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1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2013 Google Inc. | 3 * Copyright 2013 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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29 #define TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ | 29 #define TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ |
30 | 30 |
31 #include "talk/app/webrtc/peerconnectioninterface.h" | 31 #include "talk/app/webrtc/peerconnectioninterface.h" |
32 #include "talk/app/webrtc/test/fakeaudiocapturemodule.h" | 32 #include "talk/app/webrtc/test/fakeaudiocapturemodule.h" |
33 #include "talk/app/webrtc/test/fakeconstraints.h" | 33 #include "talk/app/webrtc/test/fakeconstraints.h" |
34 #include "talk/app/webrtc/test/fakevideotrackrenderer.h" | 34 #include "talk/app/webrtc/test/fakevideotrackrenderer.h" |
35 #include "webrtc/base/sigslot.h" | 35 #include "webrtc/base/sigslot.h" |
36 #include "webrtc/base/thread.h" | 36 #include "webrtc/base/thread.h" |
37 | 37 |
38 namespace webrtc { | 38 namespace webrtc { |
| 39 class DtlsIdentityStoreInterface; |
39 class PortAllocatorFactoryInterface; | 40 class PortAllocatorFactoryInterface; |
40 } | 41 } |
41 | 42 |
42 class PeerConnectionTestWrapper | 43 class PeerConnectionTestWrapper |
43 : public webrtc::PeerConnectionObserver, | 44 : public webrtc::PeerConnectionObserver, |
44 public webrtc::CreateSessionDescriptionObserver, | 45 public webrtc::CreateSessionDescriptionObserver, |
45 public sigslot::has_slots<> { | 46 public sigslot::has_slots<> { |
46 public: | 47 public: |
47 static void Connect(PeerConnectionTestWrapper* caller, | 48 static void Connect(PeerConnectionTestWrapper* caller, |
48 PeerConnectionTestWrapper* callee); | 49 PeerConnectionTestWrapper* callee); |
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113 rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface> | 114 rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface> |
114 allocator_factory_; | 115 allocator_factory_; |
115 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; | 116 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; |
116 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> | 117 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> |
117 peer_connection_factory_; | 118 peer_connection_factory_; |
118 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; | 119 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; |
119 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_; | 120 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_; |
120 }; | 121 }; |
121 | 122 |
122 #endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ | 123 #endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ |
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