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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 "webrtc/api/peerconnectioninterface.h" | 14 #include "webrtc/api/peerconnectioninterface.h" |
15 #include "webrtc/api/test/fakeaudiocapturemodule.h" | 15 #include "webrtc/api/test/fakeaudiocapturemodule.h" |
16 #include "webrtc/api/test/fakeconstraints.h" | 16 #include "webrtc/api/test/fakeconstraints.h" |
17 #include "webrtc/api/test/fakevideotrackrenderer.h" | 17 #include "webrtc/api/test/fakevideotrackrenderer.h" |
18 #include "webrtc/base/sigslot.h" | 18 #include "webrtc/base/sigslot.h" |
19 | 19 |
20 class PeerConnectionTestWrapper | 20 class PeerConnectionTestWrapper |
21 : public webrtc::PeerConnectionObserver, | 21 : public webrtc::PeerConnectionObserver, |
22 public webrtc::CreateSessionDescriptionObserver, | 22 public webrtc::CreateSessionDescriptionObserver, |
23 public sigslot::has_slots<> { | 23 public sigslot::has_slots<> { |
24 public: | 24 public: |
25 static void Connect(PeerConnectionTestWrapper* caller, | 25 static void Connect(PeerConnectionTestWrapper* caller, |
26 PeerConnectionTestWrapper* callee); | 26 PeerConnectionTestWrapper* callee); |
27 | 27 |
28 explicit PeerConnectionTestWrapper(const std::string& name); | 28 explicit PeerConnectionTestWrapper(const std::string& name, |
| 29 rtc::Thread* worker_thread); |
29 virtual ~PeerConnectionTestWrapper(); | 30 virtual ~PeerConnectionTestWrapper(); |
30 | 31 |
31 bool CreatePc(const webrtc::MediaConstraintsInterface* constraints); | 32 bool CreatePc(const webrtc::MediaConstraintsInterface* constraints); |
32 | 33 |
33 rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel( | 34 rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel( |
34 const std::string& label, | 35 const std::string& label, |
35 const webrtc::DataChannelInit& init); | 36 const webrtc::DataChannelInit& init); |
36 | 37 |
37 // Implements PeerConnectionObserver. | 38 // Implements PeerConnectionObserver. |
38 virtual void OnSignalingChange( | 39 virtual void OnSignalingChange( |
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81 void SetLocalDescription(const std::string& type, const std::string& sdp); | 82 void SetLocalDescription(const std::string& type, const std::string& sdp); |
82 void SetRemoteDescription(const std::string& type, const std::string& sdp); | 83 void SetRemoteDescription(const std::string& type, const std::string& sdp); |
83 bool CheckForConnection(); | 84 bool CheckForConnection(); |
84 bool CheckForAudio(); | 85 bool CheckForAudio(); |
85 bool CheckForVideo(); | 86 bool CheckForVideo(); |
86 rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia( | 87 rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia( |
87 bool audio, const webrtc::FakeConstraints& audio_constraints, | 88 bool audio, const webrtc::FakeConstraints& audio_constraints, |
88 bool video, const webrtc::FakeConstraints& video_constraints); | 89 bool video, const webrtc::FakeConstraints& video_constraints); |
89 | 90 |
90 std::string name_; | 91 std::string name_; |
| 92 rtc::Thread* worker_thread_; |
91 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; | 93 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; |
92 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> | 94 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> |
93 peer_connection_factory_; | 95 peer_connection_factory_; |
94 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; | 96 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; |
95 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_; | 97 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_; |
96 }; | 98 }; |
97 | 99 |
98 #endif // WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_ | 100 #endif // WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_ |
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