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https://github.com/f4exb/sdrangel.git
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209 lines
6.1 KiB
C++
209 lines
6.1 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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// (at your option) any later version. //
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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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#include <cmath>
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#include <algorithm>
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#include <QDebug>
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#include "audiofilter.h"
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// f(-3dB) = 3.6 kHz @ 48000 Hz SR (w = 0.0375):
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const float AudioFilter::m_lpa[3] = {1.0, 1.392667E+00, -5.474446E-01};
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const float AudioFilter::m_lpb[3] = {3.869430E-02, 7.738860E-02, 3.869430E-02};
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// f(-3dB) = 300 Hz @ 8000 Hz SR (w = 0.0375):
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const float AudioFilter::m_hpa[3] = {1.000000e+00, 1.667871e+00, -7.156964e-01};
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const float AudioFilter::m_hpb[3] = {8.459039e-01, -1.691760e+00, 8.459039e-01};
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AudioFilter::AudioFilter() :
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m_filterLP(m_lpa, m_lpb),
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m_filterHP(m_hpa, m_hpb),
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m_useHP(false)
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{}
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AudioFilter::~AudioFilter()
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{}
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void AudioFilter::setDecimFilters(int srHigh, int srLow, float fcHigh, float fcLow, float fgain)
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{
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double fcNormHigh = fcHigh / srHigh;
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double fcNormLow = fcLow / srLow;
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calculate2(false, fcNormHigh, m_lpva, m_lpvb, fgain);
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calculate2(true, fcNormLow, m_hpva, m_hpvb, fgain);
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m_filterLP.setCoeffs(m_lpva, m_lpvb);
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m_filterHP.setCoeffs(m_hpva, m_hpvb);
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}
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void AudioFilter::calculate2(bool highPass, double fc, float *va, float *vb, float fgain)
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{
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double a[22], b[22];
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cheby(highPass, fc, 0.5, 2, a, b, fgain); // low-pass, 0.5% ripple, 2 pole filter
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// Copy to the 2-pole filter coefficients
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for (int i=0; i<3; i++) {
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vb[i] = a[i];
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va[i] = b[i];
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}
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va[0] = 1.0;
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qDebug() << "AudioFilter::calculate2:"
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<< " highPass: " << highPass
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<< " fc: " << fc
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<< " a0: " << va[0]
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<< " a1: " << va[1]
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<< " a2: " << va[2]
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<< " b0: " << vb[0]
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<< " b1: " << vb[1]
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<< " b2: " << vb[2];
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}
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/*
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* Adapted from BASIC program in table 20-4 of
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* https://www.analog.com/media/en/technical-documentation/dsp-book/dsp_book_Ch20.pdf
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*/
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void AudioFilter::cheby(bool highPass, double fc, float pr, int np, double *a, double *b, float fgain)
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{
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double a0, a1, a2, b1, b2;
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double ta[22], tb[22];
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std::fill(a, a+22, 0.0);
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std::fill(b, b+22, 0.0);
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a[2] = 1.0;
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b[2] = 1.0;
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for (int p = 1; p <= np/2; p++)
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{
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cheby_sub(highPass, fc, pr, np, p, a0, a1, a2, b1, b2);
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// Add coefficients to the cascade
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for (int i=0; i<22; i++)
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{
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ta[i] = a[i];
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tb[i] = b[i];
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}
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for (int i=2; i<22; i++)
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{
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a[i] = a0*ta[i] + a1*ta[i-1] + a2*ta[i-2];
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b[i] = tb[i] - b1*tb[i-1] - b2*tb[i-2];
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}
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}
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// Finish combining coefficients
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b[2] = 0;
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for (int i=0; i<20; i++)
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{
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a[i] = a[i+2];
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b[i] = -b[i+2];
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}
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// Normalize the gain
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double sa = 0.0;
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double sb = 0.0;
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for (int i=0; i<20; i++)
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{
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if (highPass)
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{
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sa += i%2 == 0 ? a[i] : -a[i];
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sb += i%2 == 0 ? b[i] : -b[i];
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}
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else
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{
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sa += a[i];
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sb += b[i];
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}
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}
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double gain = sa/(1.0 -sb);
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gain /= fgain;
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for (int i=0; i<20; i++) {
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a[i] /= gain;
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}
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}
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/*
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* Adapted from BASIC subroutine in table 20-5 of
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* https://www.analog.com/media/en/technical-documentation/dsp-book/dsp_book_Ch20.pdf
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*/
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void AudioFilter::cheby_sub(bool highPass, double fc, float pr, int np, int stage, double& a0, double& a1, double& a2, double& b1, double& b2)
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{
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double rp = -cos((M_PI/(np*2)) + (stage-1)*(M_PI/np));
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double ip = sin((M_PI/(np*2)) + (stage-1)*(M_PI/np));
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// Warp from a circle to an ellipse
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double esx = 100.0 / (100.0 - pr);
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double es = sqrt(esx*esx -1.0);
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double vx = (1.0/np) * log((1.0/es) + sqrt((1.0/(es*es)) + 1.0));
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double kx = (1.0/np) * log((1.0/es) + sqrt((1.0/(es*es)) - 1.0));
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kx = (exp(kx) + exp(-kx))/2.0;
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rp = rp * ((exp(vx) - exp(-vx))/2.0) / kx;
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ip = ip * ((exp(vx) + exp(-vx))/2.0) / kx;
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double t = 2.0 * tan(0.5);
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double w = 2.0 * M_PI * fc;
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double m = rp*rp + ip*ip;
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double d = 4.0 - 4.0*rp*t + m*t*t;
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double x0 = (t*t)/d;
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double x1 = (2.0*t*t)/d;
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double x2 = (t*t)/d;
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double y1 = (8.0 - 2.0*m*t*t)/d;
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double y2 = (-4.0 - 4.0*rp*t - m*t*t)/d;
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double k;
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if (highPass) {
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k = -cos(w/2.0 + 0.5) / cos(w/2.0 - 0.5);
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} else {
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k = sin(0.5 - w/2.0) / sin(0.5 + w/2.0);
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}
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d = 1.0 + y1*k - y2*k*k;
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a0 = (x0 - x1*k + x2*k*k)/d;
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a1 = (-2.0*x0*k + x1 + x1*k*k - 2.0*x2*k)/d;
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a2 = (x0*k*k - x1*k + x2)/d;
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b1 = (2.0*k + y1 + y1*k*k - 2.0*y2*k)/d;
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b2 = (-(k*k) - y1*k + y2)/d;
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if (highPass)
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{
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a1 = -a1;
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b1 = -b1;
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}
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}
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float AudioFilter::run(const float& sample)
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{
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return m_useHP ? m_filterLP.run(m_filterHP.run(sample)) : m_filterLP.run(sample);
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}
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float AudioFilter::runHP(const float& sample)
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{
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return m_filterHP.run(sample);
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}
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float AudioFilter::runLP(const float& sample)
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{
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return m_filterLP.run(sample);
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}
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