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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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 #include <list> | 11 #include <list> |
| 12 #include <memory> |
12 | 13 |
13 #include "testing/gtest/include/gtest/gtest.h" | 14 #include "testing/gtest/include/gtest/gtest.h" |
14 | 15 |
15 #include "webrtc/audio_state.h" | 16 #include "webrtc/audio_state.h" |
16 #include "webrtc/call.h" | 17 #include "webrtc/call.h" |
17 #include "webrtc/test/mock_voice_engine.h" | 18 #include "webrtc/test/mock_voice_engine.h" |
18 | 19 |
19 namespace { | 20 namespace { |
20 | 21 |
21 struct CallHelper { | 22 struct CallHelper { |
22 CallHelper() { | 23 CallHelper() { |
23 webrtc::AudioState::Config audio_state_config; | 24 webrtc::AudioState::Config audio_state_config; |
24 audio_state_config.voice_engine = &voice_engine_; | 25 audio_state_config.voice_engine = &voice_engine_; |
25 webrtc::Call::Config config; | 26 webrtc::Call::Config config; |
26 config.audio_state = webrtc::AudioState::Create(audio_state_config); | 27 config.audio_state = webrtc::AudioState::Create(audio_state_config); |
27 call_.reset(webrtc::Call::Create(config)); | 28 call_.reset(webrtc::Call::Create(config)); |
28 } | 29 } |
29 | 30 |
30 webrtc::Call* operator->() { return call_.get(); } | 31 webrtc::Call* operator->() { return call_.get(); } |
31 | 32 |
32 private: | 33 private: |
33 testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_; | 34 testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_; |
34 rtc::scoped_ptr<webrtc::Call> call_; | 35 std::unique_ptr<webrtc::Call> call_; |
35 }; | 36 }; |
36 } // namespace | 37 } // namespace |
37 | 38 |
38 namespace webrtc { | 39 namespace webrtc { |
39 | 40 |
40 TEST(CallTest, ConstructDestruct) { | 41 TEST(CallTest, ConstructDestruct) { |
41 CallHelper call; | 42 CallHelper call; |
42 } | 43 } |
43 | 44 |
44 TEST(CallTest, CreateDestroy_AudioSendStream) { | 45 TEST(CallTest, CreateDestroy_AudioSendStream) { |
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100 streams.push_front(stream); | 101 streams.push_front(stream); |
101 } | 102 } |
102 } | 103 } |
103 for (auto s : streams) { | 104 for (auto s : streams) { |
104 call->DestroyAudioReceiveStream(s); | 105 call->DestroyAudioReceiveStream(s); |
105 } | 106 } |
106 streams.clear(); | 107 streams.clear(); |
107 } | 108 } |
108 } | 109 } |
109 } // namespace webrtc | 110 } // namespace webrtc |
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