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Side by Side Diff: talk/app/webrtc/test/peerconnectiontestwrapper.h

Issue 1151943005: Ability to specify KeyType (RSA, ECDSA) for SSLIdentity generation in libjingle (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressing ASAN, LSAN issues in unittests Created 5 years, 6 months ago
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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);
(...skipping 57 matching lines...) Expand 10 before | Expand all | Expand 10 after
106 bool CheckForAudio(); 107 bool CheckForAudio();
107 bool CheckForVideo(); 108 bool CheckForVideo();
108 rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia( 109 rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia(
109 bool audio, const webrtc::FakeConstraints& audio_constraints, 110 bool audio, const webrtc::FakeConstraints& audio_constraints,
110 bool video, const webrtc::FakeConstraints& video_constraints); 111 bool video, const webrtc::FakeConstraints& video_constraints);
111 112
112 std::string name_; 113 std::string name_;
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_;
117 rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store_;
116 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> 118 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
117 peer_connection_factory_; 119 peer_connection_factory_;
118 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; 120 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
119 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_; 121 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
120 }; 122 };
121 123
122 #endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ 124 #endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
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