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/*
dsp/SVF.h
Copyright 2002-4 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
ladder filter in Chamberlin topology. supports largely independent
f and Q adjustments and sweeps.
*/
/*
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; either version 2
of the License, or (at your option) any later version.
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 for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
/*
inspired by this music-dsp entry:
State Variable Filter (Double Sampled, Stable)
Type : 2 Pole Low, High, Band, Notch and Peaking
References :Posted by Andrew Simper
Notes :
Thanks to Laurent de Soras for the stability limit
and Steffan Diedrichsen for the correct notch output.
Code :
input = input buffer;
output = output buffer;
fs = sampling frequency;
fc = cutoff frequency normally something like:
440.0*pow(2.0, (midi_note - 69.0)/12.0);
res = resonance 0 to 1;
drive = internal distortion 0 to 0.1
freq = MIN(0.25, 2.0*sin(PI*fc/(fs*2))); // the fs*2 is because it's double sampled
damp = MIN(2.0*(1.0 - pow(res, 0.25)), MIN(2.0, 2.0/freq - freq*0.5));
notch = notch output
low = low pass output
high = high pass output
band = band pass output
peak = peaking output = low - high
--
double sampled svf loop:
for (i=0; i<numSamples; i++)
{
in = input[i];
notch = in - damp*band;
low = low + freq*band;
high = notch - low;
band = freq*high + band - drive*band*band*band;
out = 0.5*(notch or low or high or band or peak);
notch = in - damp*band;
low = low + freq*band;
high = notch - low;
band = freq*high + band - drive*band*band*band;
out += 0.5*(same out as above);
output[i] = out;
}
*/
#ifndef _DSP_SVF_H_
#define _DSP_SVF_H_
namespace DSP {
template <int OVERSAMPLE>
class SVF
{
protected:
/* loop parameters */
d_sample f, q, qnorm;
/* outputs (peak and notch left out) */
d_sample lo, band, hi;
d_sample * out;
public:
/* the type of filtering to do. */
enum {
Low = 0,
Band = 1,
High = 2
};
SVF()
{
set_out (Low);
set_f_Q (.1, .1);
}
void reset()
{
hi = band = lo = 0;
}
void set_f_Q (double fc, double Q)
{
/* this is a very tight limit */
f = min (.25, 2 * sin (M_PI * fc / OVERSAMPLE));
q = 2 * cos (pow (Q, .1) * M_PI * .5);
q = min (q, min (2., 2 / f - f * .5));
qnorm = sqrt (fabs (q) / 2. + .001);
}
void set_out (int o)
{
if (o == Low)
out = &lo;
else if (o == Band)
out = &band;
else
out = &hi;
}
void one_cycle (d_sample * s, int frames)
{
for (int i = 0; i < frames; ++i)
s[i] = process (s[i]);
}
d_sample process (d_sample x)
{
x = qnorm * x;
for (int pass = 0; pass < OVERSAMPLE; ++pass)
{
hi = x - lo - q * band;
band += f * hi;
lo += f * band;
/* zero-padding, not 0th order holding. */
x = 0;
}
/* peak and notch outputs don't belong in the loop, put them
* here (best in a template) if needed. */
return *out;
}
};
template <int STACKED, int OVERSAMPLE>
class StackedSVF
{
public:
SVF<OVERSAMPLE> svf [STACKED];
void reset()
{
for (int i = 0; i < STACKED; ++i)
svf[i].reset();
}
void set_out (int out)
{
for (int i = 0; i < STACKED; ++i)
svf[i].set_out (out);
}
void set_f_Q (double f, double Q)
{
for (int i = 0; i < STACKED; ++i)
svf[i].set_f_Q (f, Q);
}
d_sample process (d_sample x)
{
for (int i = 0; i < STACKED; ++i)
x = svf[i].process (x);
return x;
}
};
} /* namespace DSP */
#endif /* _DSP_SVF_H_ */
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