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/*
* 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
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