[DEV] create a basic LMS. not tested

drain/echoCanceller/Lms.cpp

@@ -0,0 +1,87 @@
+/** @file
+ * @author Edouard DUPIN 
+ * @copyright 2011, Edouard DUPIN, all right reserved
+ * @license APACHE v2.0 (see license file)
+ */
+
+#include <drain/debug.h>
+#include <drain/echoCanceller/Lms.h>
+
+drain::Lms::Lms(void) :
+  m_filtre(),
+  m_feedBack(),
+  m_micro(0.1f) {
+	setFilterSize(256);
+}
+
+drain::Lms::~Lms(void) {
+	
+}
+
+void drain::Lms::reset(void) {
+	// simply reset filters.
+	setFilterSize(m_filtre.size());
+}
+
+bool drain::Lms::process(int16_t* _output, int16_t* _feedback, 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 drain::Lms::process(float* _output, float* _feedback, float* _microphone, int32_t _nbSample) {
+	// add sample in the feedback history:
+	m_feedBack.resize(m_filtre.size(), 0.0f);
+	memcpy(&m_feedBack[m_filtre.size()], _feedback, _nbSample*sizeof(float));
+	for (int32_t iii=0; iii < _nbSample; iii++) {
+		_output[iii] = processValue(&m_feedBack[m_filtre.size()+iii], _microphone[iii]);
+	}
+	// remove old value:
+	m_feedBack.erase(m_feedBack.begin(), m_feedBack.begin() + (m_feedBack.size()-m_filtre.size()) );
+	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 drain::Lms::processValue(float* _feedback, float _microphone) {
+	// Error calculation.
+	float convolutionValue = convolution(_feedback, &m_filtre[0], m_filtre.size());
+	float error = _microphone - convolutionValue;
+	float out = std::avg(-1.0f, error, 1.0f);
+	updateFilter(&m_filtre[0], _feedback, 2.0f*m_micro, m_filtre.size());
+	return out;
+}
+
+void drain::Lms::setFilterSize(size_t _sampleRate, std11::chrono::microseconds _time) {
+	setFilterSize((_sampleRate*_time.count())/1000000LL);
+}
+
+void drain::Lms::setFilterSize(size_t _nbSample) {
+	m_filtre.clear();
+	m_feedBack.clear();
+	m_filtre.resize(_nbSample, 0.0f);
+	m_feedBack.resize(_nbSample, 0.0f);
+}
+

drain/echoCanceller/Lms.h

@@ -0,0 +1,36 @@
+/** @file
+ * @author Edouard DUPIN 
+ * @copyright 2011, Edouard DUPIN, all right reserved
+ * @license APACHE v2.0 (see license file)
+ */
+
+#ifndef __DRAIN_LMS_H__
+#define __DRAIN_LMS_H__
+
+#include <etk/types.h>
+#include <etk/chrono.h>
+
+namespace drain {
+	// Least Mean Square (LMS) algorithm "echo canceller"
+	class Lms {
+		public:
+			Lms(void);
+			~Lms(void);
+		public:
+			void reset(void);
+			bool process(int16_t* _output, int16_t* _feedback, int16_t* _microphone, int32_t _nbSample);
+			bool process(float* _output, float* _feedback, float* _microphone, int32_t _nbSample);
+		protected:
+			float processValue(float* _feedback, float _microphone);
+		private:
+			std::vector<float> m_filtre;
+			std::vector<float> m_feedBack;
+		public:
+			void setFilterSize(size_t _sampleRate, std11::chrono::microseconds _time);
+			void setFilterSize(size_t _nbSample);
+		protected:
+			float m_micro; //!< µ step size
+	};
+}
+
+#endif

lutin_drain.py

@@ -26,7 +26,8 @@ def create(target):
 		'drain/Volume.cpp',
 		'drain/IOFormatInterface.cpp',
 		'drain/AutoLogInOut.cpp',
-		'drain/Equalizer.cpp'
+		'drain/Equalizer.cpp',
+		'drain/echoCanceller/Lms.cpp'
 		])
 	
 	# TODO: myModule.add_optional_module_depend('speexdsp', "HAVE_SPEEX_DSP_RESAMPLE")