137 lines
7.8 KiB
C++

/** @file
* @author Edouard DUPIN
* @author Fatima MARFOUQ
* @copyright 2011, Edouard DUPIN, all right reserved
* @license APACHE v2.0 (see license file)
*/
#ifndef __AUDIO_ALGO_AEC_LMS_H__
#define __AUDIO_ALGO_AEC_LMS_H__
#include <etk/types.h>
#include <etk/chrono.h>
namespace audio {
namespace algo {
namespace aec {
/**
* @brief Least Mean Square (LMS) algorithm "echo canceller"
* base on publication: http://www.arpapress.com/Volumes/Vol7Issue1/IJRRAS_7_1_05.pdf
Electronic description:
/
o---o /|
_feedback | |/ |
>---------------->| | | >~~~~~~~~o
x(n) | |\ | |
o---o \| |
\ o--------0
| | Environement
| u(n) | transfert fonction
| |
o--------o
|
|
o---o ___ |
_microphone | |/ \ <~~~~~~o
<----------------<| | | <~~~~~~~~~~~~< Noise
d(n) | |\___/ <~~~~~~0
o---o |
|
o~~~~~< Usefull signal
s(n)
LMS Algorithm:
_microphone -----------------------------o
d(n) |
|
o--------o | o-------------o
o---> filter --------->| | o--->| | _output
| û(n) | convol-| | d(n) - y(n) |----> e(n) -------> echo-cancelled
| | -ution |----> y(n) ---->| | |
| _feedback -----o--->| | o-------------o |
| x(n) | o--------o |
| | |
| | o----------------------------------o |
| | | | |
| o-------->| |<-------o
| | û(n+1) = û(n) |
| | + mu * e(n) * x(n) |
| | |
| o----------------------------------o
| |
| |
o--------------------------------------------o
*/
class Lms {
public:
/**
* @brief Constructor
*/
Lms(void);
/**
* @brief Destructor
*/
~Lms(void);
public:
/**
* @brief Reset filter history and filter
*/
void reset(void);
/**
* @brief Process 16 bit LMS (input 16 bits)
* @param[in,out] _output output data of the LMS
* @param[in] _feedback Input feedback of the signal: x(n)
* @param[in] _microphone Input Microphone data: d(n)
*/
bool process(int16_t* _output, const int16_t* _feedback, const int16_t* _microphone, int32_t _nbSample);
/**
* @brief Process float LMS
* @param[in,out] _output output data of the LMS
* @param[in] _feedback Input feedback of the signal: x(n)
* @param[in] _microphone Input Microphone data: d(n)
*/
bool process(float* _output, const float* _feedback, const float* _microphone, int32_t _nbSample);
protected:
/**
* @brief Process a single value of the LMS
* @param[in] _feedback Pointer on the feedback data (with history and at the n(th) position
* @param[in] _microphone Microphone single sample [-1..1]
* @return New output value [-1..1]
*/
float processValue(float* _feedback, float _microphone);
public:
/**
* @brief Set filter size with specifing the filter temporal size and his samplerate
* @param[in] _sampleRate Current sample rate to apply filter
* @param[in] _time Time of the filter size
*/
void setFilterSize(size_t _sampleRate, std11::chrono::microseconds _time);
/**
* @brief Set filter size in number of sample
* @param[in] _nbSample Sample size of the filter
*/
void setFilterSize(size_t _nbSample);
/**
* @brief Set Mu value for basic LMS value
* @param[in] _val new mu value
*/
void setMu(float _val);
private:
std::vector<float> m_filter; //!< Current filter
std::vector<float> m_feedBack; //!< Feedback history
float m_mu; //!< mu step size
public:
// for debug only:
std::vector<float> getFilter() {
return m_filter;
}
};
}
}
}
#endif