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Implement variable cutoff frequency for audio filter
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f4exb
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@ -14,11 +14,16 @@
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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 <math.h>
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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 = 0.075):
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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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@ -26,12 +31,166 @@ 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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}
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{}
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AudioFilter::~AudioFilter()
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{}
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void AudioFilter::setDecimFilters(int sr, uint32_t decim)
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{
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int downSR = sr / (decim == 0 ? 1 : decim);
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double fcH = (0.45 * downSR) / (sr <= 0 ? 1 : sr); // high cut frequency normalized to SR
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double fcL = 300.0 / downSR; // low cut frequency normalized to downsampled SR
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calculate2(false, fcH, m_lpva, m_lpvb);
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calculate2(true, fcL, m_hpva, m_hpvb);
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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)
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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); // 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)
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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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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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@ -45,4 +204,4 @@ float AudioFilter::runHP(const float& sample)
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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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}
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@ -21,7 +21,7 @@
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#include "dsp/iirfilter.h"
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/**
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* 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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* 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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@ -36,6 +36,8 @@
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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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@ -45,18 +47,29 @@ public:
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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 sr, uint32_t decim);
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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); // two pole Chebyshev calculation
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void cheby(bool highPass, double fc, float pr, int np, double *a, double *b);
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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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#endif // _SDRBASE_AUDIO_AUDIOFILTER_H_
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