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Issue 1403363003: Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly i… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: that's all folks! (incl rebase), or is it? Created 5 years, 1 month 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 12
13 #include "testing/gtest/include/gtest/gtest.h" 13 #include "testing/gtest/include/gtest/gtest.h"
14 14
15 #include "webrtc/audio_state.h"
15 #include "webrtc/call.h" 16 #include "webrtc/call.h"
16 #include "webrtc/test/mock_voice_engine.h" 17 #include "webrtc/test/mock_voice_engine.h"
17 18
18 namespace { 19 namespace {
19 20
20 struct CallHelper { 21 struct CallHelper {
21 CallHelper() : voice_engine_(new webrtc::test::MockVoiceEngine()) { 22 CallHelper() {
23 EXPECT_CALL(voice_engine_,
24 RegisterVoiceEngineObserver(testing::_)).WillOnce(testing::Return(0));
25 EXPECT_CALL(voice_engine_,
26 DeRegisterVoiceEngineObserver()).WillOnce(testing::Return(0));
27 EXPECT_CALL(voice_engine_,
28 GetEventLog()).WillOnce(testing::Return(nullptr));
29 webrtc::AudioState::Config audio_state_config;
30 audio_state_config.voice_engine = &voice_engine_;
22 webrtc::Call::Config config; 31 webrtc::Call::Config config;
23 config.voice_engine = voice_engine_.get(); 32 config.audio_state = webrtc::AudioState::Create(audio_state_config);
24 call_.reset(webrtc::Call::Create(config)); 33 call_.reset(webrtc::Call::Create(config));
25 } 34 }
26 35
27 webrtc::Call* operator->() { return call_.get(); } 36 webrtc::Call* operator->() { return call_.get(); }
28 37
29 private: 38 private:
30 rtc::scoped_ptr<webrtc::test::MockVoiceEngine> voice_engine_; 39 webrtc::test::MockVoiceEngine voice_engine_;
31 rtc::scoped_ptr<webrtc::Call> call_; 40 rtc::scoped_ptr<webrtc::Call> call_;
32 }; 41 };
33 } // namespace 42 } // namespace
34 43
35 namespace webrtc { 44 namespace webrtc {
36 45
37 TEST(CallTest, ConstructDestruct) { 46 TEST(CallTest, ConstructDestruct) {
38 CallHelper call; 47 CallHelper call;
39 } 48 }
40 49
(...skipping 56 matching lines...) Expand 10 before | Expand all | Expand 10 after
97 streams.push_front(stream); 106 streams.push_front(stream);
98 } 107 }
99 } 108 }
100 for (auto s : streams) { 109 for (auto s : streams) {
101 call->DestroyAudioReceiveStream(s); 110 call->DestroyAudioReceiveStream(s);
102 } 111 }
103 streams.clear(); 112 streams.clear();
104 } 113 }
105 } 114 }
106 } // namespace webrtc 115 } // namespace webrtc
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