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Issue 2353033005: Refactoring: move ownership of RtcEventLog from Call to PeerConnection (Closed)
Patch Set: Moved the constructor Created 4 years, 2 months ago
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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 #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
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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
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