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1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2012 Google Inc. | 3 * Copyright 2012 Google Inc. |
4 * | 4 * |
5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
7 * | 7 * |
8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
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32 | 32 |
33 #include "talk/app/webrtc/datachannel.h" | 33 #include "talk/app/webrtc/datachannel.h" |
34 #include "talk/app/webrtc/dtmfsender.h" | 34 #include "talk/app/webrtc/dtmfsender.h" |
35 #include "talk/app/webrtc/mediastreamprovider.h" | 35 #include "talk/app/webrtc/mediastreamprovider.h" |
36 #include "talk/app/webrtc/peerconnectioninterface.h" | 36 #include "talk/app/webrtc/peerconnectioninterface.h" |
37 #include "talk/app/webrtc/statstypes.h" | 37 #include "talk/app/webrtc/statstypes.h" |
38 #include "talk/media/base/mediachannel.h" | 38 #include "talk/media/base/mediachannel.h" |
39 #include "webrtc/p2p/base/session.h" | 39 #include "webrtc/p2p/base/session.h" |
40 #include "talk/session/media/mediasession.h" | 40 #include "talk/session/media/mediasession.h" |
41 #include "webrtc/base/sigslot.h" | 41 #include "webrtc/base/sigslot.h" |
| 42 #include "webrtc/base/sslidentity.h" |
42 #include "webrtc/base/thread.h" | 43 #include "webrtc/base/thread.h" |
43 | 44 |
44 namespace cricket { | 45 namespace cricket { |
45 | 46 |
46 class BaseChannel; | 47 class BaseChannel; |
47 class ChannelManager; | 48 class ChannelManager; |
48 class DataChannel; | 49 class DataChannel; |
49 class StatsReport; | 50 class StatsReport; |
50 class Transport; | 51 class Transport; |
51 class VideoCapturer; | 52 class VideoCapturer; |
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118 WebRtcSession(cricket::ChannelManager* channel_manager, | 119 WebRtcSession(cricket::ChannelManager* channel_manager, |
119 rtc::Thread* signaling_thread, | 120 rtc::Thread* signaling_thread, |
120 rtc::Thread* worker_thread, | 121 rtc::Thread* worker_thread, |
121 cricket::PortAllocator* port_allocator, | 122 cricket::PortAllocator* port_allocator, |
122 MediaStreamSignaling* mediastream_signaling); | 123 MediaStreamSignaling* mediastream_signaling); |
123 virtual ~WebRtcSession(); | 124 virtual ~WebRtcSession(); |
124 | 125 |
125 bool Initialize( | 126 bool Initialize( |
126 const PeerConnectionFactoryInterface::Options& options, | 127 const PeerConnectionFactoryInterface::Options& options, |
127 const MediaConstraintsInterface* constraints, | 128 const MediaConstraintsInterface* constraints, |
128 DTLSIdentityServiceInterface* dtls_identity_service, | 129 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store, |
129 const PeerConnectionInterface::RTCConfiguration& rtc_configuration); | 130 const PeerConnectionInterface::RTCConfiguration& rtc_configuration); |
130 // Deletes the voice, video and data channel and changes the session state | 131 // Deletes the voice, video and data channel and changes the session state |
131 // to STATE_RECEIVEDTERMINATE. | 132 // to STATE_RECEIVEDTERMINATE. |
132 void Terminate(); | 133 void Terminate(); |
133 | 134 |
134 void RegisterIceObserver(IceObserver* observer) { | 135 void RegisterIceObserver(IceObserver* observer) { |
135 ice_observer_ = observer; | 136 ice_observer_ = observer; |
136 } | 137 } |
137 | 138 |
138 virtual cricket::VoiceChannel* voice_channel() { | 139 virtual cricket::VoiceChannel* voice_channel() { |
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409 PeerConnectionInterface::BundlePolicy bundle_policy_; | 410 PeerConnectionInterface::BundlePolicy bundle_policy_; |
410 | 411 |
411 // Declares the RTCP mux policy for the WebRTCSession. | 412 // Declares the RTCP mux policy for the WebRTCSession. |
412 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_; | 413 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_; |
413 | 414 |
414 DISALLOW_COPY_AND_ASSIGN(WebRtcSession); | 415 DISALLOW_COPY_AND_ASSIGN(WebRtcSession); |
415 }; | 416 }; |
416 } // namespace webrtc | 417 } // namespace webrtc |
417 | 418 |
418 #endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_ | 419 #endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_ |
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