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1 // Copyright 2014 The Chromium Authors. All rights reserved. | 1 // Copyright 2014 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include "base/command_line.h" | 5 #include "base/command_line.h" |
6 #include "content/public/common/content_switches.h" | 6 #include "content/public/common/content_switches.h" |
7 #include "content/renderer/media/mock_media_constraint_factory.h" | 7 #include "content/renderer/media/mock_media_constraint_factory.h" |
8 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" | 8 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
9 #include "content/renderer/media/webrtc_local_audio_track.h" | 9 #include "content/renderer/media/webrtc_local_audio_track.h" |
10 #include "testing/gmock/include/gmock/gmock.h" | 10 #include "testing/gmock/include/gmock/gmock.h" |
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81 sink.reset(); | 81 sink.reset(); |
82 | 82 |
83 // Verify that no more callback gets into the sink. | 83 // Verify that no more callback gets into the sink. |
84 track_->Capture(data.get(), base::TimeDelta(), 255, false, false); | 84 track_->Capture(data.get(), base::TimeDelta(), 255, false, false); |
85 } | 85 } |
86 | 86 |
87 TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) { | 87 TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) { |
88 webrtc::AudioTrackInterface* webrtc_track = | 88 webrtc::AudioTrackInterface* webrtc_track = |
89 static_cast<webrtc::AudioTrackInterface*>(adapter_.get()); | 89 static_cast<webrtc::AudioTrackInterface*>(adapter_.get()); |
90 int signal_level = 0; | 90 int signal_level = 0; |
| 91 EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level)); |
| 92 |
| 93 // Disable the audio processing in the audio track. |
| 94 CommandLine::ForCurrentProcess()->AppendSwitch( |
| 95 switches::kDisableAudioTrackProcessing); |
91 EXPECT_FALSE(webrtc_track->GetSignalLevel(&signal_level)); | 96 EXPECT_FALSE(webrtc_track->GetSignalLevel(&signal_level)); |
92 | |
93 // Enable the audio processing in the audio track. | |
94 CommandLine::ForCurrentProcess()->AppendSwitch( | |
95 switches::kEnableAudioTrackProcessing); | |
96 EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level)); | |
97 } | 97 } |
98 | 98 |
99 } // namespace content | 99 } // namespace content |
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