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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_PC_TEST_PEERCONNECTIONTESTWRAPPER_H_ | 11 #ifndef WEBRTC_PC_TEST_PEERCONNECTIONTESTWRAPPER_H_ |
| 12 #define WEBRTC_PC_TEST_PEERCONNECTIONTESTWRAPPER_H_ | 12 #define WEBRTC_PC_TEST_PEERCONNECTIONTESTWRAPPER_H_ |
| 13 | 13 |
| 14 #include <memory> | 14 #include <memory> |
| 15 | 15 |
| 16 #include "webrtc/api/peerconnectioninterface.h" | 16 #include "webrtc/api/peerconnectioninterface.h" |
| 17 #include "webrtc/api/test/fakeconstraints.h" | 17 #include "webrtc/api/test/fakeconstraints.h" |
| 18 #include "webrtc/base/sigslot.h" |
| 18 #include "webrtc/pc/test/fakeaudiocapturemodule.h" | 19 #include "webrtc/pc/test/fakeaudiocapturemodule.h" |
| 19 #include "webrtc/pc/test/fakevideotrackrenderer.h" | 20 #include "webrtc/pc/test/fakevideotrackrenderer.h" |
| 20 #include "webrtc/rtc_base/sigslot.h" | |
| 21 | 21 |
| 22 class PeerConnectionTestWrapper | 22 class PeerConnectionTestWrapper |
| 23 : public webrtc::PeerConnectionObserver, | 23 : public webrtc::PeerConnectionObserver, |
| 24 public webrtc::CreateSessionDescriptionObserver, | 24 public webrtc::CreateSessionDescriptionObserver, |
| 25 public sigslot::has_slots<> { | 25 public sigslot::has_slots<> { |
| 26 public: | 26 public: |
| 27 static void Connect(PeerConnectionTestWrapper* caller, | 27 static void Connect(PeerConnectionTestWrapper* caller, |
| 28 PeerConnectionTestWrapper* callee); | 28 PeerConnectionTestWrapper* callee); |
| 29 | 29 |
| 30 PeerConnectionTestWrapper(const std::string& name, | 30 PeerConnectionTestWrapper(const std::string& name, |
| (...skipping 70 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 101 rtc::Thread* const network_thread_; | 101 rtc::Thread* const network_thread_; |
| 102 rtc::Thread* const worker_thread_; | 102 rtc::Thread* const worker_thread_; |
| 103 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; | 103 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; |
| 104 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> | 104 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> |
| 105 peer_connection_factory_; | 105 peer_connection_factory_; |
| 106 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; | 106 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; |
| 107 std::unique_ptr<webrtc::FakeVideoTrackRenderer> renderer_; | 107 std::unique_ptr<webrtc::FakeVideoTrackRenderer> renderer_; |
| 108 }; | 108 }; |
| 109 | 109 |
| 110 #endif // WEBRTC_PC_TEST_PEERCONNECTIONTESTWRAPPER_H_ | 110 #endif // WEBRTC_PC_TEST_PEERCONNECTIONTESTWRAPPER_H_ |
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