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Side by Side Diff: talk/app/webrtc/webrtcsession.h

Issue 1176383004: DtlsIdentityStore[Interface/Impl] updated, DtlsIdentityService to be removed (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Merge w master AFTER the landing of 1268363002. "CreatePC(service,store)" using store instead of service. Created 5 years, 4 months ago
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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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