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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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