Restructure: Move fsk from common code to 'libfsk'
This commit is contained in:
Andreas Eversberg
6
src/libfsk/Makefile.am
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6
src/libfsk/Makefile.am
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AM_CPPFLAGS = -Wall -Wextra -g $(all_includes)
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noinst_LIBRARIES = libfsk.a
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libfsk_a_SOURCES = \
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fsk.c
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288
src/libfsk/fsk.c
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288
src/libfsk/fsk.c
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/* FSK audio processing (coherent FSK modem)
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*
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* (C) 2017 by Andreas Eversberg <jolly@eversberg.eu>
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* All Rights Reserved
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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 3 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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#include <stdio.h>
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#include <stdint.h>
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#include <stdlib.h>
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#include <string.h>
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#include <errno.h>
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#include <math.h>
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#include "../common/sample.h"
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#include "../common/debug.h"
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#include "fsk.h"
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#define PI M_PI
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/*
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* fsk = instance of fsk modem
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* inst = instance of user
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* send_bit() = function to be called whenever a new bit has to be sent
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* receive_bit() = function to be called whenever a new bit was received
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* samplerate = samplerate
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* bitrate = bits per second
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* f0, f1 = two frequencies for bit 0 and bit 1
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* level = level to modulate the frequencies
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* coherent = use coherent modulation (FFSK)
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* bitadjust = how much to adjust the sample clock when a bitchange was detected. (0 = nothing, don't use this, 0.5 full adjustment)
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*/
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int fsk_init(fsk_t *fsk, void *inst, int (*send_bit)(void *inst), void (*receive_bit)(void *inst, int bit, double quality, double level), int samplerate, double bitrate, double f0, double f1, double level, int coherent, double bitadjust)
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{
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double bandwidth;
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int i;
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int rc;
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PDEBUG(DDSP, DEBUG_DEBUG, "Setup FSK for Transceiver. (F0 = %.1f, F1 = %.1f, peak = %.1f)\n", f0, f1, level);
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memset(fsk, 0, sizeof(*fsk));
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/* gen sine table with deviation */
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fsk->sin_tab = calloc(65536+16384, sizeof(*fsk->sin_tab));
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if (!fsk->sin_tab) {
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fprintf(stderr, "No mem!\n");
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rc = -ENOMEM;
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goto error;
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}
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for (i = 0; i < 65536; i++)
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fsk->sin_tab[i] = sin((double)i / 65536.0 * 2.0 * PI) * level;
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fsk->inst = inst;
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fsk->rx_bit = -1;
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fsk->rx_bitadjust = bitadjust;
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fsk->receive_bit = receive_bit;
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fsk->tx_bit = -1;
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fsk->level = level;
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fsk->send_bit = send_bit;
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fsk->f0_deviation = (f0 - f1) / 2.0;
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fsk->f1_deviation = (f1 - f0) / 2.0;
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if (f0 < f1) {
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fsk->low_bit = 0;
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fsk->high_bit = 1;
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} else {
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fsk->low_bit = 1;
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fsk->high_bit = 0;
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}
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/* calculate bandwidth */
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bandwidth = fabs(f0 - f1) * 2.0;
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/* init fm demodulator */
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rc = fm_demod_init(&fsk->demod, (double)samplerate, (f0 + f1) / 2.0, bandwidth);
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if (rc < 0)
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goto error;
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fsk->bits_per_sample = (double)bitrate / (double)samplerate;
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PDEBUG(DDSP, DEBUG_DEBUG, "Bitduration of %.4f bits per sample @ %d.\n", fsk->bits_per_sample, samplerate);
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fsk->phaseshift65536[0] = f0 / (double)samplerate * 65536.0;
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PDEBUG(DDSP, DEBUG_DEBUG, "phaseshift65536[0] = %.4f\n", fsk->phaseshift65536[0]);
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fsk->phaseshift65536[1] = f1 / (double)samplerate * 65536.0;
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PDEBUG(DDSP, DEBUG_DEBUG, "phaseshift65536[1] = %.4f\n", fsk->phaseshift65536[1]);
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/* use coherent modulation, i.e. each bit has an integer number of
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* half waves and starts/ends at zero crossing
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*/
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if (coherent) {
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double waves;
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fsk->coherent = 1;
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waves = (f0 / bitrate);
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if (fabs(round(waves * 2) - (waves * 2)) > 0.001) {
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fprintf(stderr, "Failed to set coherent mode, half waves of F0 does not fit exactly into one bit, please fix!\n");
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abort();
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}
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fsk->cycles_per_bit65536[0] = waves * 65536.0;
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waves = (f1 / bitrate);
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if (fabs(round(waves * 2) - (waves * 2)) > 0.001) {
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fprintf(stderr, "Failed to set coherent mode, half waves of F1 does not fit exactly into one bit, please fix!\n");
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abort();
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}
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fsk->cycles_per_bit65536[1] = waves * 65536.0;
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}
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return 0;
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error:
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fsk_cleanup(fsk);
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return rc;
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}
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/* Cleanup transceiver instance. */
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void fsk_cleanup(fsk_t *fsk)
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{
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PDEBUG(DDSP, DEBUG_DEBUG, "Cleanup FSK for Transceiver.\n");
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if (fsk->sin_tab) {
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free(fsk->sin_tab);
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fsk->sin_tab = NULL;
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}
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fm_demod_exit(&fsk->demod);
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}
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//#define DEBUG_MODULATOR
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//#define DEBUG_FILTER
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/* Demodulates bits
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*
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* If bit is received, callback function send_bit() is called.
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*
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* We sample each bit 0.5 bits after polarity change.
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*
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* If we have a bit change, adjust sample counter towards one half bit duration.
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* We may have noise, so the bit change may be wrong or not at the correct place.
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* This can cause bit slips.
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* Therefore we change the sample counter only slightly, so bit slips may not
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* happen so quickly.
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*/
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void fsk_receive(fsk_t *fsk, sample_t *sample, int length)
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{
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sample_t I[length], Q[length], frequency[length], f;
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int i;
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int bit;
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double level, quality;
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/* demod samples to offset arround center frequency */
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fm_demodulate_real(&fsk->demod, frequency, length, sample, I, Q);
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for (i = 0; i < length; i++) {
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f = frequency[i];
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if (f < 0)
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bit = fsk->low_bit;
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else
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bit = fsk->high_bit;
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#ifdef DEBUG_FILTER
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printf("|%s| %.3f\n", debug_amplitude(f / fabs(fsk->f0_deviation)), f / fabs(fsk->f0_deviation));
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#endif
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if (fsk->rx_bit != bit) {
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#ifdef DEBUG_FILTER
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puts("bit change");
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#endif
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fsk->rx_bit = bit;
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if (fsk->rx_bitpos < 0.5) {
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fsk->rx_bitpos += fsk->rx_bitadjust;
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if (fsk->rx_bitpos > 0.5)
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fsk->rx_bitpos = 0.5;
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} else
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if (fsk->rx_bitpos > 0.5) {
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fsk->rx_bitpos -= fsk->rx_bitadjust;
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if (fsk->rx_bitpos < 0.5)
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fsk->rx_bitpos = 0.5;
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}
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}
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/* if bit counter reaches 1, we substract 1 and sample the bit */
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if (fsk->rx_bitpos >= 1.0) {
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/* peak level is the length of I/Q vector
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* since we filter out the unwanted modulation product, the vector is only half of length */
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level = sqrt(I[i] * I[i] + Q[i] * Q[i]) * 2.0;
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/* quality is defined on how accurat the target frequency it hit
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* if it is hit close to the center or close to double deviation from center, quality is close to 0 */
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if (bit == 0)
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quality = 1.0 - fabs((f - fsk->f0_deviation) / fsk->f0_deviation);
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else
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quality = 1.0 - fabs((f - fsk->f1_deviation) / fsk->f1_deviation);
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if (quality < 0)
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quality = 0;
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#ifdef DEBUG_FILTER
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printf("sample (level=%.3f, quality=%.3f)\n", level / fsk->level, quality);
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#endif
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/* adjust the values, because this is best we can get from fm demodulator */
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fsk->receive_bit(fsk->inst, bit, quality / 0.95, level);
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fsk->rx_bitpos -= 1.0;
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}
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fsk->rx_bitpos += fsk->bits_per_sample;
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}
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}
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/* modulate bits
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*
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* If first/next bit is required, callback function send_bit() is called.
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* If there is no (more) data to be transmitted, the callback functions shall
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* return -1. In this case, this function stops and returns the number of
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* samples that have been rendered so far, if any.
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*
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* For coherent mode (FSK), we round the phase on every bit change to the
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* next zero crossing. This prevents phase shifts due to rounding errors.
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*/
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int fsk_send(fsk_t *fsk, sample_t *sample, int length, int add)
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{
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int count = 0;
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double phase, phaseshift;
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phase = fsk->tx_phase65536;
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/* get next bit */
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if (fsk->tx_bit < 0) {
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next_bit:
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fsk->tx_bit = fsk->send_bit(fsk->inst);
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#ifdef DEBUG_MODULATOR
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printf("bit change to %d\n", fsk->tx_bit);
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#endif
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if (fsk->tx_bit < 0)
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goto done;
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/* correct phase when changing bit */
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if (fsk->coherent) {
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/* round phase to nearest zero crossing */
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if (phase > 16384.0 && phase < 49152.0)
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phase = 32768.0;
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else
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phase = 0;
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/* set phase according to current position in bit */
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phase += fsk->tx_bitpos * fsk->cycles_per_bit65536[fsk->tx_bit & 1];
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#ifdef DEBUG_MODULATOR
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printf("phase %.3f bitpos=%.6f\n", phase, fsk->tx_bitpos);
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#endif
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}
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}
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/* modulate bit */
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phaseshift = fsk->phaseshift65536[fsk->tx_bit & 1];
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while (count < length && fsk->tx_bitpos < 1.0) {
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if (add)
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sample[count++] += fsk->sin_tab[(uint16_t)phase];
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else
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sample[count++] = fsk->sin_tab[(uint16_t)phase];
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#ifdef DEBUG_MODULATOR
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printf("|%s|\n", debug_amplitude(fsk->sin_tab[(uint16_t)phase] / fsk->level));
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#endif
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phase += phaseshift;
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if (phase >= 65536.0)
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phase -= 65536.0;
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fsk->tx_bitpos += fsk->bits_per_sample;
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}
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if (fsk->tx_bitpos >= 1.0) {
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fsk->tx_bitpos -= 1.0;
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goto next_bit;
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}
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done:
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fsk->tx_phase65536 = phase;
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return count;
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}
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/* reset transmitter state, so we get a clean start */
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void fsk_tx_reset(fsk_t *fsk)
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{
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fsk->tx_phase65536 = 0;
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fsk->tx_bitpos = 0;
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fsk->tx_bit = -1;
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}
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30
src/libfsk/fsk.h
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30
src/libfsk/fsk.h
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@@ -0,0 +1,30 @@
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#include "../libfm/fm.h"
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typedef struct ffsk {
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void *inst;
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int (*send_bit)(void *inst);
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void (*receive_bit)(void *inst, int bit, double quality, double level);
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fm_demod_t demod;
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double bits_per_sample; /* fraction of a bit per sample */
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double *sin_tab; /* sine table with correct peak level */
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double phaseshift65536[2]; /* how much the phase of fsk synbol changes per sample */
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double cycles_per_bit65536[2]; /* cacles of one bit */
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double tx_phase65536; /* current transmit phase */
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double level; /* level (amplitude) of signal */
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int coherent; /* set, if coherent TX mode */
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double f0_deviation; /* deviation of frequencies, relative to center */
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double f1_deviation;
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int low_bit, high_bit; /* a low or high deviation means which bit? */
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int tx_bit; /* current transmitting bit (-1 if not set) */
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int rx_bit; /* current receiving bit (-1 if not yet measured) */
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double tx_bitpos; /* current transmit position in bit */
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double rx_bitpos; /* current receive position in bit (sampleclock) */
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double rx_bitadjust; /* how much does a bit change cause the sample clock to be adjusted in phase */
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} fsk_t;
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int fsk_init(fsk_t *fsk, void *inst, int (*send_bit)(void *inst), void (*receive_bit)(void *inst, int bit, double quality, double level), int samplerate, double bitrate, double f0, double f1, double level, int coherent, double bitadjust);
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void fsk_cleanup(fsk_t *fsk);
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void fsk_receive(fsk_t *fsk, sample_t *sample, int length);
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int fsk_send(fsk_t *fsk, sample_t *sample, int length, int add);
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void fsk_tx_reset(fsk_t *fsk);
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