From ab7f5ac77121aaa5b7c5690ba47ca9abc0dff497 Mon Sep 17 00:00:00 2001
From: Edouard DUPIN <yui.heero@gmail.com>
Date: Mon, 4 May 2015 22:25:32 +0200
Subject: [PATCH] [DEV] update equalizer

---
 audio/drain/BiQuadFloat.cpp | 285 ------------------------------------
 audio/drain/BiQuadFloat.h   |  97 ------------
 audio/drain/Equalizer.cpp   |  84 +++--------
 audio/drain/Equalizer.h     |  11 +-
 lutin_audio_drain.py        |   3 +-
 5 files changed, 28 insertions(+), 452 deletions(-)
 delete mode 100644 audio/drain/BiQuadFloat.cpp
 delete mode 100644 audio/drain/BiQuadFloat.h

diff --git a/audio/drain/BiQuadFloat.cpp b/audio/drain/BiQuadFloat.cpp
deleted file mode 100644
index 97e4d3f..0000000
--- a/audio/drain/BiQuadFloat.cpp
+++ /dev/null
@@ -1,285 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- */
-
-#include <audio/drain/debug.h>
-#include <audio/drain/BiQuadFloat.h>
-#include <cmath>
-
-static const char* listValues[] = {
-	"none",
-	"low-pass",
-	"high-pass",
-	"band-pass",
-	"notch",
-	"peak",
-	"low-shelf",
-	"high-shelf"
-};
-static int32_t listValuesSize = sizeof(listValues)/sizeof(char*);
-
-
-namespace etk {
-	template<> std::string to_string<enum audio::drain::filterType>(const enum audio::drain::filterType& _variable) {
-		return listValues[_variable];
-	}
-	template <> bool from_string<enum audio::drain::filterType>(enum audio::drain::filterType& _variableRet, const std::string& _value) {
-		for (int32_t iii=0; iii<listValuesSize; ++iii) {
-			if (_value == listValues[iii]) {
-				_variableRet = static_cast<enum audio::drain::filterType>(iii);
-				return true;
-			}
-		}
-		_variableRet = audio::drain::filterType_none;
-		return false;
-	}
-}
-
-
-
-audio::drain::BiQuadFloat::BiQuadFloat() {
-	reset();
-	// reset coefficients
-	m_a[0] = 1.0;
-	m_a[1] = 0.0;
-	m_a[2] = 0.0;
-	m_b[0] = 0.0;
-	m_b[1] = 0.0;
-}
-
-
-float audio::drain::BiQuadFloat::processFloat(float _sample) {
-	float result;
-	// compute
-	result =   m_a[0] * _sample
-	         + m_a[1] * m_x[0]
-	         + m_a[2] * m_x[1]
-	         - m_b[0] * m_y[0]
-	         - m_b[1] * m_y[1];
-	//update history of X
-	m_x[1] = m_x[0];
-	m_x[0] = _sample;
-	//update history of Y
-	m_y[1] = m_y[0];
-	m_y[0] = result;
-	return result;
-}
-
-void audio::drain::BiQuadFloat::processFloat(float* _input,
-                                             float* _output,
-                                             size_t _nbChunk,
-                                             int32_t _inputOffset,
-                                             int32_t _outputOffset) {
-	for (size_t iii=0; iii<_nbChunk; ++iii) {
-		// process in float the biquad.
-		*_output = processFloat(*_input);
-		// move to the sample on the same channel.
-		_input += _inputOffset;
-		_output += _outputOffset;
-	}
-}
-
-void audio::drain::BiQuadFloat::processInt16(int16_t* _input,
-                                             int16_t* _output,
-                                             size_t _nbChunk,
-                                             int32_t _inputOffset,
-                                             int32_t _outputOffset) {
-	for (size_t iii=0; iii<_nbChunk; ++iii) {
-		// process in float the biquad.
-		float out = processFloat(*_input);
-		// Limit output.
-		out = std::avg(-32768.0f, out, 32767.0f);
-		*_output = static_cast<int16_t>(out);
-		// move to the sample on the same channel.
-		_input += _inputOffset;
-		_output += _outputOffset;
-	}
-}
-
-
-void audio::drain::BiQuadFloat::setBiquad(enum audio::drain::filterType _type, double _frequencyCut, double _qualityFactor, double _gain, float _sampleRate) {
-	reset();
-	if (_sampleRate < 1) {
-		m_a[0] = 1.0;
-		m_a[1] = 0.0;
-		m_a[2] = 0.0;
-		m_b[0] = 0.0;
-		m_b[1] = 0.0;
-		return;
-	}
-	if (_frequencyCut > _sampleRate/2) {
-		_frequencyCut = _sampleRate/2;
-	} else if (_frequencyCut < 0) {
-		_frequencyCut = 0;
-	}
-	if (_qualityFactor < 0.01) {
-		_qualityFactor = 0.01;
-	}
-	double norm;
-	double V = std11::pow(10.0, std11::abs(_gain) / 20.0);
-	double K = std11::tan(M_PI * _frequencyCut / _sampleRate);
-	switch (_type) {
-		case filterType_none:
-			m_a[0] = 1.0;
-			m_a[1] = 0.0;
-			m_a[2] = 0.0;
-			m_b[0] = 0.0;
-			m_b[1] = 0.0;
-			break;
-		case filterType_lowPass:
-			norm = 1 / (1 + K / _qualityFactor + K * K);
-			m_a[0] = K * K * norm;
-			m_a[1] = 2 * m_a[0];
-			m_a[2] = m_a[0];
-			m_b[0] = 2 * (K * K - 1) * norm;
-			m_b[1] = (1 - K / _qualityFactor + K * K) * norm;
-			break;
-		case filterType_highPass:
-			norm = 1 / (1 + K / _qualityFactor + K * K);
-			m_a[0] = 1 * norm;
-			m_a[1] = -2 * m_a[0];
-			m_a[2] = m_a[0];
-			m_b[0] = 2 * (K * K - 1) * norm;
-			m_b[1] = (1 - K / _qualityFactor + K * K) * norm;
-			break;
-		case filterType_bandPass:
-			norm = 1 / (1 + K / _qualityFactor + K * K);
-			m_a[0] = K / _qualityFactor * norm;
-			m_a[1] = 0;
-			m_a[2] = -m_a[0];
-			m_b[0] = 2 * (K * K - 1) * norm;
-			m_b[1] = (1 - K / _qualityFactor + K * K) * norm;
-			break;
-		case filterType_notch:
-			norm = 1 / (1 + K / _qualityFactor + K * K);
-			m_a[0] = (1 + K * K) * norm;
-			m_a[1] = 2 * (K * K - 1) * norm;
-			m_a[2] = m_a[0];
-			m_b[0] = m_a[1];
-			m_b[1] = (1 - K / _qualityFactor + K * K) * norm;
-			break;
-		case filterType_peak:
-			if (_gain >= 0) {
-				norm = 1 / (1 + 1/_qualityFactor * K + K * K);
-				m_a[0] = (1 + V/_qualityFactor * K + K * K) * norm;
-				m_a[1] = 2 * (K * K - 1) * norm;
-				m_a[2] = (1 - V/_qualityFactor * K + K * K) * norm;
-				m_b[0] = m_a[1];
-				m_b[1] = (1 - 1/_qualityFactor * K + K * K) * norm;
-			} else {
-				norm = 1 / (1 + V/_qualityFactor * K + K * K);
-				m_a[0] = (1 + 1/_qualityFactor * K + K * K) * norm;
-				m_a[1] = 2 * (K * K - 1) * norm;
-				m_a[2] = (1 - 1/_qualityFactor * K + K * K) * norm;
-				m_b[0] = m_a[1];
-				m_b[1] = (1 - V/_qualityFactor * K + K * K) * norm;
-			}
-			break;
-		case filterType_lowShelf:
-			if (_gain >= 0) {
-				norm = 1 / (1 + M_SQRT2 * K + K * K);
-				m_a[0] = (1 + std11::sqrt(2*V) * K + V * K * K) * norm;
-				m_a[1] = 2 * (V * K * K - 1) * norm;
-				m_a[2] = (1 - std11::sqrt(2*V) * K + V * K * K) * norm;
-				m_b[0] = 2 * (K * K - 1) * norm;
-				m_b[1] = (1 - M_SQRT2 * K + K * K) * norm;
-			} else {
-				norm = 1 / (1 + std11::sqrt(2*V) * K + V * K * K);
-				m_a[0] = (1 + M_SQRT2 * K + K * K) * norm;
-				m_a[1] = 2 * (K * K - 1) * norm;
-				m_a[2] = (1 - M_SQRT2 * K + K * K) * norm;
-				m_b[0] = 2 * (V * K * K - 1) * norm;
-				m_b[1] = (1 - std11::sqrt(2*V) * K + V * K * K) * norm;
-			}
-			break;
-		case filterType_highShelf:
-			if (_gain >= 0) {
-				norm = 1 / (1 + M_SQRT2 * K + K * K);
-				m_a[0] = (V + std11::sqrt(2*V) * K + K * K) * norm;
-				m_a[1] = 2 * (K * K - V) * norm;
-				m_a[2] = (V - std11::sqrt(2*V) * K + K * K) * norm;
-				m_b[0] = 2 * (K * K - 1) * norm;
-				m_b[1] = (1 - M_SQRT2 * K + K * K) * norm;
-			} else {
-				norm = 1 / (V + std11::sqrt(2*V) * K + K * K);
-				m_a[0] = (1 + M_SQRT2 * K + K * K) * norm;
-				m_a[1] = 2 * (K * K - 1) * norm;
-				m_a[2] = (1 - M_SQRT2 * K + K * K) * norm;
-				m_b[0] = 2 * (K * K - V) * norm;
-				m_b[1] = (V - std11::sqrt(2*V) * K + K * K) * norm;
-			}
-			break;
-	}
-}
-
-void audio::drain::BiQuadFloat::reset() {
-	m_x[0] = 0;
-	m_y[1] = 0;
-	m_x[0] = 0;
-	m_y[1] = 0;
-}
-
-void audio::drain::BiQuadFloat::setBiquadCoef(float _a0, float _a1, float _a2, float _b0, float _b1) {
-	m_a[0] = _a0;
-	m_a[1] = _a1;
-	m_a[2] = _a2;
-	m_b[0] = _b0;
-	m_b[1] = _b1;
-	reset();
-}
-void audio::drain::BiQuadFloat::getBiquadCoef(float& _a0, float& _a1, float& _a2, float& _b0, float& _b1) {
-	_a0 = m_a[0];
-	_a1 = m_a[1];
-	_a2 = m_a[2];
-	_b0 = m_b[0];
-	_b1 = m_b[1];
-}
-
-std::vector<float> audio::drain::BiQuadFloat::getCoef() {
-	std::vector<float> out;
-	out.push_back(m_a[0]);
-	out.push_back(m_a[1]);
-	out.push_back(m_a[2]);
-	out.push_back(m_b[0]);
-	out.push_back(m_b[1]);
-	return out;
-}
-
-
-std::vector<std::pair<float,float> > audio::drain::BiQuadFloat::calculateTheory(double _sampleRate) {
-	std::vector<std::pair<float,float> > out;
-	double norm;
-	
-	bool buildLinear = true;
-	
-	size_t len = 512;
-	for (size_t iii=0; iii < len; iii++) {
-		double w;
-		if (buildLinear == true) {
-			// 0 to pi, linear scale
-			w = iii / (len - 1.0) * M_PI;
-		} else {
-			// 0.001 to 1, times pi, log scale
-			w = std11::exp(std11::log(1.0 / 0.001) * iii / (len - 1.0)) * 0.001 * M_PI;
-		}
-		double freq = iii / (len - 1.0) * _sampleRate / 2.0;
-		double phi = std11::pow(std11::sin(w/2.0), 2.0);
-		double y =   std11::log(   std11::pow(m_a[0]+m_a[1]+m_a[2], 2.0)
-		                         - 4.0*(m_a[0]*m_a[1] + 4.0*m_a[0]*m_a[2] + m_a[1]*m_a[2])*phi
-		                         + 16.0*m_a[0]*m_a[2]*phi*phi)
-		           - std11::log(   std11::pow(1.0+m_b[0]+m_b[1], 2.0)
-		                         - 4.0*(m_b[0] + 4.0*m_b[1] + m_b[0]*m_b[1])*phi
-		                         + 16.0*m_b[1]*phi*phi);
-		y = y * 10.0 / M_LN10;
-		
-		if (y <= -200) {
-			y = -200.0;
-		}
-		
-		//APPL_DEBUG("theory = " << freq << " power=" << y);
-		out.push_back(std::make_pair<float,float>(freq, y));
-	}
-	return out;
-}
diff --git a/audio/drain/BiQuadFloat.h b/audio/drain/BiQuadFloat.h
deleted file mode 100644
index 7078898..0000000
--- a/audio/drain/BiQuadFloat.h
+++ /dev/null
@@ -1,97 +0,0 @@
-/** @file
- * @author Edouard DUPIN 
- * @copyright 2011, Edouard DUPIN, all right reserved
- * @license APACHE v2.0 (see license file)
- */
-
-#ifndef __AUDIO_DRAIN_ALGO_BIQUAD_FLOAT_H__
-#define __AUDIO_DRAIN_ALGO_BIQUAD_FLOAT_H__
-
-#include <audio/drain/Algo.h>
-#include <etk/memory.h>
-
-namespace audio {
-	namespace drain {
-		enum filterType {
-			filterType_none, //!< no filter (pass threw...)
-			filterType_lowPass, //!< low pass filter
-			filterType_highPass, //!< High pass filter
-			filterType_bandPass, //!< band pass filter
-			filterType_notch, //!< Notch Filter
-			filterType_peak, //!< Peaking band EQ filter
-			filterType_lowShelf, //!< Low shelf filter
-			filterType_highShelf, //!< High shelf filter
-		};
-		class BiQuadFloat {
-			public:
-				BiQuadFloat();
-			protected:
-				float m_x[2]; //!< X history
-				float m_y[2]; //!< Y histiry
-				float m_a[3]; //!< A bi-Quad coef
-				float m_b[2]; //!< B bi-Quad coef
-			public:
-				/**
-				 * @brief Set the bi-quad value and type
-				 * @param[in] _type Type of biquad.
-				 * @param[in] _frequencyCut Cut Frequency. [0..sampleRate/2]
-				 * @param[in] _qualityFactor Q factor of quality (good value of 0.707 ==> permit to not ower gain) limit [0.01 .. 10]
-				 * @param[in] _gain Gain to apply (for notch, peak, lowShelf and highShelf) limit : -30, +30
-				 * @param[in] _sampleRate Sample rate of the signal
-				 */
-				void setBiquad(enum audio::drain::filterType _type, double _frequencyCut, double _qualityFactor, double _gain, float _sampleRate);
-				/**
-				 * @brief Set direct Coefficients
-				 */
-				void setBiquadCoef(float _a0, float _a1, float _a2, float _b0, float _b1);
-				/**
-				 * @brief Get direct Coefficients
-				 */
-				void getBiquadCoef(float& _a0, float& _a1, float& _a2, float& _b0, float& _b1);
-				/**
-				 * @brief Get direct Coefficients
-				 */
-				std::vector<float> getCoef();
-				/**
-				 * @brief Reset bequad filter (only history not value).
-				 */
-				void reset();
-			protected:
-				/**
-				 * @brief process single sample in float.
-				 * @param[in] _sample Sample to process
-				 * @return updataed value
-				 */
-				float processFloat(float _sample);
-			public:
-				/**
-				 * @brief Porcess function.
-				 * param[in] _input Pointer on the input data.
-				 * param[in,out] _output Poirter on the output data (can be the same as input (inplace availlable).
-				 * param[in] _nbChunk Number of qample to process.
-				 * param[in] _inputOffset Offset to add when read input data.
-				 * param[in] _outputOffset Offset to add when write output data.
-				 */
-				void processFloat(float* _input,
-				                  float* _output,
-				                  size_t _nbChunk,
-				                  int32_t _inputOffset,
-				                  int32_t _outputOffset);
-				//! @previous
-				void processInt16(int16_t* _input,
-				                  int16_t* _output,
-				                  size_t _nbChunk,
-				                  int32_t _inputOffset,
-				                  int32_t _outputOffset);
-				/**
-				 * @brief calculate respond of the filter:
-				 * @param[in] _sampleRate input qample rate
-				 * @retrun list of frequency/power in dB
-				 */
-				std::vector<std::pair<float,float> > calculateTheory(double _sampleRate);
-		};
-	}
-}
-
-#endif
-
diff --git a/audio/drain/Equalizer.cpp b/audio/drain/Equalizer.cpp
index 1543fa7..9355b59 100644
--- a/audio/drain/Equalizer.cpp
+++ b/audio/drain/Equalizer.cpp
@@ -19,6 +19,7 @@ void audio::drain::Equalizer::init() {
 	audio::drain::Algo::init();
 	audio::drain::Algo::m_type = "Equalizer";
 	m_supportedFormat.push_back(audio::format_int16);
+	m_supportedFormat.push_back(audio::format_int16);
 	configureBiQuad();
 }
 
@@ -34,9 +35,7 @@ audio::drain::Equalizer::~Equalizer() {
 
 void audio::drain::Equalizer::configurationChange() {
 	audio::drain::Algo::configurationChange();
-	if (m_biquads.size() != getOutputFormat().getMap().size()) {
-		configureBiQuad();
-	}
+	configureBiQuad();
 }
 
 bool audio::drain::Equalizer::process(audio::Time& _time,
@@ -49,27 +48,7 @@ bool audio::drain::Equalizer::process(audio::Time& _time,
 	if (_input == nullptr) {
 		return false;
 	}
-	if (getOutputFormat().getFormat() == audio::format_float) {
-		for (size_t jjj=0; jjj<getOutputFormat().getMap().size(); ++jjj) {
-			// get pointer on data:
-			float* data = static_cast<float*>(_input);
-			// move to sample offset:
-			data += jjj;
-			for (size_t iii=0; iii<m_biquads[jjj].size(); ++iii) {
-				m_biquads[jjj][iii].processFloat(data, data, _inputNbChunk, getInputFormat().getMap().size(), getOutputFormat().getMap().size() );
-			}
-		}
-	} else if (getOutputFormat().getFormat() == audio::format_int16) {
-		for (size_t jjj=0; jjj<getOutputFormat().getMap().size(); ++jjj) {
-			// get pointer on data:
-			int16_t* data = static_cast<int16_t*>(_input);
-			// move to sample offset:
-			data += jjj;
-			for (size_t iii=0; iii<m_biquads[jjj].size(); ++iii) {
-				m_biquads[jjj][iii].processInt16(data, data, _inputNbChunk, getInputFormat().getMap().size(), getOutputFormat().getMap().size() );
-			}
-		}
-	}
+	m_algo.process(_output, _input, _inputNbChunk);
 	return true;
 }
 
@@ -79,11 +58,7 @@ bool audio::drain::Equalizer::setParameter(const std::string& _parameter, const
 		m_config = ejson::Object::create(_value);
 		configureBiQuad();
 	} else if (_parameter == "reset") {
-		for (int32_t iii=0; iii<m_biquads.size(); ++iii) {
-			for (int32_t jjj=0; jjj<m_biquads[iii].size(); ++jjj) {
-				m_biquads[iii][jjj].reset();
-			}
-		}
+		// TODO : m_algo.reset();
 		return true;
 	}
 	return false;
@@ -97,8 +72,7 @@ std::string audio::drain::Equalizer::getParameterProperty(const std::string& _pa
 	return "error";
 }
 
-static audio::drain::BiQuadFloat getBiquad(const std11::shared_ptr<const ejson::Object>& _object, float _frequency) {
-	audio::drain::BiQuadFloat out;
+void audio::drain::Equalizer::addBiquad(int32_t _idBiquad, const std11::shared_ptr<const ejson::Object>& _object) {
 	// get type:
 	std::string typeString = _object->getStringValue("type", "none");
 	if (typeString == "direct-value") {
@@ -107,23 +81,31 @@ static audio::drain::BiQuadFloat getBiquad(const std11::shared_ptr<const ejson::
 		double a2 = _object->getNumberValue("a2", 0.0);
 		double b0 = _object->getNumberValue("b0", 0.0);
 		double b1 = _object->getNumberValue("b1", 0.0);
-		out.setBiquadCoef(a0, a1, a2, b0, b1);
+		if (_idBiquad < 0) {
+			m_algo.addBiquad(a0, a1, a2, b0, b1);
+		} else {
+			m_algo.addBiquad(_idBiquad, a0, a1, a2, b0, b1);
+		}
 	} else {
-		enum audio::drain::filterType type;
+		enum audio::algo::drain::biQuadType type;
 		if (etk::from_string(type, typeString) == false) {
 			DRAIN_ERROR("Can not parse equalizer type:'" << typeString << "'");
 		}
 		double gain = _object->getNumberValue("gain", 0.0);
 		double frequency = _object->getNumberValue("cut-frequency", 0.0);
 		double quality = _object->getNumberValue("quality", 0.0);
-		out.setBiquad(type, frequency, quality, gain, _frequency);
+		if (_idBiquad < 0) {
+			m_algo.addBiquad(type, frequency, quality, gain);
+		} else {
+			m_algo.addBiquad(_idBiquad, type, frequency, quality, gain);
+		}
 	}
-	return out;
 }
 
 void audio::drain::Equalizer::configureBiQuad() {
-	m_biquads.clear();
-	m_biquads.resize(getOutputFormat().getMap().size());
+	m_algo.init(getOutputFormat().getFrequency(),
+	            getOutputFormat().getMap().size(),
+	            getOutputFormat().getFormat());
 	if (m_config == nullptr) {
 		return;
 	}
@@ -137,19 +119,14 @@ void audio::drain::Equalizer::configureBiQuad() {
 				DRAIN_ERROR("Parse the configuration error : not a correct parameter:" << kkk);
 				continue;
 			}
-			// declare biquad:
-			audio::drain::BiQuadFloat biquad = getBiquad(tmpObject, getOutputFormat().getFrequency());
-			// add this bequad for every Channel:
-			for (size_t iii=0; iii<m_biquads.size(); ++iii) {
-				m_biquads[iii].push_back(biquad);
-			}
+			// Add biquad...
+			addBiquad(-1, tmpObject);
 		}
 		return;
 	}
 	for (size_t iii=0; iii<getOutputFormat().getMap().size(); ++iii) {
 		std::string channelName = etk::to_string(getOutputFormat().getMap()[iii]);
 		const std11::shared_ptr<const ejson::Array> channelConfig = m_config->getArray(channelName);
-		
 		if (channelConfig == nullptr) {
 			// no config ... not a problem ...
 			continue;
@@ -161,26 +138,13 @@ void audio::drain::Equalizer::configureBiQuad() {
 				DRAIN_ERROR("Parse the configuration error : not a correct parameter:" << kkk);
 				continue;
 			}
-			// declare biquad:
-			audio::drain::BiQuadFloat biquad = getBiquad(tmpObject, getOutputFormat().getFrequency());
-			// add this bequad for specific channel:
-			m_biquads[iii].push_back(biquad);
+			// add biquad:
+			addBiquad(kkk, tmpObject);
 		}
 	}
 	return;
 }
 
 std::vector<std::pair<float,float> > audio::drain::Equalizer::calculateTheory() {
-	std::vector<std::pair<float,float> > out;
-	for (size_t iii=0; iii<m_biquads[0].size(); ++iii) {
-		if (iii == 0) {
-			out = m_biquads[0][iii].calculateTheory(getOutputFormat().getFrequency());
-		} else {
-			std::vector<std::pair<float,float> > tmp = m_biquads[0][iii].calculateTheory(getOutputFormat().getFrequency());
-			for (size_t jjj=0; jjj< out.size(); ++jjj) {
-				out[jjj].second += tmp[jjj].second;
-			}
-		}
-	}
-	return out;
+	return m_algo.calculateTheory();
 }
\ No newline at end of file
diff --git a/audio/drain/Equalizer.h b/audio/drain/Equalizer.h
index ade7b98..7158c2b 100644
--- a/audio/drain/Equalizer.h
+++ b/audio/drain/Equalizer.h
@@ -10,7 +10,7 @@
 #include <audio/drain/Algo.h>
 #include <etk/memory.h>
 #include <ejson/Object.h>
-#include <audio/drain/BiQuadFloat.h>
+#include <audio/algo/drain/Equalizer.h>
 
 namespace audio {
 	namespace drain {
@@ -41,14 +41,9 @@ namespace audio {
 				virtual bool setParameter(const std::string& _parameter, const std::string& _value);
 				virtual std::string getParameter(const std::string& _parameter) const;
 				virtual std::string getParameterProperty(const std::string& _parameter) const;
-				
 			protected:
-				/**
-				 * @brief repesent all the biquad to process:
-				 * The first vector represent the number of channel to process
-				 * The second vector represent the number of biquad to process
-				 */
-				std::vector<std::vector<BiQuadFloat> > m_biquads;
+				void addBiquad(int32_t _idBiquad, const std11::shared_ptr<const ejson::Object>& _object);
+				audio::algo::drain::Equalizer m_algo;
 				/**
 				 * @brief Configure biquad with the  user spec.
 				 */
diff --git a/lutin_audio_drain.py b/lutin_audio_drain.py
index 34d6cdd..fa4169c 100644
--- a/lutin_audio_drain.py
+++ b/lutin_audio_drain.py
@@ -14,7 +14,6 @@ def create(target):
 		'audio/drain/debug.cpp',
 		'audio/drain/airtalgo.cpp',
 		'audio/drain/Algo.cpp',
-		'audio/drain/BiQuadFloat.cpp',
 		'audio/drain/ChannelReorder.cpp',
 		'audio/drain/CircularBuffer.cpp',
 		'audio/drain/EndPointCallback.cpp',
@@ -32,7 +31,7 @@ def create(target):
 	
 	# TODO: myModule.add_optional_module_depend('speexdsp', "HAVE_SPEEX_DSP_RESAMPLE")
 	myModule.compile_flags_CC("-DHAVE_SPEEX_DSP_RESAMPLE")
-	myModule.add_module_depend(['etk', 'audio', 'ejson', 'speexdsp'])
+	myModule.add_module_depend(['etk', 'audio', 'ejson', 'speexdsp', 'audio_algo_drain'])
 	myModule.add_export_path(tools.get_current_path(__file__))
 	
 	# add the currrent module at the