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sdrangel/sdrbase/dsp/filtermbe.h

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///////////////////////////////////////////////////////////////////////////////////
// Copyright (C) 2016 F4EXB //
// written by Edouard Griffiths //
// //
// This program is free software; you can redistribute it and/or modify //
// it under the terms of the GNU General Public License as published by //
// the Free Software Foundation as version 3 of the License, or //
// //
// This program is distributed in the hope that it will be useful, //
// but WITHOUT ANY WARRANTY; without even the implied warranty of //
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
// GNU General Public License V3 for more details. //
// //
// You should have received a copy of the GNU General Public License //
// along with this program. If not, see <http://www.gnu.org/licenses/>. //
///////////////////////////////////////////////////////////////////////////////////
#ifndef SDRBASE_DSP_FILTERMBE_H_
#define SDRBASE_DSP_FILTERMBE_H_
/**
* Uses the generic IIR filter internally
*
* Low pass / High pass:
*
* This is a 2 pole Chebyshev (recursive) filter using coefficients found in table 20-1 (low pass)
* or table 20-2 (high pass) of http://www.analog.com/media/en/technical-documentation/dsp-book/dsp_book_Ch20.pdf
*
* For low pass fc = 0.075
* For high oass fc = 0.01
*
* Convention taken here exchanges A and B coefficients as shown in this image:
* https://cdn.mikroe.com/ebooks/img/8/2016/02/digital-filter-design-chapter-03-image-2-9.gif
* So A applies to Y and B to X
*
* At the interpolated sampling frequency of 48 kHz the -3 dB corner is at 48 * .075 = 3.6 kHz which is perfect for voice
* The high pass has a 3 dB corner of 48 * 0.01 = 0.48 kHz
*
* Low pass:
*
* b0 = 3.869430E-02 (a0 = 1.0)
* b1 = 7.738860E-02 a1 = 1.392667E+00
* b2 = 3.869430E-02 a2 = -5.474446E-01
*
* High pass:
*
* b0 = 9.567529E-01 (a0 = 1.0)
* b1 = -1.913506E+00 a1 = 1.911437E+00
* b2 = 9.567529E-01 a2 = -9.155749E-01
*
* given x[n] is the new input sample and y[n] the returned output sample:
*
* y[n] = b0*x[n] + b1*x[n] + b2*x[n] + a1*y[n-1] + a2*y[n-2]
*
* This one works directly with floats
*
*
*/
#include "iirfilter.h"
#include "export.h"
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class SDRBASE_API MBEAudioInterpolatorFilter
{
public:
MBEAudioInterpolatorFilter();
~MBEAudioInterpolatorFilter();
void useHP(bool useHP) { m_useHP = useHP; }
bool usesHP() const { return m_useHP; }
float run(const float& sample);
float runHP(const float& sample);
float runLP(const float& sample);
private:
IIRFilter<float, 2> m_filterLP;
IIRFilter<float, 2> m_filterHP;
bool m_useHP;
static const float m_lpa[3], m_lpb[3]; // low pass coefficients
static const float m_hpa[3], m_hpb[3]; // band pass coefficients
};
#endif /* SDRBASE_DSP_FILTERMBE_H_ */