| OLD | NEW |
| 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 #include <memory> |
| 13 | 13 |
| 14 #include "webrtc/api/call/audio_state.h" | 14 #include "webrtc/api/call/audio_state.h" |
| 15 #include "webrtc/call.h" | 15 #include "webrtc/call.h" |
| 16 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
| 16 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h" | 17 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h" |
| 17 #include "webrtc/test/gtest.h" | 18 #include "webrtc/test/gtest.h" |
| 18 #include "webrtc/test/mock_voice_engine.h" | 19 #include "webrtc/test/mock_voice_engine.h" |
| 19 | 20 |
| 20 namespace { | 21 namespace { |
| 21 | 22 |
| 22 struct CallHelper { | 23 struct CallHelper { |
| 23 explicit CallHelper( | 24 explicit CallHelper( |
| 24 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr) | 25 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr) |
| 25 : voice_engine_(decoder_factory) { | 26 : voice_engine_(decoder_factory) { |
| 26 webrtc::AudioState::Config audio_state_config; | 27 webrtc::AudioState::Config audio_state_config; |
| 27 audio_state_config.voice_engine = &voice_engine_; | 28 audio_state_config.voice_engine = &voice_engine_; |
| 28 webrtc::Call::Config config; | 29 webrtc::Call::Config config(&event_log_); |
| 29 config.audio_state = webrtc::AudioState::Create(audio_state_config); | 30 config.audio_state = webrtc::AudioState::Create(audio_state_config); |
| 30 call_.reset(webrtc::Call::Create(config)); | 31 call_.reset(webrtc::Call::Create(config)); |
| 31 } | 32 } |
| 32 | 33 |
| 33 webrtc::Call* operator->() { return call_.get(); } | 34 webrtc::Call* operator->() { return call_.get(); } |
| 34 | 35 |
| 35 private: | 36 private: |
| 36 testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_; | 37 testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_; |
| 38 webrtc::RtcEventLogNullImpl event_log_; |
| 37 std::unique_ptr<webrtc::Call> call_; | 39 std::unique_ptr<webrtc::Call> call_; |
| 38 }; | 40 }; |
| 39 } // namespace | 41 } // namespace |
| 40 | 42 |
| 41 namespace webrtc { | 43 namespace webrtc { |
| 42 | 44 |
| 43 TEST(CallTest, ConstructDestruct) { | 45 TEST(CallTest, ConstructDestruct) { |
| 44 CallHelper call; | 46 CallHelper call; |
| 45 } | 47 } |
| 46 | 48 |
| (...skipping 62 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 109 streams.push_front(stream); | 111 streams.push_front(stream); |
| 110 } | 112 } |
| 111 } | 113 } |
| 112 for (auto s : streams) { | 114 for (auto s : streams) { |
| 113 call->DestroyAudioReceiveStream(s); | 115 call->DestroyAudioReceiveStream(s); |
| 114 } | 116 } |
| 115 streams.clear(); | 117 streams.clear(); |
| 116 } | 118 } |
| 117 } | 119 } |
| 118 } // namespace webrtc | 120 } // namespace webrtc |
| OLD | NEW |