449 lines
14 KiB
C
449 lines
14 KiB
C
/* FSK audio processing (FSK/FFSK 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 "../libsample/sample.h"
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#include "../libdebug/debug.h"
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#include "fsk.h"
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#define PI M_PI
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/* uncomment to see the modulated curve */
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//#define DEBUG_MODULATOR
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/* uncomment to see the shape of the filter */
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//#define DEBUG_MODULATOR_SHAPE
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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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* 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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* ffsk = use FFSK modulation (each symbol ends at zero crossing)
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*/
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int fsk_mod_init(fsk_mod_t *fsk, void *inst, int (*send_bit)(void *inst), int samplerate, double bitrate, double f0, double f1, double level, int ffsk, int filter)
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{
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double temp;
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int i;
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int rc;
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PDEBUG(DDSP, DEBUG_DEBUG, "Setup FSK for Transmitter. (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, 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->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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fsk->bits65536_per_sample = (double)bitrate / (double)samplerate * 65536.0;
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PDEBUG(DDSP, DEBUG_DEBUG, "Bitduration of %.4f bits per sample @ %d.\n", fsk->bits65536_per_sample / 65536.0, samplerate);
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fsk->phaseshift65536[0] = f0 / (double)samplerate * 65536.0;
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fsk->phaseshift65536[1] = f1 / (double)samplerate * 65536.0;
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PDEBUG(DDSP, DEBUG_DEBUG, "F0 = %.0f Hz (phaseshift65536[0] = %.4f)\n", f0, fsk->phaseshift65536[0]);
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PDEBUG(DDSP, DEBUG_DEBUG, "F1 = %.0f Hz (phaseshift65536[1] = %.4f)\n", f1, fsk->phaseshift65536[1]);
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/* use ffsk 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 (ffsk) {
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double waves;
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if (filter) {
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PDEBUG(DDSP, DEBUG_ERROR, "Cannot use FFSK with filter.\n");
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rc = -EINVAL;
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goto error;
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}
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PDEBUG(DDSP, DEBUG_DEBUG, "enable FFSK modulation mode\n");
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fsk->ffsk = 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 FFSK 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 FFSK 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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} else {
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fsk->cycles_per_bit65536[0] = f0 / bitrate * 65536.0;
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fsk->cycles_per_bit65536[1] = f1 / bitrate * 65536.0;
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}
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PDEBUG(DDSP, DEBUG_DEBUG, "F0 = %.0f Hz (cycles_per_bit65536[0] = %.4f)\n", f0, fsk->cycles_per_bit65536[0]);
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PDEBUG(DDSP, DEBUG_DEBUG, "F1 = %.0f Hz (cycles_per_bit65536[1] = %.4f)\n", f1, fsk->cycles_per_bit65536[1]);
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/* if filter is enabled, use a cosine shaped curve to change the phase each sample */
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if (filter) {
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fsk->phase_tab_0_1 = calloc(65536 + 65536, sizeof(*fsk->sin_tab));
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if (!fsk->phase_tab_0_1) {
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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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fsk->phase_tab_1_0 = fsk->phase_tab_0_1 + 65536;
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for (i = 0; i < 65536; i++) {
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temp = cos((double)i / 65536.0 * PI) / 2 + 0.5; /* half cosine going from 1 to 0 */
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fsk->phase_tab_0_1[i] = temp * fsk->phaseshift65536[0] + (1.0 - temp) * fsk->phaseshift65536[1];
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fsk->phase_tab_1_0[i] = temp * fsk->phaseshift65536[1] + (1.0 - temp) * fsk->phaseshift65536[0];
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#ifdef DEBUG_MODULATOR_SHAPE
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fsk->phase_tab_0_1[i] = 1.0 - temp;
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fsk->phase_tab_1_0[i] = temp;
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#endif
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}
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PDEBUG(DDSP, DEBUG_DEBUG, "Enable filter to smooth FSK transmission.\n");
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fsk->filter = 1;
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}
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/* must reset, because bit states must be initialized */
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fsk_mod_reset(fsk);
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return 0;
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error:
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fsk_mod_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_mod_cleanup(fsk_mod_t *fsk)
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{
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PDEBUG(DDSP, DEBUG_DEBUG, "Cleanup FSK for Transmitter.\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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if (fsk->phase_tab_0_1) {
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free(fsk->phase_tab_0_1);
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fsk->phase_tab_0_1 = NULL;
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fsk->phase_tab_1_0 = NULL;
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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 FFSK mode, 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_mod_send(fsk_mod_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_last_bit = fsk->tx_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 from %d to %d\n", fsk->tx_last_bit, fsk->tx_bit);
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#endif
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if (fsk->tx_bit < 0) {
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fsk_mod_reset(fsk);
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goto done;
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}
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fsk->tx_bit &= 1;
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/* correct phase when changing bit */
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if (fsk->ffsk) {
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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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#ifdef DEBUG_MODULATOR
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printf("phase %.3f bitpos=%.6f\n", phase / 65536.0, fsk->tx_bitpos65536 / 65536.0);
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#endif
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}
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if (!fsk->filter) {
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/* change phase forward to the current bit position */
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phase += fsk->tx_bitpos65536 / 65536.0 * fsk->cycles_per_bit65536[fsk->tx_bit];
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if (phase >= 65536.0)
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phase -= 65536.0;
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}
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}
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/* modulate bit */
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if (!fsk->filter || fsk->tx_last_bit < 0 || fsk->tx_last_bit == fsk->tx_bit) {
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/* without filtering or when there is no bit change */
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phaseshift = fsk->phaseshift65536[fsk->tx_bit];
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while (count < length && fsk->tx_bitpos65536 < 65536.0) {
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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| %d\n", debug_amplitude(fsk->sin_tab[(uint16_t)phase] / fsk->level), fsk->tx_bit);
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#endif
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#ifdef DEBUG_MODULATOR_SHAPE
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printf("|%s| %d\n", debug_amplitude(fsk->tx_bit), fsk->tx_bit);
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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_bitpos65536 += fsk->bits65536_per_sample;
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}
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} else if (fsk->tx_bit > fsk->tx_last_bit) {
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/* with cosine shape filter, going from phase of 0 to phase of 1 */
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while (count < length && fsk->tx_bitpos65536 < 65536.0) {
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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| 0->1\n", debug_amplitude(fsk->sin_tab[(uint16_t)phase] / fsk->level));
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#endif
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#ifdef DEBUG_MODULATOR_SHAPE
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printf("|%s|0->1\n", debug_amplitude(fsk->phase_tab_0_1[(uint16_t)fsk->tx_bitpos65536]));
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#endif
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phase += fsk->phase_tab_0_1[(uint16_t)fsk->tx_bitpos65536];
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if (phase >= 65536.0)
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phase -= 65536.0;
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fsk->tx_bitpos65536 += fsk->bits65536_per_sample;
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}
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} else {
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/* with cosine shape filter, going from phase of 1 to phase of 0 */
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while (count < length && fsk->tx_bitpos65536 < 65536.0) {
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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|1->0\n", debug_amplitude(fsk->sin_tab[(uint16_t)phase] / fsk->level));
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#endif
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#ifdef DEBUG_MODULATOR_SHAPE
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printf("|%s|1->0\n", debug_amplitude(fsk->phase_tab_1_0[(uint16_t)fsk->tx_bitpos65536]));
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#endif
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phase += fsk->phase_tab_1_0[(uint16_t)fsk->tx_bitpos65536];
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if (phase >= 65536.0)
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phase -= 65536.0;
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fsk->tx_bitpos65536 += fsk->bits65536_per_sample;
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}
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}
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if (fsk->tx_bitpos65536 >= 65536.0) {
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fsk->tx_bitpos65536 -= 65536.0;
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if (!fsk->filter) {
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/* change phase back to the point when bit has changed */
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phase -= fsk->tx_bitpos65536 / 65536.0 * fsk->cycles_per_bit65536[fsk->tx_bit];
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if (phase < 0.0)
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phase += 65536.0;
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}
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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_mod_reset(fsk_mod_t *fsk)
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{
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fsk->tx_phase65536 = 0.0;
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fsk->tx_bitpos65536 = 0.0;
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fsk->tx_bit = -1;
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fsk->tx_last_bit = -1;
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}
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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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* 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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* 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_demod_init(fsk_demod_t *fsk, void *inst, void (*receive_bit)(void *inst, int bit, double quality, double level), int samplerate, double bitrate, double f0, double f1, double bitadjust)
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{
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double bandwidth;
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int rc;
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PDEBUG(DDSP, DEBUG_DEBUG, "Setup FSK for Receiver. (F0 = %.1f, F1 = %.1f)\n", f0, f1);
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memset(fsk, 0, sizeof(*fsk));
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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->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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return 0;
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error:
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fsk_demod_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_demod_cleanup(fsk_demod_t *fsk)
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{
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PDEBUG(DDSP, DEBUG_DEBUG, "Cleanup FSK for Receiver.\n");
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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_demod_receive(fsk_demod_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 around 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) / 2), 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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/* if we have a pulse before we sampled a bit after last pulse */
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if (fsk->rx_change) {
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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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#ifdef DEBUG_FILTER
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printf("prematurely bit change (level=%.3f)\n", level);
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#endif
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/* quality is 0.0, because a prematurely level change is caused by noise and has nothing to measure. */
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fsk->receive_bit(fsk->inst, fsk->rx_bit, 0.0, level);
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}
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fsk->rx_change = 1;
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}
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/* if bit counter reaches 1, we subtract 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, quality);
|
|
#endif
|
|
fsk->receive_bit(fsk->inst, bit, quality, level);
|
|
fsk->rx_bitpos -= 1.0;
|
|
fsk->rx_change = 0;
|
|
}
|
|
fsk->rx_bitpos += fsk->bits_per_sample;
|
|
}
|
|
}
|
|
|