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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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| 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 | 36 |
| 37 namespace webrtc { | |
| 38 class DtlsIdentityStoreInterface; | |
| 39 class PortAllocatorFactoryInterface; | |
| 40 } | |
| 41 | |
| 42 class PeerConnectionTestWrapper | 37 class PeerConnectionTestWrapper |
| 43 : public webrtc::PeerConnectionObserver, | 38 : public webrtc::PeerConnectionObserver, |
| 44 public webrtc::CreateSessionDescriptionObserver, | 39 public webrtc::CreateSessionDescriptionObserver, |
| 45 public sigslot::has_slots<> { | 40 public sigslot::has_slots<> { |
| 46 public: | 41 public: |
| 47 static void Connect(PeerConnectionTestWrapper* caller, | 42 static void Connect(PeerConnectionTestWrapper* caller, |
| 48 PeerConnectionTestWrapper* callee); | 43 PeerConnectionTestWrapper* callee); |
| 49 | 44 |
| 50 explicit PeerConnectionTestWrapper(const std::string& name); | 45 explicit PeerConnectionTestWrapper(const std::string& name); |
| 51 virtual ~PeerConnectionTestWrapper(); | 46 virtual ~PeerConnectionTestWrapper(); |
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| 103 void SetLocalDescription(const std::string& type, const std::string& sdp); | 98 void SetLocalDescription(const std::string& type, const std::string& sdp); |
| 104 void SetRemoteDescription(const std::string& type, const std::string& sdp); | 99 void SetRemoteDescription(const std::string& type, const std::string& sdp); |
| 105 bool CheckForConnection(); | 100 bool CheckForConnection(); |
| 106 bool CheckForAudio(); | 101 bool CheckForAudio(); |
| 107 bool CheckForVideo(); | 102 bool CheckForVideo(); |
| 108 rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia( | 103 rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia( |
| 109 bool audio, const webrtc::FakeConstraints& audio_constraints, | 104 bool audio, const webrtc::FakeConstraints& audio_constraints, |
| 110 bool video, const webrtc::FakeConstraints& video_constraints); | 105 bool video, const webrtc::FakeConstraints& video_constraints); |
| 111 | 106 |
| 112 std::string name_; | 107 std::string name_; |
| 113 rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface> | |
| 114 allocator_factory_; | |
| 115 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; | 108 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; |
| 116 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> | 109 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> |
| 117 peer_connection_factory_; | 110 peer_connection_factory_; |
| 118 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; | 111 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; |
| 119 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_; | 112 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_; |
| 120 }; | 113 }; |
| 121 | 114 |
| 122 #endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ | 115 #endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ |
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