27 27
28 class MediaStreamObserver; 28 class MediaStreamObserver;
29 class RemoteMediaStreamFactory; 29 class RemoteMediaStreamFactory;
30 class VideoRtpReceiver;
30 31
31 // Populates |session_options| from |rtc_options|, and returns true if options 32 // Populates |session_options| from |rtc_options|, and returns true if options
32 // are valid. 33 // are valid.
33 // |session_options|->transport_options map entries must exist in order for 34 // |session_options|->transport_options map entries must exist in order for
34 // them to be populated from |rtc_options|. 35 // them to be populated from |rtc_options|.
35 bool ExtractMediaSessionOptions( 36 bool ExtractMediaSessionOptions(
36 const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options, 37 const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
37 cricket::MediaSessionOptions* session_options); 38 cricket::MediaSessionOptions* session_options);
38 39
39 // Populates |session_options| from |constraints|, and returns true if all 40 // Populates |session_options| from |constraints|, and returns true if all
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160 uint32_t ssrc; 161 uint32_t ssrc;
161 }; 162 };
162 typedef std::vector<TrackInfo> TrackInfos; 163 typedef std::vector<TrackInfo> TrackInfos;
163 164
164 // Implements MessageHandler. 165 // Implements MessageHandler.
165 void OnMessage(rtc::Message* msg) override; 166 void OnMessage(rtc::Message* msg) override;
166 167
167 void CreateAudioReceiver(MediaStreamInterface* stream, 168 void CreateAudioReceiver(MediaStreamInterface* stream,
168 AudioTrackInterface* audio_track, 169 AudioTrackInterface* audio_track,
169 uint32_t ssrc); 170 uint32_t ssrc);
171
170 void CreateVideoReceiver(MediaStreamInterface* stream, 172 void CreateVideoReceiver(MediaStreamInterface* stream,
171 VideoTrackInterface* video_track, 173 const std::string& track_id,
172 uint32_t ssrc); 174 uint32_t ssrc);
173 void DestroyAudioReceiver(MediaStreamInterface* stream, 175 void DestroyAudioReceiver(MediaStreamInterface* stream,
174 AudioTrackInterface* audio_track); 176 AudioTrackInterface* audio_track);
175 void DestroyVideoReceiver(MediaStreamInterface* stream, 177 void DestroyVideoReceiver(MediaStreamInterface* stream,
176 VideoTrackInterface* video_track); 178 VideoTrackInterface* video_track);
177 void DestroyAudioSender(MediaStreamInterface* stream, 179 void DestroyAudioSender(MediaStreamInterface* stream,
178 AudioTrackInterface* audio_track, 180 AudioTrackInterface* audio_track,
179 uint32_t ssrc); 181 uint32_t ssrc);
180 void DestroyVideoSender(MediaStreamInterface* stream, 182 void DestroyVideoSender(MediaStreamInterface* stream,
181 VideoTrackInterface* video_track); 183 VideoTrackInterface* video_track);
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396 // because its destruction fires signals (such as VoiceChannelDestroyed) 398 // because its destruction fires signals (such as VoiceChannelDestroyed)
397 // which will trigger some final actions in PeerConnection... 399 // which will trigger some final actions in PeerConnection...
398 rtc::scoped_ptr<WebRtcSession> session_; 400 rtc::scoped_ptr<WebRtcSession> session_;
399 // ... But stats_ depends on session_ so it should be destroyed even earlier. 401 // ... But stats_ depends on session_ so it should be destroyed even earlier.
400 rtc::scoped_ptr<StatsCollector> stats_; 402 rtc::scoped_ptr<StatsCollector> stats_;
401 }; 403 };
402 404
403 } // namespace webrtc 405 } // namespace webrtc
404 406
405 #endif // WEBRTC_API_PEERCONNECTION_H_ 407 #endif // WEBRTC_API_PEERCONNECTION_H_
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