wdsp/fmd.c
Uladzimir Karpenka 89c8a0e2b5 first commit
2026-06-01 15:58:45 +03:00

385 lines
11 KiB
C

/* fmd.c
This file is part of a program that implements a Software-Defined Radio.
Copyright (C) 2013, 2023 Warren Pratt, NR0V
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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
The author can be reached by email at
warren@wpratt.com
*/
#include "comm.h"
void calc_fmd (FMD a)
{
// pll
a->omega_min = TWOPI * a->fmin / a->rate;
a->omega_max = TWOPI * a->fmax / a->rate;
a->g1 = 1.0 - exp(-2.0 * a->omegaN * a->zeta / a->rate);
a->g2 = -a->g1 + 2.0 * (1 - exp(-a->omegaN * a->zeta / a->rate) * cos(a->omegaN / a->rate * sqrt(1.0 - a->zeta * a->zeta)));
a->phs = 0.0;
a->fil_out = 0.0;
a->omega = 0.0;
a->pllpole = a->omegaN * sqrt(2.0 * a->zeta * a->zeta + 1.0 + sqrt((2.0 * a->zeta * a->zeta + 1.0) * (2.0 * a->zeta * a->zeta + 1.0) + 1)) / TWOPI;
// dc removal
a->mtau = exp(-1.0 / (a->rate * a->tau));
a->onem_mtau = 1.0 - a->mtau;
a->fmdc = 0.0;
// pll audio gain
a->again = a->rate / (a->deviation * TWOPI);
// CTCSS Removal
a->sntch = create_snotch(1, a->size, a->out, a->out, (int)a->rate, a->ctcss_freq, 0.0002);
// detector limiter
a->plim = create_wcpagc (
1, // run - always ON
5, // mode
1, // 0 for max(I,Q), 1 for envelope
a->out, // input buff pointer
a->out, // output buff pointer
a->size, // io_buffsize
(int)a->rate, // sample rate
0.001, // tau_attack
0.008, // tau_decay
4, // n_tau
a->lim_gain, // max_gain (sets threshold, initial value)
1.0, // var_gain / slope
1.0, // fixed_gain
1.0, // max_input
0.9, // out_targ
0.250, // tau_fast_backaverage
0.004, // tau_fast_decay
4.0, // pop_ratio
0, // hang_enable
0.500, // tau_hang_backmult
0.500, // hangtime
2.000, // hang_thresh
0.100); // tau_hang_decay
}
void decalc_fmd (FMD a)
{
destroy_wcpagc(a->plim);
destroy_snotch(a->sntch);
}
FMD create_fmd( int run, int size, double* in, double* out, int rate, double deviation, double f_low, double f_high,
double fmin, double fmax, double zeta, double omegaN, double tau, double afgain, int sntch_run, double ctcss_freq, int nc_de, int mp_de, int nc_aud, int mp_aud)
{
FMD a = (FMD) malloc0 (sizeof (fmd));
double* impulse;
a->run = run;
a->size = size;
a->in = in;
a->out = out;
a->rate = (double)rate;
a->deviation = deviation;
a->f_low = f_low;
a->f_high = f_high;
a->fmin = fmin;
a->fmax = fmax;
a->zeta = zeta;
a->omegaN = omegaN;
a->tau = tau;
a->afgain = afgain;
a->sntch_run = sntch_run;
a->ctcss_freq = ctcss_freq;
a->nc_de = nc_de;
a->mp_de = mp_de;
a->nc_aud = nc_aud;
a->mp_aud = mp_aud;
a->lim_run = 0;
a->lim_pre_gain = 0.4;
a->lim_gain = 2.5;
calc_fmd (a);
// de-emphasis filter
a->audio = (double *) malloc0 (a->size * sizeof (complex));
impulse = fc_impulse (a->nc_de, a->f_low, a->f_high, +20.0 * log10(a->f_high / a->f_low), 0.0, 1, a->rate, 1.0 / (2.0 * a->size), 0, 0);
a->pde = create_fircore (a->size, a->audio, a->out, a->nc_de, a->mp_de, impulse);
_aligned_free (impulse);
// audio filter
impulse = fir_bandpass(a->nc_aud, 0.8 * a->f_low, 1.1 * a->f_high, a->rate, 0, 1, a->afgain / (2.0 * a->size));
a->paud = create_fircore (a->size, a->out, a->out, a->nc_aud, a->mp_aud, impulse);
_aligned_free (impulse);
return a;
}
void destroy_fmd (FMD a)
{
destroy_fircore (a->paud);
destroy_fircore (a->pde);
_aligned_free (a->audio);
decalc_fmd (a);
_aligned_free (a);
}
void flush_fmd (FMD a)
{
memset (a->audio, 0, a->size * sizeof (complex));
flush_fircore (a->pde);
flush_fircore (a->paud);
a->phs = 0.0;
a->fil_out = 0.0;
a->omega = 0.0;
a->fmdc = 0.0;
flush_snotch (a->sntch);
flush_wcpagc (a->plim);
}
void xfmd (FMD a)
{
if (a->run)
{
int i;
double det, del_out;
double vco[2], corr[2];
for (i = 0; i < a->size; i++)
{
// pll
vco[0] = cos (a->phs);
vco[1] = sin (a->phs);
corr[0] = + a->in[2 * i + 0] * vco[0] + a->in[2 * i + 1] * vco[1];
corr[1] = - a->in[2 * i + 0] * vco[1] + a->in[2 * i + 1] * vco[0];
if ((corr[0] == 0.0) && (corr[1] == 0.0)) corr[0] = 1.0;
det = atan2 (corr[1], corr[0]);
del_out = a->fil_out;
a->omega += a->g2 * det;
if (a->omega < a->omega_min) a->omega = a->omega_min;
if (a->omega > a->omega_max) a->omega = a->omega_max;
a->fil_out = a->g1 * det + a->omega;
a->phs += del_out;
while (a->phs >= TWOPI) a->phs -= TWOPI;
while (a->phs < 0.0) a->phs += TWOPI;
// dc removal, gain, & demod output
a->fmdc = a->mtau * a->fmdc + a->onem_mtau * a->fil_out;
a->audio[2 * i + 0] = a->again * (a->fil_out - a->fmdc);
a->audio[2 * i + 1] = a->audio[2 * i + 0];
}
// de-emphasis
xfircore (a->pde);
// audio filter
xfircore (a->paud);
// CTCSS Removal
xsnotch (a->sntch);
if (a->lim_run)
{
for (i = 0; i < 2 * a->size; i++)
a->out[i] *= a->lim_pre_gain;
xwcpagc (a->plim);
}
}
else if (a->in != a->out)
memcpy (a->out, a->in, a->size * sizeof (complex));
}
void setBuffers_fmd (FMD a, double* in, double* out)
{
decalc_fmd (a);
a->in = in;
a->out = out;
calc_fmd (a);
setBuffers_fircore (a->pde, a->audio, a->out);
setBuffers_fircore (a->paud, a->out, a->out);
setBuffers_wcpagc (a->plim, a->out, a->out);
}
void setSamplerate_fmd (FMD a, int rate)
{
double* impulse;
decalc_fmd (a);
a->rate = rate;
calc_fmd (a);
// de-emphasis filter
impulse = fc_impulse (a->nc_de, a->f_low, a->f_high, +20.0 * log10(a->f_high / a->f_low), 0.0, 1, a->rate, 1.0 / (2.0 * a->size), 0, 0);
setImpulse_fircore (a->pde, impulse, 1);
_aligned_free (impulse);
// audio filter
impulse = fir_bandpass(a->nc_aud, 0.8 * a->f_low, 1.1 * a->f_high, a->rate, 0, 1, a->afgain / (2.0 * a->size));
setImpulse_fircore (a->paud, impulse, 1);
_aligned_free (impulse);
setSamplerate_wcpagc (a->plim, (int)a->rate);
}
void setSize_fmd (FMD a, int size)
{
double* impulse;
decalc_fmd (a);
_aligned_free (a->audio);
a->size = size;
calc_fmd (a);
a->audio = (double *) malloc0 (a->size * sizeof (complex));
// de-emphasis filter
destroy_fircore (a->pde);
impulse = fc_impulse (a->nc_de, a->f_low, a->f_high, +20.0 * log10(a->f_high / a->f_low), 0.0, 1, a->rate, 1.0 / (2.0 * a->size), 0, 0);
a->pde = create_fircore (a->size, a->audio, a->out, a->nc_de, a->mp_de, impulse);
_aligned_free (impulse);
// audio filter
destroy_fircore (a->paud);
impulse = fir_bandpass(a->nc_aud, 0.8 * a->f_low, 1.1 * a->f_high, a->rate, 0, 1, a->afgain / (2.0 * a->size));
a->paud = create_fircore (a->size, a->out, a->out, a->nc_aud, a->mp_aud, impulse);
_aligned_free (impulse);
setSize_wcpagc (a->plim, a->size);
}
/********************************************************************************************************
* *
* RXA Properties *
* *
********************************************************************************************************/
PORT
void SetRXAFMDeviation (int channel, double deviation)
{
FMD a;
EnterCriticalSection (&ch[channel].csDSP);
a = rxa[channel].fmd.p;
a->deviation = deviation;
a->again = a->rate / (a->deviation * TWOPI);
LeaveCriticalSection (&ch[channel].csDSP);
}
PORT
void SetRXACTCSSFreq (int channel, double freq)
{
FMD a;
EnterCriticalSection (&ch[channel].csDSP);
a = rxa[channel].fmd.p;
a->ctcss_freq = freq;
SetSNCTCSSFreq (a->sntch, a->ctcss_freq);
LeaveCriticalSection (&ch[channel].csDSP);
}
PORT
void SetRXACTCSSRun (int channel, int run)
{
FMD a;
EnterCriticalSection (&ch[channel].csDSP);
a = rxa[channel].fmd.p;
a->sntch_run = run;
SetSNCTCSSRun (a->sntch, a->sntch_run);
LeaveCriticalSection (&ch[channel].csDSP);
}
PORT
void SetRXAFMNCde (int channel, int nc)
{
FMD a;
double* impulse;
EnterCriticalSection (&ch[channel].csDSP);
a = rxa[channel].fmd.p;
if (a->nc_de != nc)
{
a->nc_de = nc;
impulse = fc_impulse (a->nc_de, a->f_low, a->f_high, +20.0 * log10(a->f_high / a->f_low), 0.0, 1, a->rate, 1.0 / (2.0 * a->size), 0, 0);
setNc_fircore (a->pde, a->nc_de, impulse);
_aligned_free (impulse);
}
LeaveCriticalSection (&ch[channel].csDSP);
}
PORT
void SetRXAFMMPde (int channel, int mp)
{
FMD a;
a = rxa[channel].fmd.p;
if (a->mp_de != mp)
{
a->mp_de = mp;
setMp_fircore (a->pde, a->mp_de);
}
}
PORT
void SetRXAFMNCaud (int channel, int nc)
{
FMD a;
double* impulse;
EnterCriticalSection (&ch[channel].csDSP);
a = rxa[channel].fmd.p;
if (a->nc_aud != nc)
{
a->nc_aud = nc;
impulse = fir_bandpass(a->nc_aud, 0.8 * a->f_low, 1.1 * a->f_high, a->rate, 0, 1, a->afgain / (2.0 * a->size));
setNc_fircore (a->paud, a->nc_aud, impulse);
_aligned_free (impulse);
}
LeaveCriticalSection (&ch[channel].csDSP);
}
PORT
void SetRXAFMMPaud (int channel, int mp)
{
FMD a;
a = rxa[channel].fmd.p;
if (a->mp_aud != mp)
{
a->mp_aud = mp;
setMp_fircore (a->paud, a->mp_aud);
}
}
PORT
void SetRXAFMLimRun (int channel, int run)
{
FMD a;
a = rxa[channel].fmd.p;
EnterCriticalSection(&ch[channel].csDSP);
if (a->lim_run != run)
{
a->lim_run = run;
}
LeaveCriticalSection(&ch[channel].csDSP);
}
PORT
void SetRXAFMLimGain (int channel, double gaindB)
{
double gain = pow(10.0, gaindB / 20.0);
FMD a = rxa[channel].fmd.p;
EnterCriticalSection(&ch[channel].csDSP);
if (a->lim_gain != gain)
{
decalc_fmd(a);
a->lim_gain = gain;
calc_fmd(a);
}
LeaveCriticalSection(&ch[channel].csDSP);
}
PORT
void SetRXAFMAFFilter(int channel, double low, double high)
{
FMD a = rxa[channel].fmd.p;
double* impulse;
EnterCriticalSection(&ch[channel].csDSP);
if (a->f_low != low || a->f_high != high)
{
a->f_low = low;
a->f_high = high;
// de-emphasis filter
impulse = fc_impulse (a->nc_de, a->f_low, a->f_high, +20.0 * log10(a->f_high / a->f_low), 0.0, 1, a->rate, 1.0 / (2.0 * a->size), 0, 0);
setImpulse_fircore (a->pde, impulse, 1);
_aligned_free (impulse);
// audio filter
impulse = fir_bandpass (a->nc_aud, 0.8 * a->f_low, 1.1 * a->f_high, a->rate, 0, 1, a->afgain / (2.0 * a->size));
setImpulse_fircore (a->paud, impulse, 1);
_aligned_free (impulse);
}
LeaveCriticalSection(&ch[channel].csDSP);
}