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1 /* | 1 /* |
2 * Copyright 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2013 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #ifndef WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_ | 11 #ifndef WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_ |
12 #define WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_ | 12 #define WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_ |
13 | 13 |
| 14 #include <memory> |
| 15 |
14 #include "webrtc/api/peerconnectioninterface.h" | 16 #include "webrtc/api/peerconnectioninterface.h" |
15 #include "webrtc/api/test/fakeaudiocapturemodule.h" | 17 #include "webrtc/api/test/fakeaudiocapturemodule.h" |
16 #include "webrtc/api/test/fakeconstraints.h" | 18 #include "webrtc/api/test/fakeconstraints.h" |
17 #include "webrtc/api/test/fakevideotrackrenderer.h" | 19 #include "webrtc/api/test/fakevideotrackrenderer.h" |
18 #include "webrtc/base/sigslot.h" | 20 #include "webrtc/base/sigslot.h" |
19 | 21 |
20 class PeerConnectionTestWrapper | 22 class PeerConnectionTestWrapper |
21 : public webrtc::PeerConnectionObserver, | 23 : public webrtc::PeerConnectionObserver, |
22 public webrtc::CreateSessionDescriptionObserver, | 24 public webrtc::CreateSessionDescriptionObserver, |
23 public sigslot::has_slots<> { | 25 public sigslot::has_slots<> { |
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87 rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia( | 89 rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia( |
88 bool audio, const webrtc::FakeConstraints& audio_constraints, | 90 bool audio, const webrtc::FakeConstraints& audio_constraints, |
89 bool video, const webrtc::FakeConstraints& video_constraints); | 91 bool video, const webrtc::FakeConstraints& video_constraints); |
90 | 92 |
91 std::string name_; | 93 std::string name_; |
92 rtc::Thread* worker_thread_; | 94 rtc::Thread* worker_thread_; |
93 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; | 95 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; |
94 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> | 96 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> |
95 peer_connection_factory_; | 97 peer_connection_factory_; |
96 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; | 98 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; |
97 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_; | 99 std::unique_ptr<webrtc::FakeVideoTrackRenderer> renderer_; |
98 }; | 100 }; |
99 | 101 |
100 #endif // WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_ | 102 #endif // WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_ |
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