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[DEV] NLMS that not work so good...

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This commit is contained in:
Edouard DUPIN 2015-04-02 23:13:36 +02:00
parent e08d0888a8
commit b3ddc41dae
12 changed files with 404 additions and 41 deletions
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26
audio/algo/aec/Lms.cpp
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@ -1,11 +1,16 @@
/** @file
* @author Edouard DUPIN
* @author Edouard DUPIN
* @author Fatima MARFOUQ
* @copyright 2011, Edouard DUPIN, all right reserved
* @license APACHE v2.0 (see license file)
*/
#include <audio/algo/aec/debug.h>
#include <audio/algo/aec/Lms.h>
#include <audio/algo/aec/updateFilter.h>
#include <audio/algo/aec/convolution.h>
audio::algo::aec::Lms::Lms(void) :
m_filter(),
@ -50,27 +55,12 @@ bool audio::algo::aec::Lms::process(float* _output, const float* _feedback, cons
return true;
}
static float convolution(float* _dataMinus, float* _dataPlus, size_t _count) {
float out = 0.0f;
for (size_t iii = 0; iii < _count; ++iii) {
out += *_dataMinus-- * *_dataPlus++;
}
return out;
}
static void updateFilter(float* _filter, float* _data, float _value, int32_t _count) {
for (size_t iii = 0; iii < _count; ++iii) {
*(_filter++) += *_data-- * _value;
}
}
float audio::algo::aec::Lms::processValue(float* _feedback, float _microphone) {
// Error calculation.
float convolutionValue = convolution(_feedback, &m_filter[0], m_filter.size());
float convolutionValue = audio::algo::aec::convolution(_feedback, &m_filter[0], m_filter.size());
float error = _microphone - convolutionValue;
float out = std::avg(-1.0f, error, 1.0f);
updateFilter(&m_filter[0], _feedback, error*m_mu, m_filter.size());
audio::algo::aec::updateFilter(&m_filter[0], _feedback, error*m_mu, m_filter.size());
return out;
}

13
audio/algo/aec/Lms.h
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@ -1,11 +1,12 @@
/** @file
* @author Edouard DUPIN
* @author Edouard DUPIN
* @author Fatima MARFOUQ
* @copyright 2011, Edouard DUPIN, all right reserved
* @license APACHE v2.0 (see license file)
*/
#ifndef __DRAIN_LMS_H__
#define __DRAIN_LMS_H__
#ifndef __AUDIO_ALGO_AEC_LMS_H__
#define __AUDIO_ALGO_AEC_LMS_H__
#include <etk/types.h>
#include <etk/chrono.h>
@ -44,8 +45,8 @@ namespace audio {
d(n) |
|
o--------o | o-------------o
o---> filter --------->| | o--->| |
| û(n) | convol-| | d(n) - y(n) |----> e(n) -------> echo-cancelled output
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 |
@ -54,7 +55,7 @@ namespace audio {
| | | | |
| o-------->| |<-------o
| | û(n+1) = û(n) |
| | + 2 * mu * e(n) * x(n) |
| | + mu * e(n) * x(n) |
| | |
| o----------------------------------o
| |

85
audio/algo/aec/Nlms.cpp Normal file
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@ -0,0 +1,85 @@
/** @file
* @author Edouard DUPIN
* @author Fatima MARFOUQ
* @copyright 2011, Edouard DUPIN, all right reserved
* @license APACHE v2.0 (see license file)
*/
#include <audio/algo/aec/debug.h>
#include <audio/algo/aec/Nlms.h>
#include <audio/algo/aec/updateFilter.h>
#include <audio/algo/aec/convolution.h>
#include <audio/algo/aec/power.h>
audio::algo::aec::Nlms::Nlms(void) :
m_filter(),
m_feedBack() {
setFilterSize(256);
}
audio::algo::aec::Nlms::~Nlms(void) {
}
void audio::algo::aec::Nlms::reset(void) {
// simply reset filters.
setFilterSize(m_filter.size());
}
bool audio::algo::aec::Nlms::process(int16_t* _output, const int16_t* _feedback, const int16_t* _microphone, int32_t _nbSample) {
float output[_nbSample];
float feedback[_nbSample];
float microphone[_nbSample];
for (size_t iii=0; iii<_nbSample; ++iii) {
microphone[iii] = float(_microphone[iii])/32767.0f;
feedback[iii] = float(_feedback[iii])/32767.0f;
}
bool ret = process(output, feedback, microphone, _nbSample);
for (size_t iii=0; iii<_nbSample; ++iii) {
_output[iii] = int16_t(float(output[iii])*32767.0f);
}
return ret;
}
bool audio::algo::aec::Nlms::process(float* _output, const float* _feedback, const float* _microphone, int32_t _nbSample) {
// add sample in the feedback history:
m_feedBack.resize(m_filter.size()+_nbSample, 0.0f);
memcpy(&m_feedBack[m_filter.size()], _feedback, _nbSample*sizeof(float));
for (int32_t iii=0; iii < _nbSample; iii++) {
_output[iii] = processValue(&m_feedBack[m_filter.size()+iii], _microphone[iii]);
}
// remove old value:
m_feedBack.erase(m_feedBack.begin(), m_feedBack.begin() + (m_feedBack.size()-m_filter.size()) );
return true;
}
float audio::algo::aec::Nlms::processValue(float* _feedback, float _microphone) {
// Error calculation.
float convolutionValue = audio::algo::aec::convolution(_feedback, &m_filter[0], m_filter.size());
float error = _microphone - convolutionValue;
float out = std::avg(-1.0f, error, 1.0f);
// calculate mu:
float mu = audio::algo::aec::power(_feedback, m_filter.size());
//mu = *_feedback * *_feedback;
//AA_AEC_WARNING("Mu =" << mu);
if (mu <= 1.5f) {
// Not enought power in output
mu = 0.0001; // arbitrary
} else {
mu = 1.0f/mu;
//AA_AEC_WARNING("Mu =" << mu);
}
audio::algo::aec::updateFilter(&m_filter[0], _feedback, error*mu, m_filter.size());
return out;
}
void audio::algo::aec::Nlms::setFilterSize(size_t _sampleRate, std11::chrono::microseconds _time) {
setFilterSize((_sampleRate*_time.count())/1000000LL);
}
void audio::algo::aec::Nlms::setFilterSize(size_t _nbSample) {
m_filter.clear();
m_feedBack.clear();
m_filter.resize(_nbSample, 0.0f);
m_feedBack.resize(_nbSample, 0.0f);
}

130
audio/algo/aec/Nlms.h Normal file
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@ -0,0 +1,130 @@
/** @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_NLMS_H__
#define __AUDIO_ALGO_AEC_NLMS_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 Nlms {
public:
/**
* @brief Constructor
*/
Nlms(void);
/**
* @brief Destructor
*/
~Nlms(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);
private:
std::vector<float> m_filter; //!< Current filter
std::vector<float> m_feedBack; //!< Feedback history
public:
// for debug only:
std::vector<float> getFilter() {
return m_filter;
}
};
}
}
}
#endif

17
audio/algo/aec/convolution.cpp Normal file
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@ -0,0 +1,17 @@
/** @file
* @author Edouard DUPIN
* @author Fatima MARFOUQ
* @copyright 2011, Edouard DUPIN, all right reserved
* @license APACHE v2.0 (see license file)
*/
#include <audio/algo/aec/convolution.h>
float audio::algo::aec::convolution(float* _dataMinus, float* _dataPlus, size_t _count) {
float out = 0.0f;
for (size_t iii = 0; iii < _count; ++iii) {
out += *_dataMinus-- * *_dataPlus++;
}
return out;
}

21
audio/algo/aec/convolution.h Normal file
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@ -0,0 +1,21 @@
/** @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_CONVOLUTION_H__
#define __AUDIO_ALGO_AEC_CONVOLUTION_H__
#include <etk/types.h>
namespace audio {
namespace algo {
namespace aec {
float convolution(float* _dataMinus, float* _dataPlus, size_t _count);
}
}
}
#endif

19
audio/algo/aec/power.cpp Normal file
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@ -0,0 +1,19 @@
/** @file
* @author Edouard DUPIN
* @author Fatima MARFOUQ
* @copyright 2011, Edouard DUPIN, all right reserved
* @license APACHE v2.0 (see license file)
*/
#include <audio/algo/aec/power.h>
float audio::algo::aec::power(float* _data, int32_t _count) {
float out = 0;
for (size_t iii = 0; iii < _count; ++iii) {
out += *_data * *_data;
_data--;
}
return out;
}

21
audio/algo/aec/power.h Normal file
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@ -0,0 +1,21 @@
/** @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_POWER_H__
#define __AUDIO_ALGO_AEC_POWER_H__
#include <etk/types.h>
namespace audio {
namespace algo {
namespace aec {
float power(float* _data, int32_t _count);
}
}
}
#endif

16
audio/algo/aec/updateFilter.cpp Normal file
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@ -0,0 +1,16 @@
/** @file
* @author Edouard DUPIN
* @author Fatima MARFOUQ
* @copyright 2011, Edouard DUPIN, all right reserved
* @license APACHE v2.0 (see license file)
*/
#include <audio/algo/aec/updateFilter.h>
void audio::algo::aec::updateFilter(float* _filter, float* _data, float _value, int32_t _count) {
for (size_t iii = 0; iii < _count; ++iii) {
*(_filter++) += *_data-- * _value;
}
}

21
audio/algo/aec/updateFilter.h Normal file
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@ -0,0 +1,21 @@
/** @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_UPDATE_FILTER_H__
#define __AUDIO_ALGO_AEC_UPDATE_FILTER_H__
#include <etk/types.h>
namespace audio {
namespace algo {
namespace aec {
void updateFilter(float* _filter, float* _data, float _value, int32_t _count);
}
}
}
#endif

6
lutin_audio_algo_aec.py
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@ -11,7 +11,11 @@ def create(target):
myModule = module.Module(__file__, 'audio_algo_aec', 'LIBRARY')
myModule.add_src_file([
'audio/algo/aec/debug.cpp',
'audio/algo/aec/Lms.cpp'
'audio/algo/aec/convolution.cpp',
'audio/algo/aec/power.cpp',
'audio/algo/aec/updateFilter.cpp',
'audio/algo/aec/Lms.cpp',
'audio/algo/aec/Nlms.cpp'
])
myModule.add_module_depend(['etk'])
myModule.add_export_path(tools.get_current_path(__file__))

70
test/main.cpp
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@ -7,6 +7,7 @@
#include <test/debug.h>
#include <etk/etk.h>
#include <audio/algo/aec/Lms.h>
#include <audio/algo/aec/Nlms.h>
#include <etk/os/FSNode.h>
@ -56,6 +57,7 @@ int main(int _argc, const char** _argv) {
std::string micName = "";
int32_t filterSize = 0;
float mu = 0.0f;
bool nlms = false;
for (int32_t iii=0; iii<_argc ; ++iii) {
std::string data = _argv[iii];
if (etk::start_with(data,"--fb=")) {
@ -68,14 +70,17 @@ int main(int _argc, const char** _argv) {
} else if (etk::start_with(data,"--mu=")) {
data = &data[5];
mu = etk::string_to_float(data);
} else if (data == "--nlms") {
nlms = true;
} else if ( data == "-h"
|| data == "--help") {
APPL_INFO("Help : ");
APPL_INFO(" ./xxx --fb=file.raw --mic=file.raw");
APPL_INFO(" --fb Feedback file");
APPL_INFO(" --mic Microphone file");
APPL_INFO(" --filter-size Size of the filter");
APPL_INFO(" --mu Mu value -1.0< mu < -1.0");
APPL_INFO(" --fb=YYY.raw Feedback file");
APPL_INFO(" --mic=XXX.raw Microphone file");
APPL_INFO(" --filter-size=xxx Size of the filter");
APPL_INFO(" --mu=0.xx Mu value -1.0< mu < -1.0");
APPL_INFO(" --nlms NLMS version");
exit(0);
}
}
@ -90,24 +95,57 @@ int main(int _argc, const char** _argv) {
APPL_INFO("Read Microphone:");
std::vector<int16_t> micData = read(micName);
APPL_INFO(" " << micData.size() << " samples");
audio::algo::aec::Lms algo;
if (filterSize != 0) {
algo.setFilterSize(filterSize);
}
if (mu != 0.0f) {
algo.setMu(mu);
}
// resize output :
std::vector<int16_t> output;
output.resize(std::min(fbData.size(), micData.size()), 0);
// process in chunk of 256 samples
int32_t blockSize = 256;
for (int32_t iii=0; iii<output.size()/blockSize; ++iii) {
APPL_INFO("Process : " << iii*blockSize << "/" << int32_t(output.size()/blockSize)*blockSize);
algo.process(&output[iii*blockSize], &fbData[iii*blockSize], &micData[iii*blockSize], blockSize);
// end filter :
std::vector<float> filter;
if (nlms == false) {
APPL_INFO("***********************");
APPL_INFO("** LMS **");
APPL_INFO("***********************");
audio::algo::aec::Lms algo;
if (filterSize != 0) {
algo.setFilterSize(filterSize);
}
if (mu != 0.0f) {
algo.setMu(mu);
}
int32_t lastPourcent = -1;
for (int32_t iii=0; iii<output.size()/blockSize; ++iii) {
if (lastPourcent != 100*iii / (output.size()/blockSize)) {
lastPourcent = 100*iii / (output.size()/blockSize);
APPL_INFO("Process : " << iii*blockSize << "/" << int32_t(output.size()/blockSize)*blockSize << " " << lastPourcent << "/100");
} else {
APPL_VERBOSE("Process : " << iii*blockSize << "/" << int32_t(output.size()/blockSize)*blockSize);
}
algo.process(&output[iii*blockSize], &fbData[iii*blockSize], &micData[iii*blockSize], blockSize);
}
filter = algo.getFilter();
} else {
APPL_INFO("***********************");
APPL_INFO("** NLMS (power) **");
APPL_INFO("***********************");
audio::algo::aec::Nlms algo;
if (filterSize != 0) {
algo.setFilterSize(filterSize);
}
int32_t lastPourcent = -1;
for (int32_t iii=0; iii<output.size()/blockSize; ++iii) {
if (lastPourcent != 100*iii / (output.size()/blockSize)) {
lastPourcent = 100*iii / (output.size()/blockSize);
APPL_INFO("Process : " << iii*blockSize << "/" << int32_t(output.size()/blockSize)*blockSize << " " << lastPourcent << "/100");
} else {
APPL_VERBOSE("Process : " << iii*blockSize << "/" << int32_t(output.size()/blockSize)*blockSize);
}
algo.process(&output[iii*blockSize], &fbData[iii*blockSize], &micData[iii*blockSize], blockSize);
}
filter = algo.getFilter();
}
write("output.raw", output);
write("filter.raw", algo.getFilter());
write("filter.raw", filter);
}