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Side by Side Diff: webrtc/api/test/peerconnectiontestwrapper.h

Issue 1930463002: Replace scoped_ptr with unique_ptr in webrtc/api/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 7 months ago
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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<> {
(...skipping 63 matching lines...) Expand 10 before | Expand all | Expand 10 after
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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