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direwolf/src/dsp.c

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//
// This file is part of Dire Wolf, an amateur radio packet TNC.
//
// Copyright (C) 2011, 2012, 2013, 2015, 2019 John Langner, WB2OSZ
//
// 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, see <http://www.gnu.org/licenses/>.
//
/*------------------------------------------------------------------
*
* Name: dsp.c
*
* Purpose: Generate the filters used by the demodulators.
*
*----------------------------------------------------------------*/
#include "direwolf.h"
#include <stdlib.h>
#include <stdio.h>
#include <math.h>
#include <unistd.h>
#include <string.h>
#include <ctype.h>
#include <assert.h>
#include "audio.h"
#include "fsk_demod_state.h"
#include "fsk_gen_filter.h"
#include "textcolor.h"
#include "dsp.h"
//#include "fsk_demod_agc.h" /* for M_FILTER_SIZE, etc. */
#define MIN(a,b) ((a)<(b)?(a):(b))
#define MAX(a,b) ((a)>(b)?(a):(b))
// Don't remove this. It serves as a reminder that an experiment is underway.
#if defined(TUNE_MS_FILTER_SIZE) || defined(TUNE_MS2_FILTER_SIZE) || defined(TUNE_AGC_FAST) || defined(TUNE_LPF_BAUD) || defined(TUNE_PLL_LOCKED) || defined(TUNE_PROFILE)
#define DEBUG1 1 // Don't remove this.
#endif
/*------------------------------------------------------------------
*
* Name: window
*
* Purpose: Filter window shape functions.
*
* Inputs: type - BP_WINDOW_HAMMING, etc.
* size - Number of filter taps.
* j - Index in range of 0 to size-1.
*
* Returns: Multiplier for the window shape.
*
*----------------------------------------------------------------*/
float window (bp_window_t type, int size, int j)
{
float center;
float w;
center = 0.5 * (size - 1);
switch (type) {
case BP_WINDOW_COSINE:
w = cos((j - center) / size * M_PI);
//w = sin(j * M_PI / (size - 1));
break;
case BP_WINDOW_HAMMING:
w = 0.53836 - 0.46164 * cos((j * 2 * M_PI) / (size - 1));
break;
case BP_WINDOW_BLACKMAN:
w = 0.42659 - 0.49656 * cos((j * 2 * M_PI) / (size - 1))
+ 0.076849 * cos((j * 4 * M_PI) / (size - 1));
break;
case BP_WINDOW_FLATTOP:
w = 1.0 - 1.93 * cos((j * 2 * M_PI) / (size - 1))
+ 1.29 * cos((j * 4 * M_PI) / (size - 1))
- 0.388 * cos((j * 6 * M_PI) / (size - 1))
+ 0.028 * cos((j * 8 * M_PI) / (size - 1));
break;
case BP_WINDOW_TRUNCATED:
default:
w = 1.0;
break;
}
return (w);
}
/*------------------------------------------------------------------
*
* Name: gen_lowpass
*
* Purpose: Generate low pass filter kernel.
*
* Inputs: fc - Cutoff frequency as fraction of sampling frequency.
* filter_size - Number of filter taps.
* wtype - Window type, BP_WINDOW_HAMMING, etc.
* lp_delay_fract - Fudge factor for the delay value.
*
* Outputs: lp_filter
*
* Returns: Signal delay thru the filter in number of audio samples.
*
*----------------------------------------------------------------*/
int gen_lowpass (float fc, float *lp_filter, int filter_size, bp_window_t wtype, float lp_delay_fract)
{
int j;
float G;
#if DEBUG1
text_color_set(DW_COLOR_DEBUG);
dw_printf ("Lowpass, size=%d, fc=%.2f\n", filter_size, fc);
dw_printf (" j shape sinc final\n");
#endif
assert (filter_size >= 3 && filter_size <= MAX_FILTER_SIZE);
for (j=0; j<filter_size; j++) {
float center;
float sinc;
float shape;
center = 0.5 * (filter_size - 1);
if (j - center == 0) {
sinc = 2 * fc;
}
else {
sinc = sin(2 * M_PI * fc * (j-center)) / (M_PI*(j-center));
}
shape = window (wtype, filter_size, j);
lp_filter[j] = sinc * shape;
#if DEBUG1
dw_printf ("%6d %6.2f %6.3f %6.3f\n", j, shape, sinc, lp_filter[j] ) ;
#endif
}
/*
* Normalize lowpass for unity gain at DC.
*/
G = 0;
for (j=0; j<filter_size; j++) {
G += lp_filter[j];
}
for (j=0; j<filter_size; j++) {
lp_filter[j] = lp_filter[j] / G;
}
// Calculate the signal delay.
// If a signal at level 0 steps to level 1, this is the time that it would
// take for the output to reach 0.5.
//
// Examples:
//
// Filter has one tap with value of 1.0.
// Output is immediate so I would call this delay of 0.
//
// Filter coefficients: 0.2, 0.2, 0.2, 0.2, 0.2
// "1" inputs Out
// 1 0.2
// 2 0.4
// 3 0.6
//
// In this case, the output does not change immediately.
// It takes two more samples to reach the half way point
// so it has a delay of 2.
float sum = 0;
int delay = 0;
if (lp_delay_fract == 0) lp_delay_fract = 0.5;
for (j=0; j<filter_size; j++) {
sum += lp_filter[j];
#if DEBUG1
dw_printf ("lp_filter[%d] = %.3f sum = %.3f lp_delay_fract = %.3f\n", j, lp_filter[j], sum, lp_delay_fract);
#endif
if (sum > lp_delay_fract) {
delay = j;
break;
}
}
#if DEBUG1
dw_printf ("Low Pass Delay = %d samples\n", delay) ;
#endif
// Hmmm. This might have been wasted effort. The result is always half the number of taps.
if (delay < 2 || delay > filter_size - 2) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Internal error, %s %d, delay %d for size %d\n", __func__, __LINE__, delay, filter_size);
}
return (delay);
} /* end gen_lowpass */
#undef DEBUG1
/*------------------------------------------------------------------
*
* Name: gen_bandpass
*
* Purpose: Generate band pass filter kernel for the prefilter.
* This is NOT for the mark/space filters.
*
* Inputs: f1 - Lower cutoff frequency as fraction of sampling frequency.
* f2 - Upper cutoff frequency...
* filter_size - Number of filter taps.
* wtype - Window type, BP_WINDOW_HAMMING, etc.
*
* Outputs: bp_filter
*
* Reference: http://www.labbookpages.co.uk/audio/firWindowing.html
*
* Does it need to be an odd length?
*
*----------------------------------------------------------------*/
void gen_bandpass (float f1, float f2, float *bp_filter, int filter_size, bp_window_t wtype)
{
int j;
float w;
float G;
float center = 0.5 * (filter_size - 1);
#if DEBUG1
text_color_set(DW_COLOR_DEBUG);
dw_printf ("Bandpass, size=%d\n", filter_size);
dw_printf (" j shape sinc final\n");
#endif
assert (filter_size >= 3 && filter_size <= MAX_FILTER_SIZE);
for (j=0; j<filter_size; j++) {
float sinc;
float shape;
if (j - center == 0) {
sinc = 2 * (f2 - f1);
}
else {
sinc = sin(2 * M_PI * f2 * (j-center)) / (M_PI*(j-center))
- sin(2 * M_PI * f1 * (j-center)) / (M_PI*(j-center));
}
shape = window (wtype, filter_size, j);
bp_filter[j] = sinc * shape;
#if DEBUG1
dw_printf ("%6d %6.2f %6.3f %6.3f\n", j, shape, sinc, bp_filter[j] ) ;
#endif
}
/*
* Normalize bandpass for unity gain in middle of passband.
* Can't use same technique as for lowpass.
* Instead compute gain in middle of passband.
* See http://dsp.stackexchange.com/questions/4693/fir-filter-gain
*/
w = 2 * M_PI * (f1 + f2) / 2;
G = 0;
for (j=0; j<filter_size; j++) {
G += 2 * bp_filter[j] * cos((j-center)*w); // is this correct?
}
#if DEBUG1
dw_printf ("Before normalizing, G=%.3f\n", G);
#endif
for (j=0; j<filter_size; j++) {
bp_filter[j] = bp_filter[j] / G;
}
} /* end gen_bandpass */
/*------------------------------------------------------------------
*
* Name: gen_ms
*
* Purpose: Generate mark and space filters.
*
* Inputs: fc - Tone frequency, i.e. mark or space.
* sps - Samples per second.
* filter_size - Number of filter taps.
* wtype - Window type, BP_WINDOW_HAMMING, etc.
*
* Outputs: bp_filter
*
* Reference: http://www.labbookpages.co.uk/audio/firWindowing.html
*
* Does it need to be an odd length?
*
*----------------------------------------------------------------*/
void gen_ms (int fc, int sps, float *sin_table, float *cos_table, int filter_size, int wtype)
{
int j;
float Gs = 0, Gc = 0;;
for (j=0; j<filter_size; j++) {
float center = 0.5f * (filter_size - 1);
float am = ((float)(j - center) / (float)sps) * ((float)fc) * (2.0f * (float)M_PI);
float shape = window (wtype, filter_size, j);
sin_table[j] = sinf(am) * shape;
cos_table[j] = cosf(am) * shape;
Gs += sin_table[j] * sinf(am);
Gc += cos_table[j] * cosf(am);
#if DEBUG1
dw_printf ("%6d %6.2f %6.2f %6.2f\n", j, shape, sin_table[j], cos_table[j]) ;
#endif
}
/* Normalize for unity gain */
#if DEBUG1
dw_printf ("Before normalizing, Gs = %.2f, Gc = %.2f\n", Gs, Gc) ;
#endif
for (j=0; j<filter_size; j++) {
sin_table[j] = sin_table[j] / Gs;
cos_table[j] = cos_table[j] / Gc;
}
} /* end gen_ms */
/* end dsp.c */
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