mirror of https://github.com/wb2osz/direwolf.git
373 lines
9.0 KiB
C
373 lines
9.0 KiB
C
//
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// This file is part of Dire Wolf, an amateur radio packet TNC.
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//
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// Copyright (C) 2011, 2012, 2013, 2015, 2019 John Langner, WB2OSZ
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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, either version 2 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 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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/*------------------------------------------------------------------
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*
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* Name: dsp.c
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*
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* Purpose: Generate the filters used by the demodulators.
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*
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*----------------------------------------------------------------*/
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#include "direwolf.h"
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#include <stdlib.h>
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#include <stdio.h>
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#include <math.h>
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#include <unistd.h>
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#include <string.h>
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#include <ctype.h>
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#include <assert.h>
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#include "audio.h"
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#include "fsk_demod_state.h"
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#include "fsk_gen_filter.h"
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#include "textcolor.h"
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#include "dsp.h"
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//#include "fsk_demod_agc.h" /* for M_FILTER_SIZE, etc. */
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#define MIN(a,b) ((a)<(b)?(a):(b))
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#define MAX(a,b) ((a)>(b)?(a):(b))
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// Don't remove this. It serves as a reminder that an experiment is underway.
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#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)
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#define DEBUG1 1 // Don't remove this.
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#endif
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/*------------------------------------------------------------------
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*
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* Name: window
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*
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* Purpose: Filter window shape functions.
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*
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* Inputs: type - BP_WINDOW_HAMMING, etc.
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* size - Number of filter taps.
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* j - Index in range of 0 to size-1.
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*
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* Returns: Multiplier for the window shape.
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*
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*----------------------------------------------------------------*/
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float window (bp_window_t type, int size, int j)
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{
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float center;
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float w;
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center = 0.5 * (size - 1);
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switch (type) {
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case BP_WINDOW_COSINE:
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w = cos((j - center) / size * M_PI);
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//w = sin(j * M_PI / (size - 1));
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break;
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case BP_WINDOW_HAMMING:
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w = 0.53836 - 0.46164 * cos((j * 2 * M_PI) / (size - 1));
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break;
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case BP_WINDOW_BLACKMAN:
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w = 0.42659 - 0.49656 * cos((j * 2 * M_PI) / (size - 1))
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+ 0.076849 * cos((j * 4 * M_PI) / (size - 1));
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break;
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case BP_WINDOW_FLATTOP:
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w = 1.0 - 1.93 * cos((j * 2 * M_PI) / (size - 1))
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+ 1.29 * cos((j * 4 * M_PI) / (size - 1))
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- 0.388 * cos((j * 6 * M_PI) / (size - 1))
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+ 0.028 * cos((j * 8 * M_PI) / (size - 1));
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break;
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case BP_WINDOW_TRUNCATED:
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default:
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w = 1.0;
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break;
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}
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return (w);
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}
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/*------------------------------------------------------------------
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*
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* Name: gen_lowpass
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*
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* Purpose: Generate low pass filter kernel.
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*
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* Inputs: fc - Cutoff frequency as fraction of sampling frequency.
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* filter_size - Number of filter taps.
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* wtype - Window type, BP_WINDOW_HAMMING, etc.
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* lp_delay_fract - Fudge factor for the delay value.
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*
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* Outputs: lp_filter
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*
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* Returns: Signal delay thru the filter in number of audio samples.
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*
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*----------------------------------------------------------------*/
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int gen_lowpass (float fc, float *lp_filter, int filter_size, bp_window_t wtype, float lp_delay_fract)
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{
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int j;
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float G;
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#if DEBUG1
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text_color_set(DW_COLOR_DEBUG);
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dw_printf ("Lowpass, size=%d, fc=%.2f\n", filter_size, fc);
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dw_printf (" j shape sinc final\n");
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#endif
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assert (filter_size >= 3 && filter_size <= MAX_FILTER_SIZE);
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for (j=0; j<filter_size; j++) {
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float center;
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float sinc;
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float shape;
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center = 0.5 * (filter_size - 1);
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if (j - center == 0) {
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sinc = 2 * fc;
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}
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else {
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sinc = sin(2 * M_PI * fc * (j-center)) / (M_PI*(j-center));
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}
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shape = window (wtype, filter_size, j);
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lp_filter[j] = sinc * shape;
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#if DEBUG1
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dw_printf ("%6d %6.2f %6.3f %6.3f\n", j, shape, sinc, lp_filter[j] ) ;
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#endif
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}
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/*
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* Normalize lowpass for unity gain at DC.
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*/
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G = 0;
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for (j=0; j<filter_size; j++) {
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G += lp_filter[j];
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}
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for (j=0; j<filter_size; j++) {
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lp_filter[j] = lp_filter[j] / G;
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}
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// Calculate the signal delay.
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// If a signal at level 0 steps to level 1, this is the time that it would
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// take for the output to reach 0.5.
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//
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// Examples:
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//
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// Filter has one tap with value of 1.0.
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// Output is immediate so I would call this delay of 0.
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//
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// Filter coefficients: 0.2, 0.2, 0.2, 0.2, 0.2
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// "1" inputs Out
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// 1 0.2
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// 2 0.4
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// 3 0.6
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//
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// In this case, the output does not change immediately.
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// It takes two more samples to reach the half way point
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// so it has a delay of 2.
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float sum = 0;
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int delay = 0;
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if (lp_delay_fract == 0) lp_delay_fract = 0.5;
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for (j=0; j<filter_size; j++) {
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sum += lp_filter[j];
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#if DEBUG1
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dw_printf ("lp_filter[%d] = %.3f sum = %.3f lp_delay_fract = %.3f\n", j, lp_filter[j], sum, lp_delay_fract);
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#endif
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if (sum > lp_delay_fract) {
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delay = j;
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break;
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}
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}
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#if DEBUG1
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dw_printf ("Low Pass Delay = %d samples\n", delay) ;
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#endif
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// Hmmm. This might have been wasted effort. The result is always half the number of taps.
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if (delay < 2 || delay > filter_size - 2) {
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text_color_set(DW_COLOR_ERROR);
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dw_printf ("Internal error, %s %d, delay %d for size %d\n", __func__, __LINE__, delay, filter_size);
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}
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return (delay);
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} /* end gen_lowpass */
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#undef DEBUG1
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/*------------------------------------------------------------------
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*
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* Name: gen_bandpass
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*
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* Purpose: Generate band pass filter kernel for the prefilter.
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* This is NOT for the mark/space filters.
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*
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* Inputs: f1 - Lower cutoff frequency as fraction of sampling frequency.
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* f2 - Upper cutoff frequency...
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* filter_size - Number of filter taps.
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* wtype - Window type, BP_WINDOW_HAMMING, etc.
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*
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* Outputs: bp_filter
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*
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* Reference: http://www.labbookpages.co.uk/audio/firWindowing.html
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*
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* Does it need to be an odd length?
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*
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*----------------------------------------------------------------*/
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void gen_bandpass (float f1, float f2, float *bp_filter, int filter_size, bp_window_t wtype)
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{
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int j;
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float w;
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float G;
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float center = 0.5 * (filter_size - 1);
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#if DEBUG1
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text_color_set(DW_COLOR_DEBUG);
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dw_printf ("Bandpass, size=%d\n", filter_size);
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dw_printf (" j shape sinc final\n");
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#endif
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assert (filter_size >= 3 && filter_size <= MAX_FILTER_SIZE);
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for (j=0; j<filter_size; j++) {
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float sinc;
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float shape;
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if (j - center == 0) {
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sinc = 2 * (f2 - f1);
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}
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else {
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sinc = sin(2 * M_PI * f2 * (j-center)) / (M_PI*(j-center))
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- sin(2 * M_PI * f1 * (j-center)) / (M_PI*(j-center));
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}
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shape = window (wtype, filter_size, j);
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bp_filter[j] = sinc * shape;
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#if DEBUG1
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dw_printf ("%6d %6.2f %6.3f %6.3f\n", j, shape, sinc, bp_filter[j] ) ;
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#endif
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}
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/*
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* Normalize bandpass for unity gain in middle of passband.
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* Can't use same technique as for lowpass.
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* Instead compute gain in middle of passband.
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* See http://dsp.stackexchange.com/questions/4693/fir-filter-gain
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*/
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w = 2 * M_PI * (f1 + f2) / 2;
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G = 0;
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for (j=0; j<filter_size; j++) {
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G += 2 * bp_filter[j] * cos((j-center)*w); // is this correct?
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}
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#if DEBUG1
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dw_printf ("Before normalizing, G=%.3f\n", G);
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#endif
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for (j=0; j<filter_size; j++) {
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bp_filter[j] = bp_filter[j] / G;
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}
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} /* end gen_bandpass */
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/*------------------------------------------------------------------
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*
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* Name: gen_ms
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*
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* Purpose: Generate mark and space filters.
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*
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* Inputs: fc - Tone frequency, i.e. mark or space.
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* sps - Samples per second.
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* filter_size - Number of filter taps.
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* wtype - Window type, BP_WINDOW_HAMMING, etc.
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*
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* Outputs: bp_filter
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*
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* Reference: http://www.labbookpages.co.uk/audio/firWindowing.html
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*
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* Does it need to be an odd length?
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*
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*----------------------------------------------------------------*/
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void gen_ms (int fc, int sps, float *sin_table, float *cos_table, int filter_size, int wtype)
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{
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int j;
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float Gs = 0, Gc = 0;;
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for (j=0; j<filter_size; j++) {
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float center = 0.5f * (filter_size - 1);
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float am = ((float)(j - center) / (float)sps) * ((float)fc) * (2.0f * (float)M_PI);
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float shape = window (wtype, filter_size, j);
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sin_table[j] = sinf(am) * shape;
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cos_table[j] = cosf(am) * shape;
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Gs += sin_table[j] * sinf(am);
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Gc += cos_table[j] * cosf(am);
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#if DEBUG1
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dw_printf ("%6d %6.2f %6.2f %6.2f\n", j, shape, sin_table[j], cos_table[j]) ;
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#endif
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}
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/* Normalize for unity gain */
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#if DEBUG1
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dw_printf ("Before normalizing, Gs = %.2f, Gc = %.2f\n", Gs, Gc) ;
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#endif
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for (j=0; j<filter_size; j++) {
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sin_table[j] = sin_table[j] / Gs;
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cos_table[j] = cos_table[j] / Gc;
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}
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} /* end gen_ms */
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/* end dsp.c */
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