OLD | NEW |
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, |
(...skipping 15 matching lines...) Expand all Loading... |
26 */ | 26 */ |
27 | 27 |
28 #ifndef TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ | 28 #ifndef TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ |
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" | |
37 | 36 |
38 namespace webrtc { | 37 namespace webrtc { |
39 class DtlsIdentityStoreInterface; | 38 class DtlsIdentityStoreInterface; |
40 class PortAllocatorFactoryInterface; | 39 class PortAllocatorFactoryInterface; |
41 } | 40 } |
42 | 41 |
43 class PeerConnectionTestWrapper | 42 class PeerConnectionTestWrapper |
44 : public webrtc::PeerConnectionObserver, | 43 : public webrtc::PeerConnectionObserver, |
45 public webrtc::CreateSessionDescriptionObserver, | 44 public webrtc::CreateSessionDescriptionObserver, |
46 public sigslot::has_slots<> { | 45 public sigslot::has_slots<> { |
(...skipping 67 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
114 rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface> | 113 rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface> |
115 allocator_factory_; | 114 allocator_factory_; |
116 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; | 115 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; |
117 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> | 116 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> |
118 peer_connection_factory_; | 117 peer_connection_factory_; |
119 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; | 118 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; |
120 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_; | 119 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_; |
121 }; | 120 }; |
122 | 121 |
123 #endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ | 122 #endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ |
OLD | NEW |