220 lines
6.1 KiB
C
220 lines
6.1 KiB
C
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H
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#define WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H
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#include "../source/audio_device_utility.h"
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#include "typedefs.h"
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#include "audio_device.h"
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#include "audio_device_test_defines.h"
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#include "file_wrapper.h"
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#include "list_wrapper.h"
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#include "resampler.h"
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#if defined(MAC_IPHONE) || defined(ANDROID)
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#define USE_SLEEP_AS_PAUSE
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#else
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//#define USE_SLEEP_AS_PAUSE
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#endif
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// Sets the default pause time if using sleep as pause
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#define DEFAULT_PAUSE_TIME 5000
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#if defined(USE_SLEEP_AS_PAUSE)
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#define PAUSE(a) AudioDeviceUtility::Sleep(a);
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#else
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#define PAUSE(a) AudioDeviceUtility::WaitForKey();
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#endif
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#define SLEEP(a) AudioDeviceUtility::Sleep(a);
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#define ADM_AUDIO_LAYER AudioDeviceModule::kPlatformDefaultAudio
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//#define ADM_AUDIO_LAYER AudioDeviceModule::kLinuxPulseAudio
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enum TestType
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{
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TTInvalid = -1,
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TTAll = 0,
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TTAudioLayerSelection = 1,
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TTDeviceEnumeration = 2,
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TTDeviceSelection = 3,
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TTAudioTransport = 4,
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TTSpeakerVolume = 5,
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TTMicrophoneVolume = 6,
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TTSpeakerMute = 7,
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TTMicrophoneMute = 8,
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TTMicrophoneBoost = 9,
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TTMicrophoneAGC = 10,
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TTLoopback = 11,
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TTDeviceRemoval = 13,
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TTMobileAPI = 14,
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TTTest = 66,
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};
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class ProcessThread;
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namespace webrtc
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{
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class AudioDeviceModule;
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class AudioEventObserver;
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class AudioTransport;
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// ----------------------------------------------------------------------------
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// AudioEventObserver
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// ----------------------------------------------------------------------------
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class AudioEventObserver: public AudioDeviceObserver
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{
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public:
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virtual void OnErrorIsReported(const ErrorCode error);
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virtual void OnWarningIsReported(const WarningCode warning);
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AudioEventObserver(AudioDeviceModule* audioDevice);
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~AudioEventObserver();
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public:
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ErrorCode _error;
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WarningCode _warning;
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private:
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AudioDeviceModule* _audioDevice;
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};
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// ----------------------------------------------------------------------------
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// AudioTransport
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// ----------------------------------------------------------------------------
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class AudioTransportImpl: public AudioTransport
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{
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public:
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virtual WebRtc_Word32
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RecordedDataIsAvailable(const WebRtc_Word8* audioSamples,
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const WebRtc_UWord32 nSamples,
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const WebRtc_UWord8 nBytesPerSample,
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const WebRtc_UWord8 nChannels,
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const WebRtc_UWord32 samplesPerSec,
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const WebRtc_UWord32 totalDelayMS,
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const WebRtc_Word32 clockDrift,
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const WebRtc_UWord32 currentMicLevel,
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WebRtc_UWord32& newMicLevel);
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virtual WebRtc_Word32 NeedMorePlayData(const WebRtc_UWord32 nSamples,
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const WebRtc_UWord8 nBytesPerSample,
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const WebRtc_UWord8 nChannels,
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const WebRtc_UWord32 samplesPerSec,
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WebRtc_Word8* audioSamples,
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WebRtc_UWord32& nSamplesOut);
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AudioTransportImpl(AudioDeviceModule* audioDevice);
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~AudioTransportImpl();
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public:
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WebRtc_Word32 SetFilePlayout(bool enable, const WebRtc_Word8* fileName =
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NULL);
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void SetFullDuplex(bool enable);
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void SetSpeakerVolume(bool enable)
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{
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_speakerVolume = enable;
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}
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;
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void SetSpeakerMute(bool enable)
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{
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_speakerMute = enable;
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}
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;
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void SetMicrophoneMute(bool enable)
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{
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_microphoneMute = enable;
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}
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;
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void SetMicrophoneVolume(bool enable)
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{
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_microphoneVolume = enable;
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}
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;
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void SetMicrophoneBoost(bool enable)
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{
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_microphoneBoost = enable;
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}
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;
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void SetLoopbackMeasurements(bool enable)
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{
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_loopBackMeasurements = enable;
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}
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;
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void SetMicrophoneAGC(bool enable)
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{
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_microphoneAGC = enable;
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}
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;
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private:
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AudioDeviceModule* _audioDevice;
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bool _playFromFile;
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bool _fullDuplex;
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bool _speakerVolume;
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bool _speakerMute;
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bool _microphoneVolume;
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bool _microphoneMute;
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bool _microphoneBoost;
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bool _microphoneAGC;
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bool _loopBackMeasurements;
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FileWrapper& _playFile;
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WebRtc_UWord32 _recCount;
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WebRtc_UWord32 _playCount;
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ListWrapper _audioList;
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Resampler _resampler;
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};
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// ----------------------------------------------------------------------------
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// FuncTestManager
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// ----------------------------------------------------------------------------
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class FuncTestManager
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{
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public:
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FuncTestManager();
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~FuncTestManager();
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WebRtc_Word32 Init();
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WebRtc_Word32 Close();
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WebRtc_Word32 DoTest(const TestType testType);
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private:
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WebRtc_Word32 TestAudioLayerSelection();
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WebRtc_Word32 TestDeviceEnumeration();
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WebRtc_Word32 TestDeviceSelection();
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WebRtc_Word32 TestAudioTransport();
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WebRtc_Word32 TestSpeakerVolume();
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WebRtc_Word32 TestMicrophoneVolume();
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WebRtc_Word32 TestSpeakerMute();
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WebRtc_Word32 TestMicrophoneMute();
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WebRtc_Word32 TestMicrophoneBoost();
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WebRtc_Word32 TestLoopback();
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WebRtc_Word32 TestDeviceRemoval();
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WebRtc_Word32 TestExtra();
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WebRtc_Word32 TestMicrophoneAGC();
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WebRtc_Word32 SelectPlayoutDevice();
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WebRtc_Word32 SelectRecordingDevice();
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WebRtc_Word32 TestAdvancedMBAPI();
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private:
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ProcessThread* _processThread;
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AudioDeviceModule* _audioDevice;
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AudioEventObserver* _audioEventObserver;
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AudioTransportImpl* _audioTransport;
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};
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} // namespace webrtc
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#endif // #ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H
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