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76 lines
3.2 KiB
C++
76 lines
3.2 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2019 Edouard Griffiths, F4EXB //
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// //
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// This program is free software; you can redistribute it and/or modify //
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// it under the terms of the GNU General Public License as published by //
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// the Free Software Foundation as version 3 of the License, or //
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// //
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// This program is distributed in the hope that it will be useful, //
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// but WITHOUT ANY WARRANTY; without even the implied warranty of //
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
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// GNU General Public License V3 for more details. //
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// //
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// You should have received a copy of the GNU General Public License //
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// along with this program. If not, see <http://www.gnu.org/licenses/>. //
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///////////////////////////////////////////////////////////////////////////////////
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#ifndef _SDRBASE_AUDIO_AUDIOFILTER_H_
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#define _SDRBASE_AUDIO_AUDIOFILTER_H_
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#include "export.h"
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#include "dsp/iirfilter.h"
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/**
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* By default this is a 2 pole lowpass Chebyshev (recursive) filter at fc=0.075 using coefficients found in table 20-1 of
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* http://www.analog.com/media/en/technical-documentation/dsp-book/dsp_book_Ch20.pdf
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*
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* At the interpolated sampling frequency of 48 kHz the -3 dB corner is at 48 * .075 = 3.6 kHz which is perfect for voice
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*
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* a0= 3.869430E-02
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* a1= 7.738860E-02 b1= 1.392667E+00
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* a2= 3.869430E-02 b2= -5.474446E-01
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*
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* given x[n] is the new input sample and y[n] the returned output sample:
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*
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* y[n] = a0*x[n] + a1*x[n] + a2*x[n] + b1*y[n-1] + b2*y[n-2]
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*
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* This one works directly with floats
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*
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* It can be generalized using the program found in tables 20-4 and 20-5 of the same document. This form is used as a
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* decimation filter and can be set with the setDecimFilters method
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*/
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class SDRBASE_API AudioFilter {
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public:
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AudioFilter();
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~AudioFilter();
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void useHP(bool useHP) { m_useHP = useHP; }
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bool usesHP() const { return m_useHP; }
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void setDecimFilters(int srHigh, int srLow, float fcHigh, float fcLow, float gain = 1.0f);
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float run(const float& sample);
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float runHP(const float& sample);
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float runLP(const float& sample);
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private:
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void calculate2(bool highPass, double fc, float *a, float *b, float fgain); // two pole Chebyshev calculation
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void cheby(bool highPass, double fc, float pr, int np, double *a, double *b, float fgain);
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void cheby_sub(bool highPass, double fc, float pr, int np, int stage,
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double& a0, double& a1, double& a2, double& b1, double& b2);
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IIRFilter<float, 2> m_filterLP;
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IIRFilter<float, 2> m_filterHP;
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bool m_useHP;
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float m_lpva[3];
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float m_lpvb[3];
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float m_hpva[3];
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float m_hpvb[3];
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static const float m_lpa[3];
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static const float m_lpb[3];
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static const float m_hpa[3];
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static const float m_hpb[3];
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};
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#endif // _SDRBASE_AUDIO_AUDIOFILTER_H_
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