xref: /external/chromium_org/third_party/webrtc/modules/audio_device/test/func_test_manager.h

/*
 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H
#define WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H

#include "webrtc/modules/audio_device/audio_device_utility.h"

#include <list>
#include <string>

#include "webrtc/common_audio/resampler/include/resampler.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/modules/audio_device/test/audio_device_test_defines.h"
#include "webrtc/system_wrappers/interface/file_wrapper.h"
#include "webrtc/typedefs.h"

#if defined(WEBRTC_IOS) || defined(ANDROID)
#define USE_SLEEP_AS_PAUSE
#else
//#define USE_SLEEP_AS_PAUSE
#endif

// Sets the default pause time if using sleep as pause
#define DEFAULT_PAUSE_TIME 5000

#if defined(USE_SLEEP_AS_PAUSE)
#define PAUSE(a) SleepMs(a);
#else
#define PAUSE(a) AudioDeviceUtility::WaitForKey();
#endif

#define ADM_AUDIO_LAYER AudioDeviceModule::kPlatformDefaultAudio
//#define ADM_AUDIO_LAYER AudioDeviceModule::kLinuxPulseAudio

enum TestType
{
    TTInvalid = -1,
    TTAll = 0,
    TTAudioLayerSelection = 1,
    TTDeviceEnumeration = 2,
    TTDeviceSelection = 3,
    TTAudioTransport = 4,
    TTSpeakerVolume = 5,
    TTMicrophoneVolume = 6,
    TTSpeakerMute = 7,
    TTMicrophoneMute = 8,
    TTMicrophoneBoost = 9,
    TTMicrophoneAGC = 10,
    TTLoopback = 11,
    TTDeviceRemoval = 13,
    TTMobileAPI = 14,
    TTTest = 66,
};

struct AudioPacket
{
    uint8_t dataBuffer[4 * 960];
    uint16_t nSamples;
    uint16_t nBytesPerSample;
    uint8_t nChannels;
    uint32_t samplesPerSec;
};

class ProcessThread;

namespace webrtc
{

class AudioDeviceModule;
class AudioEventObserver;
class AudioTransport;

// ----------------------------------------------------------------------------
//  AudioEventObserver
// ----------------------------------------------------------------------------

class AudioEventObserver: public AudioDeviceObserver
{
public:
    virtual void OnErrorIsReported(const ErrorCode error);
    virtual void OnWarningIsReported(const WarningCode warning);
    AudioEventObserver(AudioDeviceModule* audioDevice);
    ~AudioEventObserver();
public:
    ErrorCode _error;
    WarningCode _warning;
};

// ----------------------------------------------------------------------------
//  AudioTransport
// ----------------------------------------------------------------------------

class AudioTransportImpl: public AudioTransport
{
public:
    virtual int32_t
        RecordedDataIsAvailable(const void* audioSamples,
                                const uint32_t nSamples,
                                const uint8_t nBytesPerSample,
                                const uint8_t nChannels,
                                const uint32_t samplesPerSec,
                                const uint32_t totalDelayMS,
                                const int32_t clockDrift,
                                const uint32_t currentMicLevel,
                                const bool keyPressed,
                                uint32_t& newMicLevel);

    virtual int32_t NeedMorePlayData(const uint32_t nSamples,
                                     const uint8_t nBytesPerSample,
                                     const uint8_t nChannels,
                                     const uint32_t samplesPerSec,
                                     void* audioSamples,
                                     uint32_t& nSamplesOut,
                                     int64_t* elapsed_time_ms,
                                     int64_t* ntp_time_ms);

    virtual int OnDataAvailable(const int voe_channels[],
                                int number_of_voe_channels,
                                const int16_t* audio_data,
                                int sample_rate,
                                int number_of_channels,
                                int number_of_frames,
                                int audio_delay_milliseconds,
                                int current_volume,
                                bool key_pressed,
                                bool need_audio_processing);

    virtual void PushCaptureData(int voe_channel, const void* audio_data,
                                 int bits_per_sample, int sample_rate,
                                 int number_of_channels,
                                 int number_of_frames);

    virtual void PullRenderData(int bits_per_sample, int sample_rate,
                                int number_of_channels, int number_of_frames,
                                void* audio_data,
                                int64_t* elapsed_time_ms,
                                int64_t* ntp_time_ms);

    AudioTransportImpl(AudioDeviceModule* audioDevice);
    ~AudioTransportImpl();

public:
    int32_t SetFilePlayout(bool enable, const char* fileName = NULL);
    void SetFullDuplex(bool enable);
    void SetSpeakerVolume(bool enable)
    {
        _speakerVolume = enable;
    }
    ;
    void SetSpeakerMute(bool enable)
    {
        _speakerMute = enable;
    }
    ;
    void SetMicrophoneMute(bool enable)
    {
        _microphoneMute = enable;
    }
    ;
    void SetMicrophoneVolume(bool enable)
    {
        _microphoneVolume = enable;
    }
    ;
    void SetMicrophoneBoost(bool enable)
    {
        _microphoneBoost = enable;
    }
    ;
    void SetLoopbackMeasurements(bool enable)
    {
        _loopBackMeasurements = enable;
    }
    ;
    void SetMicrophoneAGC(bool enable)
    {
        _microphoneAGC = enable;
    }
    ;

private:
    typedef std::list<AudioPacket*> AudioPacketList;
    AudioDeviceModule* _audioDevice;

    bool _playFromFile;
    bool _fullDuplex;
    bool _speakerVolume;
    bool _speakerMute;
    bool _microphoneVolume;
    bool _microphoneMute;
    bool _microphoneBoost;
    bool _microphoneAGC;
    bool _loopBackMeasurements;

    FileWrapper& _playFile;

    uint32_t _recCount;
    uint32_t _playCount;
    AudioPacketList _audioList;

    Resampler _resampler;
};

// ----------------------------------------------------------------------------
//  FuncTestManager
// ----------------------------------------------------------------------------

class FuncTestManager
{
public:
    FuncTestManager();
    ~FuncTestManager();
    int32_t Init();
    int32_t Close();
    int32_t DoTest(const TestType testType);
private:
    int32_t TestAudioLayerSelection();
    int32_t TestDeviceEnumeration();
    int32_t TestDeviceSelection();
    int32_t TestAudioTransport();
    int32_t TestSpeakerVolume();
    int32_t TestMicrophoneVolume();
    int32_t TestSpeakerMute();
    int32_t TestMicrophoneMute();
    int32_t TestMicrophoneBoost();
    int32_t TestLoopback();
    int32_t TestDeviceRemoval();
    int32_t TestExtra();
    int32_t TestMicrophoneAGC();
    int32_t SelectPlayoutDevice();
    int32_t SelectRecordingDevice();
    int32_t TestAdvancedMBAPI();
private:
    // Paths to where the resource files to be used for this test are located.
    std::string _playoutFile48;
    std::string _playoutFile44;
    std::string _playoutFile16;
    std::string _playoutFile8;

    ProcessThread* _processThread;
    AudioDeviceModule* _audioDevice;
    AudioEventObserver* _audioEventObserver;
    AudioTransportImpl* _audioTransport;
};

}  // namespace webrtc

#endif  // #ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H