Move FFSK modem from NMT to common code, so it can be used by other networks
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92ce6d4a42
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@ -24,6 +24,7 @@ libcommon_a_SOURCES = \
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compandor.c \
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fft.c \
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fm_modulation.c \
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ffsk.c \
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sender.c \
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display_wave.c \
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display_status.c \
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@ -0,0 +1,256 @@
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/* FFSK audio processing (NMT / Radiocom 2000)
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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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#define CHAN ffsk->channel
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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 "ffsk.h"
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#define PI M_PI
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#define BIT_RATE 1200 /* baud rate */
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#define FILTER_STEPS 0.1 /* step every 1/12000 sec */
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/* two signaling tones */
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static double ffsk_freq[2] = {
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1800.0,
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1200.0,
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};
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static sample_t dsp_tone_bit[2][2][65536]; /* polarity, bit, phase */
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/* global init for FFSK */
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void ffsk_global_init(double peak_fsk)
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{
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int i;
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double s;
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PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table for FFSK tones.\n");
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for (i = 0; i < 65536; i++) {
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s = sin((double)i / 65536.0 * 2.0 * PI);
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/* bit(1) 1 cycle */
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dsp_tone_bit[0][1][i] = s * peak_fsk;
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dsp_tone_bit[1][1][i] = -s * peak_fsk;
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/* bit(0) 1.5 cycles */
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s = sin((double)i / 65536.0 * 3.0 * PI);
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dsp_tone_bit[0][0][i] = s * peak_fsk;
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dsp_tone_bit[1][0][i] = -s * peak_fsk;
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}
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}
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/* Init FFSK */
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int ffsk_init(ffsk_t *ffsk, void *inst, void (*receive_bit)(void *inst, int bit, double quality, double level), int channel, int samplerate)
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{
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sample_t *spl;
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int i;
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/* a symbol rate of 1200 Hz, times check interval of FILTER_STEPS */
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if (samplerate < (double)BIT_RATE / (double)FILTER_STEPS) {
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PDEBUG(DDSP, DEBUG_ERROR, "Sample rate must be at least 12000 Hz to process FSK+supervisory signal.\n");
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return -EINVAL;
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}
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memset(ffsk, 0, sizeof(*ffsk));
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ffsk->inst = inst;
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ffsk->receive_bit = receive_bit;
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ffsk->channel = channel;
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ffsk->samplerate = samplerate;
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ffsk->samples_per_bit = (double)ffsk->samplerate / (double)BIT_RATE;
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ffsk->bits_per_sample = 1.0 / ffsk->samples_per_bit;
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PDEBUG(DDSP, DEBUG_DEBUG, "Use %.4f samples for full bit duration @ %d.\n", ffsk->samples_per_bit, ffsk->samplerate);
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/* allocate ring buffers, one bit duration */
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ffsk->filter_size = floor(ffsk->samples_per_bit); /* buffer holds one bit (rounded down) */
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spl = calloc(1, ffsk->filter_size * sizeof(*spl));
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if (!spl) {
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PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
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ffsk_cleanup(ffsk);
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return -ENOMEM;
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}
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ffsk->filter_spl = spl;
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ffsk->filter_bit = -1;
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/* count symbols */
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for (i = 0; i < 2; i++)
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audio_goertzel_init(&ffsk->goertzel[i], ffsk_freq[i], ffsk->samplerate);
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ffsk->phaseshift65536 = 65536.0 / ffsk->samples_per_bit;
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PDEBUG(DDSP, DEBUG_DEBUG, "fsk_phaseshift = %.4f\n", ffsk->phaseshift65536);
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return 0;
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}
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/* Cleanup transceiver instance. */
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void ffsk_cleanup(ffsk_t *ffsk)
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{
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PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Cleanup DSP for Transceiver.\n");
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if (ffsk->filter_spl) {
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free(ffsk->filter_spl);
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ffsk->filter_spl = NULL;
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}
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}
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//#define DEBUG_MODULATOR
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//#define DEBUG_FILTER
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//#define DEBUG_QUALITY
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/* Filter one chunk of audio an detect tone, quality and loss of signal.
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* The chunk is a window of 1/1200s. This window slides over audio stream
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* and is processed every 1/12000s. (one step) */
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static inline void ffsk_decode_step(ffsk_t *ffsk, int pos)
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{
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double level, result[2], softbit, quality;
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int max;
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sample_t *spl;
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int bit;
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max = ffsk->filter_size;
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spl = ffsk->filter_spl;
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level = audio_level(spl, max);
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/* limit level to prevent division by zero */
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if (level < 0.001)
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level = 0.001;
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audio_goertzel(ffsk->goertzel, spl, max, pos, result, 2);
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/* calculate soft bit from both frequencies */
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softbit = (result[1] / level - result[0] / level + 1.0) / 2.0;
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//printf("%.3f: %.3f\n", level, softbit);
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/* scale it, since both filters overlap by some percent */
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#define MIN_QUALITY 0.33
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softbit = (softbit - MIN_QUALITY) / (1.0 - MIN_QUALITY - MIN_QUALITY);
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#ifdef DEBUG_FILTER
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// printf("|%s", debug_amplitude(result[0]/level));
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// printf("|%s| low=%.3f high=%.3f level=%d\n", debug_amplitude(result[1]/level), result[0]/level, result[1]/level, (int)level);
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printf("|%s| softbit=%.3f\n", debug_amplitude(softbit), softbit);
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#endif
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if (softbit > 1)
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softbit = 1;
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if (softbit < 0)
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softbit = 0;
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if (softbit > 0.5)
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bit = 1;
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else
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bit = 0;
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if (ffsk->filter_bit != bit) {
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/* If we have a bit change, move 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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#ifdef DEBUG_FILTER
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puts("bit change");
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#endif
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ffsk->filter_bit = bit;
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if (ffsk->filter_sample < 5)
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ffsk->filter_sample++;
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if (ffsk->filter_sample > 5)
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ffsk->filter_sample--;
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} else if (--ffsk->filter_sample == 0) {
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/* if sample counter bit reaches 0, we reset sample counter to one bit duration */
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#ifdef DEBUG_FILTER
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puts("sample");
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#endif
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// quality = result[bit] / level;
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if (softbit > 0.5)
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quality = softbit * 2.0 - 1.0;
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else
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quality = 1.0 - softbit * 2.0;
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#ifdef DEBUG_QUALITY
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printf("|%s| quality=%.2f ", debug_amplitude(softbit), quality);
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printf("|%s|\n", debug_amplitude(quality));
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#endif
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/* adjust level, so a peak level becomes 100% */
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ffsk->receive_bit(ffsk->inst, bit, quality, level / 0.63662);
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ffsk->filter_sample = 10;
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}
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}
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void ffsk_receive(ffsk_t *ffsk, sample_t *sample, int length)
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{
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sample_t *spl;
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int max, pos;
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double step, bps;
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int i;
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/* write received samples to decode buffer */
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max = ffsk->filter_size;
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pos = ffsk->filter_pos;
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step = ffsk->filter_step;
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bps = ffsk->bits_per_sample;
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spl = ffsk->filter_spl;
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for (i = 0; i < length; i++) {
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#ifdef DEBUG_MODULATOR
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printf("|%s|\n", debug_amplitude((double)samples[i] / 2333.0 /*fsk peak*/ / 2.0));
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#endif
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/* write into ring buffer */
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spl[pos++] = sample[i];
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if (pos == max)
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pos = 0;
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/* if 1/10th of a bit duration is reached, decode buffer */
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step += bps;
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if (step >= FILTER_STEPS) {
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step -= FILTER_STEPS;
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ffsk_decode_step(ffsk, pos);
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}
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}
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ffsk->filter_step = step;
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ffsk->filter_pos = pos;
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}
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/* render frame */
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int ffsk_render_frame(ffsk_t *ffsk, const char *frame, int length, sample_t *sample)
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{
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int bit, polarity;
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double phaseshift, phase;
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int count = 0, i;
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polarity = ffsk->polarity;
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phaseshift = ffsk->phaseshift65536;
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phase = ffsk->phase65536;
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for (i = 0; i < length; i++) {
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bit = (frame[i] == '1');
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do {
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*sample++ = dsp_tone_bit[polarity][bit][(uint16_t)phase];
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count++;
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phase += phaseshift;
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} while (phase < 65536.0);
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phase -= 65536.0;
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/* flip polarity when we have 1.5 sine waves */
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if (bit == 0)
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polarity = 1 - polarity;
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}
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ffsk->phase65536 = phase;
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ffsk->polarity = polarity;
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/* return number of samples created for frame */
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return count;
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}
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@ -0,0 +1,27 @@
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#include "../common/goertzel.h"
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typedef struct ffsk {
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void *inst;
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void (*receive_bit)(void *inst, int bit, double quality, double level);
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int channel; /* channel number */
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int samplerate; /* current sample rate */
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double samples_per_bit; /* number of samples for one bit (1200 Baud) */
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double bits_per_sample; /* fraction of a bit per sample */
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goertzel_t goertzel[2]; /* filter for fsk decoding */
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int polarity; /* current polarity state of bit */
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sample_t *filter_spl; /* array to hold ring buffer for bit decoding */
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int filter_size; /* size of ring buffer */
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int filter_pos; /* position to write next sample */
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double filter_step; /* counts bit duration, to trigger decoding every 10th bit */
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int filter_bit; /* last bit state, so we detect a bit change */
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int filter_sample; /* count until it is time to sample bit */
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double phaseshift65536; /* how much the phase of fsk synbol changes per sample */
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double phase65536; /* current phase */
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} ffsk_t;
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void ffsk_global_init(double peak_fsk);
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int ffsk_init(ffsk_t *ffsk, void *inst, void (*receive_bit)(void *inst, int bit, double quality, double level), int channel, int samplerate);
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void ffsk_cleanup(ffsk_t *ffsk);
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void ffsk_receive(ffsk_t *ffsk, sample_t *sample, int lenght);
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int ffsk_render_frame(ffsk_t *ffsk, const char *frame, int length, sample_t *sample);
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@ -25,7 +25,6 @@
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#include "../common/debug.h"
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#include "../common/timer.h"
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#include "nmt.h"
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#include "dsp.h"
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#define MUTE_DURATION 0.300 /* 200ms, and about 95ms for the frame itself */
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#endif
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/* render wave form */
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dms->frame_length = fsk_render_frame(nmt, frame, 127, dms->frame_spl);
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test_dms_frame(frame, 127); // used by test program
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dms->frame_length = ffsk_render_frame(&nmt->ffsk, frame, 127, dms->frame_spl);
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dms->frame_valid = 1;
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dms->frame_pos = 0;
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if (dms->frame_length > dms->frame_size) {
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@ -335,7 +335,8 @@ static void dms_encode_rr(nmt_t *nmt, uint8_t d, uint8_t s, uint8_t n)
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#endif
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/* render wave form */
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dms->frame_length = fsk_render_frame(nmt, frame, 77, dms->frame_spl);
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test_dms_frame(frame, 77); // used by test program
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dms->frame_length = ffsk_render_frame(&nmt->ffsk, frame, 77, dms->frame_spl);
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dms->frame_valid = 1;
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dms->frame_pos = 0;
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if (dms->frame_length > dms->frame_size) {
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@ -662,7 +663,7 @@ void fsk_receive_bit_dms(nmt_t *nmt, int bit, double quality, double level)
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memset(dms->rx_frame_quality, 0, sizeof(dms->rx_frame_quality));
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/* set muting of receive path */
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nmt->fsk_filter_mute = (int)((double)nmt->sender.samplerate * MUTE_DURATION);
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nmt->rx_mute = (int)((double)nmt->sender.samplerate * MUTE_DURATION);
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return;
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}
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void dms_all_sent(nmt_t *nmt);
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void dms_receive(nmt_t *nmt, const uint8_t *data, int length, int eight_bits);
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void test_dms_frame(const char *frame, int length);
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260
src/nmt/dsp.c
260
src/nmt/dsp.c
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@ -59,9 +59,8 @@
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#define COMPANDOR_0DB 1.0 /* A level of 0dBm (1.0) shall be unaccected */
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#define TX_PEAK_FSK (4200.0 / 1800.0 * 1000.0 / DBM0_DEVIATION)
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#define TX_PEAK_SUPER (300.0 / 4015.0 * 1000.0 / DBM0_DEVIATION)
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#define BIT_RATE 1200
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#define MAX_DISPLAY 1.4 /* something above dBm0 */
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#define BIT_RATE 1200 /* baud rate */
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#define FILTER_STEPS 0.1 /* step every 1/12000 sec */
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#define DIALTONE_HZ 425.0 /* dial tone frequency */
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#define TX_PEAK_DIALTONE 0.5 /* dial tone peak FIXME */
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#define SUPER_DURATION 0.25 /* duration of supervisory signal measurement */
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#define SUPER_DETECT_COUNT 6 /* number of measures to detect supervisory signal */
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#define MUTE_DURATION 0.280 /* a tiny bit more than two frames */
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/* two signaling tones */
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static double fsk_freq[2] = {
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1800.0,
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1200.0,
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};
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/* two supervisory tones */
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static double super_freq[5] = {
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3955.0, /* 0-Signal 1 */
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};
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/* table for fast sine generation */
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static sample_t dsp_tone_bit[2][2][65536]; /* polarity, bit, phase */
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static sample_t dsp_sine_super[65536];
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static sample_t dsp_sine_dialtone[65536];
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/* global init for FSK */
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/* global init for FFSK */
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void dsp_init(void)
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{
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int i;
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double s;
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PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table for supervisory signal.\n");
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PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table for supervisory signal and dial tone.\n");
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for (i = 0; i < 65536; i++) {
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s = sin((double)i / 65536.0 * 2.0 * PI);
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/* supervisor sine */
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dsp_sine_super[i] = s * TX_PEAK_SUPER;
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/* dialtone sine */
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dsp_sine_dialtone[i] = s * TX_PEAK_DIALTONE;
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/* bit(1) 1 cycle */
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dsp_tone_bit[0][1][i] = s * TX_PEAK_FSK;
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dsp_tone_bit[1][1][i] = -s * TX_PEAK_FSK;
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/* bit(0) 1.5 cycles */
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s = sin((double)i / 65536.0 * 3.0 * PI);
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dsp_tone_bit[0][0][i] = s * TX_PEAK_FSK;
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dsp_tone_bit[1][0][i] = -s * TX_PEAK_FSK;
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}
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ffsk_global_init(TX_PEAK_FSK);
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}
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static void fsk_receive_bit(void *inst, int bit, double quality, double level);
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/* Init FSK of transceiver */
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int dsp_init_sender(nmt_t *nmt, double deviation_factor)
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{
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sample_t *spl;
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double samples_per_bit;
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int i;
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/* attack (3ms) and recovery time (13.5ms) according to NMT specs */
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init_compandor(&nmt->cstate, 8000, 3.0, 13.5, COMPANDOR_0DB);
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/* a symbol rate of 1200 Hz, times check interval of FILTER_STEPS */
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if (nmt->sender.samplerate < (double)BIT_RATE / (double)FILTER_STEPS) {
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PDEBUG(DDSP, DEBUG_ERROR, "Sample rate must be at least 12000 Hz to process FSK+supervisory signal.\n");
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return -EINVAL;
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}
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PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Init DSP for Transceiver.\n");
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/* set modulation parameters */
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@ -135,22 +119,16 @@ int dsp_init_sender(nmt_t *nmt, double deviation_factor)
|
|||
PDEBUG(DDSP, DEBUG_DEBUG, "Using FSK level of %.3f (%.3f KHz deviation @ 1500 Hz)\n", TX_PEAK_FSK * deviation_factor, 3.5 * deviation_factor);
|
||||
PDEBUG(DDSP, DEBUG_DEBUG, "Using Supervisory level of %.3f (%.3f KHz deviation @ 4015 Hz)\n", TX_PEAK_SUPER * deviation_factor, 0.3 * deviation_factor);
|
||||
|
||||
nmt->fsk_samples_per_bit = (double)nmt->sender.samplerate / (double)BIT_RATE;
|
||||
nmt->fsk_bits_per_sample = 1.0 / nmt->fsk_samples_per_bit;
|
||||
PDEBUG(DDSP, DEBUG_DEBUG, "Use %.4f samples for full bit duration @ %d.\n", nmt->fsk_samples_per_bit, nmt->sender.samplerate);
|
||||
|
||||
/* allocate ring buffers, one bit duration */
|
||||
nmt->fsk_filter_size = floor(nmt->fsk_samples_per_bit); /* buffer holds one bit (rounded down) */
|
||||
spl = calloc(1, nmt->fsk_filter_size * sizeof(*spl));
|
||||
if (!spl) {
|
||||
PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
|
||||
return -ENOMEM;
|
||||
/* init ffsk */
|
||||
if (ffsk_init(&nmt->ffsk, nmt, fsk_receive_bit, nmt->sender.kanal, nmt->sender.samplerate) < 0) {
|
||||
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "FFSK init failed!\n");
|
||||
return -EINVAL;
|
||||
}
|
||||
nmt->fsk_filter_spl = spl;
|
||||
nmt->fsk_filter_bit = -1;
|
||||
|
||||
/* allocate transmit buffer for a complete frame, add 10 to be safe */
|
||||
nmt->frame_size = 166.0 * (double)nmt->fsk_samples_per_bit + 10;
|
||||
|
||||
samples_per_bit = (double)nmt->sender.samplerate / (double)BIT_RATE;
|
||||
nmt->frame_size = 166.0 * samples_per_bit + 10;
|
||||
spl = calloc(nmt->frame_size, sizeof(*spl));
|
||||
if (!spl) {
|
||||
PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
|
||||
|
@ -159,7 +137,7 @@ int dsp_init_sender(nmt_t *nmt, double deviation_factor)
|
|||
nmt->frame_spl = spl;
|
||||
|
||||
/* allocate DMS transmit buffer for a complete frame, add 10 to be safe */
|
||||
nmt->dms.frame_size = 127.0 * (double)nmt->fsk_samples_per_bit + 10;
|
||||
nmt->dms.frame_size = 127.0 * samples_per_bit + 10;
|
||||
spl = calloc(nmt->dms.frame_size, sizeof(*spl));
|
||||
if (!spl) {
|
||||
PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
|
||||
|
@ -176,12 +154,6 @@ int dsp_init_sender(nmt_t *nmt, double deviation_factor)
|
|||
}
|
||||
nmt->super_filter_spl = spl;
|
||||
|
||||
/* count symbols */
|
||||
for (i = 0; i < 2; i++)
|
||||
audio_goertzel_init(&nmt->fsk_goertzel[i], fsk_freq[i], nmt->sender.samplerate);
|
||||
nmt->fsk_phaseshift65536 = 65536.0 / nmt->fsk_samples_per_bit;
|
||||
PDEBUG(DDSP, DEBUG_DEBUG, "fsk_phaseshift = %.4f\n", nmt->fsk_phaseshift65536);
|
||||
|
||||
/* count supervidory tones */
|
||||
for (i = 0; i < 5; i++) {
|
||||
audio_goertzel_init(&nmt->super_goertzel[i], super_freq[i], nmt->sender.samplerate);
|
||||
|
@ -207,6 +179,8 @@ void dsp_cleanup_sender(nmt_t *nmt)
|
|||
{
|
||||
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Cleanup DSP for Transceiver.\n");
|
||||
|
||||
ffsk_cleanup(&nmt->ffsk);
|
||||
|
||||
if (nmt->frame_spl) {
|
||||
free(nmt->frame_spl);
|
||||
nmt->frame_spl = NULL;
|
||||
|
@ -215,10 +189,6 @@ void dsp_cleanup_sender(nmt_t *nmt)
|
|||
free(nmt->dms.frame_spl);
|
||||
nmt->dms.frame_spl = NULL;
|
||||
}
|
||||
if (nmt->fsk_filter_spl) {
|
||||
free(nmt->fsk_filter_spl);
|
||||
nmt->fsk_filter_spl = NULL;
|
||||
}
|
||||
if (nmt->super_filter_spl) {
|
||||
free(nmt->super_filter_spl);
|
||||
nmt->super_filter_spl = NULL;
|
||||
|
@ -226,29 +196,38 @@ void dsp_cleanup_sender(nmt_t *nmt)
|
|||
}
|
||||
|
||||
/* Check for SYNC bits, then collect data bits */
|
||||
static void fsk_receive_bit(nmt_t *nmt, int bit, double quality, double level)
|
||||
static void fsk_receive_bit(void *inst, int bit, double quality, double level)
|
||||
{
|
||||
double frames_elapsed;
|
||||
nmt_t *nmt = (nmt_t *)inst;
|
||||
uint64_t frames_elapsed;
|
||||
int i;
|
||||
|
||||
/* normalize FSK level */
|
||||
level /= TX_PEAK_FSK;
|
||||
|
||||
nmt->rx_bits_count++;
|
||||
|
||||
if (nmt->dms_call)
|
||||
fsk_receive_bit_dms(nmt, bit, quality, level);
|
||||
|
||||
// printf("bit=%d quality=%.4f\n", bit, quality);
|
||||
if (!nmt->fsk_filter_in_sync) {
|
||||
nmt->fsk_filter_sync = (nmt->fsk_filter_sync << 1) | bit;
|
||||
if (!nmt->rx_in_sync) {
|
||||
nmt->rx_sync = (nmt->rx_sync << 1) | bit;
|
||||
|
||||
/* level and quality */
|
||||
nmt->fsk_filter_level[nmt->fsk_filter_count & 0xff] = level;
|
||||
nmt->fsk_filter_quality[nmt->fsk_filter_count & 0xff] = quality;
|
||||
nmt->fsk_filter_count++;
|
||||
nmt->rx_level[nmt->rx_count & 0xff] = level;
|
||||
nmt->rx_quality[nmt->rx_count & 0xff] = quality;
|
||||
nmt->rx_count++;
|
||||
|
||||
/* check if pattern 1010111100010010 matches */
|
||||
if (nmt->fsk_filter_sync != 0xaf12)
|
||||
if (nmt->rx_sync != 0xaf12)
|
||||
return;
|
||||
|
||||
/* average level and quality */
|
||||
level = quality = 0;
|
||||
for (i = 0; i < 16; i++) {
|
||||
level += nmt->fsk_filter_level[(nmt->fsk_filter_count - 1 - i) & 0xff];
|
||||
quality += nmt->fsk_filter_quality[(nmt->fsk_filter_count - 1 - i) & 0xff];
|
||||
level += nmt->rx_level[(nmt->rx_count - 1 - i) & 0xff];
|
||||
quality += nmt->rx_quality[(nmt->rx_count - 1 - i) & 0xff];
|
||||
}
|
||||
level /= 16.0; quality /= 16.0;
|
||||
// printf("sync (level = %.2f, quality = %.2f\n", level, quality);
|
||||
|
@ -262,114 +241,38 @@ static void fsk_receive_bit(nmt_t *nmt, int bit, double quality, double level)
|
|||
nmt->rx_bits_count_current = nmt->rx_bits_count - 26.0;
|
||||
|
||||
/* rest sync register */
|
||||
nmt->fsk_filter_sync = 0;
|
||||
nmt->fsk_filter_in_sync = 1;
|
||||
nmt->fsk_filter_count = 0;
|
||||
nmt->rx_sync = 0;
|
||||
nmt->rx_in_sync = 1;
|
||||
nmt->rx_count = 0;
|
||||
|
||||
/* set muting of receive path */
|
||||
nmt->fsk_filter_mute = (int)((double)nmt->sender.samplerate * MUTE_DURATION);
|
||||
nmt->rx_mute = (int)((double)nmt->sender.samplerate * MUTE_DURATION);
|
||||
return;
|
||||
}
|
||||
|
||||
/* read bits */
|
||||
nmt->fsk_filter_frame[nmt->fsk_filter_count] = bit + '0';
|
||||
nmt->fsk_filter_level[nmt->fsk_filter_count] = level;
|
||||
nmt->fsk_filter_quality[nmt->fsk_filter_count] = quality;
|
||||
if (++nmt->fsk_filter_count != 140)
|
||||
nmt->rx_frame[nmt->rx_count] = bit + '0';
|
||||
nmt->rx_level[nmt->rx_count] = level;
|
||||
nmt->rx_quality[nmt->rx_count] = quality;
|
||||
if (++nmt->rx_count != 140)
|
||||
return;
|
||||
|
||||
/* end of frame */
|
||||
nmt->fsk_filter_frame[140] = '\0';
|
||||
nmt->fsk_filter_in_sync = 0;
|
||||
nmt->rx_frame[140] = '\0';
|
||||
nmt->rx_in_sync = 0;
|
||||
|
||||
/* average level and quality */
|
||||
level = quality = 0;
|
||||
for (i = 0; i < 140; i++) {
|
||||
level += nmt->fsk_filter_level[i];
|
||||
quality += nmt->fsk_filter_quality[i];
|
||||
level += nmt->rx_level[i];
|
||||
quality += nmt->rx_quality[i];
|
||||
}
|
||||
level /= 140.0; quality /= 140.0;
|
||||
|
||||
/* send telegramm */
|
||||
frames_elapsed = (nmt->rx_bits_count_current - nmt->rx_bits_count_last) / 166.0;
|
||||
frames_elapsed = (nmt->rx_bits_count_current - nmt->rx_bits_count_last + 83) / 166; /* round to nearest frame */
|
||||
/* convert level so that received level at TX_PEAK_FSK results in 1.0 (100%) */
|
||||
nmt_receive_frame(nmt, nmt->fsk_filter_frame, quality, level, frames_elapsed);
|
||||
}
|
||||
|
||||
//#define DEBUG_MODULATOR
|
||||
//#define DEBUG_FILTER
|
||||
//#define DEBUG_QUALITY
|
||||
|
||||
/* Filter one chunk of audio an detect tone, quality and loss of signal.
|
||||
* The chunk is a window of 1/1200s. This window slides over audio stream
|
||||
* and is processed every 1/12000s. (one step) */
|
||||
static inline void fsk_decode_step(nmt_t *nmt, int pos)
|
||||
{
|
||||
double level, result[2], softbit, quality;
|
||||
int max;
|
||||
sample_t *spl;
|
||||
int bit;
|
||||
|
||||
max = nmt->fsk_filter_size;
|
||||
spl = nmt->fsk_filter_spl;
|
||||
|
||||
/* count time in bits */
|
||||
nmt->rx_bits_count += FILTER_STEPS;
|
||||
|
||||
level = audio_level(spl, max);
|
||||
/* limit level to prevent division by zero */
|
||||
if (level < 0.001)
|
||||
level = 0.001;
|
||||
|
||||
audio_goertzel(nmt->fsk_goertzel, spl, max, pos, result, 2);
|
||||
|
||||
/* calculate soft bit from both frequencies */
|
||||
softbit = (result[1] / level - result[0] / level + 1.0) / 2.0;
|
||||
//printf("%.3f: %.3f\n", level, softbit);
|
||||
/* scale it, since both filters overlap by some percent */
|
||||
#define MIN_QUALITY 0.33
|
||||
softbit = (softbit - MIN_QUALITY) / (1.0 - MIN_QUALITY - MIN_QUALITY);
|
||||
#ifdef DEBUG_FILTER
|
||||
// printf("|%s", debug_amplitude(result[0]/level));
|
||||
// printf("|%s| low=%.3f high=%.3f level=%d\n", debug_amplitude(result[1]/level), result[0]/level, result[1]/level, (int)level);
|
||||
printf("|%s| softbit=%.3f\n", debug_amplitude(softbit), softbit);
|
||||
#endif
|
||||
if (softbit > 1)
|
||||
softbit = 1;
|
||||
if (softbit < 0)
|
||||
softbit = 0;
|
||||
if (softbit > 0.5)
|
||||
bit = 1;
|
||||
else
|
||||
bit = 0;
|
||||
|
||||
if (nmt->fsk_filter_bit != bit) {
|
||||
/* if we have a bit change, reset sample counter to one half bit duration */
|
||||
#ifdef DEBUG_FILTER
|
||||
puts("bit change");
|
||||
#endif
|
||||
nmt->fsk_filter_bit = bit;
|
||||
nmt->fsk_filter_sample = 5;
|
||||
} else if (--nmt->fsk_filter_sample == 0) {
|
||||
/* if sample counter bit reaches 0, we reset sample counter to one bit duration */
|
||||
#ifdef DEBUG_FILTER
|
||||
puts("sample");
|
||||
#endif
|
||||
// quality = result[bit] / level;
|
||||
if (softbit > 0.5)
|
||||
quality = softbit * 2.0 - 1.0;
|
||||
else
|
||||
quality = 1.0 - softbit * 2.0;
|
||||
#ifdef DEBUG_QUALITY
|
||||
printf("|%s| quality=%.2f ", debug_amplitude(softbit), quality);
|
||||
printf("|%s|\n", debug_amplitude(quality));
|
||||
#endif
|
||||
/* adjust level, so a peak level becomes 100% */
|
||||
fsk_receive_bit(nmt, bit, quality, level / 0.63662 / TX_PEAK_FSK);
|
||||
if (nmt->dms_call)
|
||||
fsk_receive_bit_dms(nmt, bit, quality, level / 0.63662 / TX_PEAK_FSK);
|
||||
nmt->fsk_filter_sample = 10;
|
||||
}
|
||||
nmt_receive_frame(nmt, nmt->rx_frame, quality, level, frames_elapsed);
|
||||
}
|
||||
|
||||
/* compare supervisory signal against noise floor on 3900 Hz */
|
||||
|
@ -425,7 +328,6 @@ void sender_receive(sender_t *sender, sample_t *samples, int length)
|
|||
nmt_t *nmt = (nmt_t *) sender;
|
||||
sample_t *spl;
|
||||
int max, pos;
|
||||
double step, bps;
|
||||
int i;
|
||||
|
||||
/* write received samples to decode buffer */
|
||||
|
@ -442,34 +344,15 @@ void sender_receive(sender_t *sender, sample_t *samples, int length)
|
|||
}
|
||||
nmt->super_filter_pos = pos;
|
||||
|
||||
/* write received samples to decode buffer */
|
||||
max = nmt->fsk_filter_size;
|
||||
pos = nmt->fsk_filter_pos;
|
||||
step = nmt->fsk_filter_step;
|
||||
bps = nmt->fsk_bits_per_sample;
|
||||
spl = nmt->fsk_filter_spl;
|
||||
ffsk_receive(&nmt->ffsk, samples, length);
|
||||
|
||||
/* muting audio while receiving frame */
|
||||
for (i = 0; i < length; i++) {
|
||||
#ifdef DEBUG_MODULATOR
|
||||
printf("|%s|\n", debug_amplitude((double)samples[i] / TX_PEAK_FSK / 2.0));
|
||||
#endif
|
||||
/* write into ring buffer */
|
||||
spl[pos++] = samples[i];
|
||||
if (pos == max)
|
||||
pos = 0;
|
||||
/* muting audio while receiving frame */
|
||||
if (nmt->fsk_filter_mute && !nmt->sender.loopback) {
|
||||
if (nmt->rx_mute && !nmt->sender.loopback) {
|
||||
samples[i] = 0;
|
||||
nmt->fsk_filter_mute--;
|
||||
}
|
||||
/* if 1/10th of a bit duration is reached, decode buffer */
|
||||
step += bps;
|
||||
if (step >= FILTER_STEPS) {
|
||||
step -= FILTER_STEPS;
|
||||
fsk_decode_step(nmt, pos);
|
||||
nmt->rx_mute--;
|
||||
}
|
||||
}
|
||||
nmt->fsk_filter_step = step;
|
||||
nmt->fsk_filter_pos = pos;
|
||||
|
||||
if ((nmt->dsp_mode == DSP_MODE_AUDIO || nmt->dsp_mode == DSP_MODE_DTMF)
|
||||
&& nmt->trans && nmt->trans->callref) {
|
||||
|
@ -494,35 +377,6 @@ void sender_receive(sender_t *sender, sample_t *samples, int length)
|
|||
nmt->sender.rxbuf_pos = 0;
|
||||
}
|
||||
|
||||
/* render frame */
|
||||
int fsk_render_frame(nmt_t *nmt, const char *frame, int length, sample_t *sample)
|
||||
{
|
||||
int bit, polarity;
|
||||
double phaseshift, phase;
|
||||
int count = 0, i;
|
||||
|
||||
polarity = nmt->fsk_polarity;
|
||||
phaseshift = nmt->fsk_phaseshift65536;
|
||||
phase = nmt->fsk_phase65536;
|
||||
for (i = 0; i < length; i++) {
|
||||
bit = (frame[i] == '1');
|
||||
do {
|
||||
*sample++ = dsp_tone_bit[polarity][bit][(uint16_t)phase];
|
||||
count++;
|
||||
phase += phaseshift;
|
||||
} while (phase < 65536.0);
|
||||
phase -= 65536.0;
|
||||
/* flip polarity when we have 1.5 sine waves */
|
||||
if (bit == 0)
|
||||
polarity = 1 - polarity;
|
||||
}
|
||||
nmt->fsk_phase65536 = phase;
|
||||
nmt->fsk_polarity = polarity;
|
||||
|
||||
/* return number of samples created for frame */
|
||||
return count;
|
||||
}
|
||||
|
||||
static int fsk_frame(nmt_t *nmt, sample_t *samples, int length)
|
||||
{
|
||||
const char *frame;
|
||||
|
@ -539,7 +393,7 @@ next_frame:
|
|||
return length;
|
||||
}
|
||||
/* render frame */
|
||||
nmt->frame_length = fsk_render_frame(nmt, frame, 166, nmt->frame_spl);
|
||||
nmt->frame_length = ffsk_render_frame(&nmt->ffsk, frame, 166, nmt->frame_spl);
|
||||
nmt->frame_pos = 0;
|
||||
if (nmt->frame_length > nmt->frame_size) {
|
||||
PDEBUG_CHAN(DDSP, DEBUG_ERROR, "Frame exceeds buffer, please fix!\n");
|
||||
|
|
|
@ -2,7 +2,6 @@
|
|||
void dsp_init(void);
|
||||
int dsp_init_sender(nmt_t *nmt, double deviation_factor);
|
||||
void dsp_cleanup_sender(nmt_t *nmt);
|
||||
int fsk_render_frame(nmt_t *nmt, const char *frame, int length, sample_t *sample);
|
||||
void nmt_set_dsp_mode(nmt_t *nmt, enum dsp_mode mode);
|
||||
void super_reset(nmt_t *nmt);
|
||||
|
||||
|
|
|
@ -427,3 +427,6 @@ fail:
|
|||
return 0;
|
||||
}
|
||||
|
||||
// dummy, will be replaced by DMS test program
|
||||
void test_dms_frame(const char __attribute__((unused)) *frame, int __attribute__((unused)) length) {}
|
||||
|
||||
|
|
|
@ -1527,7 +1527,7 @@ static void tx_active(nmt_t *nmt, frame_t *frame)
|
|||
* general handlers to call sub handling
|
||||
*/
|
||||
|
||||
void nmt_receive_frame(nmt_t *nmt, const char *bits, double quality, double level, double frames_elapsed)
|
||||
void nmt_receive_frame(nmt_t *nmt, const char *bits, double quality, double level, int frames_elapsed)
|
||||
{
|
||||
frame_t frame;
|
||||
int rc;
|
||||
|
@ -1541,7 +1541,7 @@ void nmt_receive_frame(nmt_t *nmt, const char *bits, double quality, double leve
|
|||
}
|
||||
|
||||
/* frame counter */
|
||||
nmt->rx_frame_count += (int)(frames_elapsed + 0.5);
|
||||
nmt->rx_frame_count += frames_elapsed;
|
||||
|
||||
PDEBUG_CHAN(DNMT, (nmt->sender.loopback) ? DEBUG_NOTICE : DEBUG_DEBUG, "Received frame %s\n", nmt_frame_name(frame.mt));
|
||||
|
||||
|
|
|
@ -1,8 +1,8 @@
|
|||
#include "../common/goertzel.h"
|
||||
#include "../common/sender.h"
|
||||
#include "../common/compandor.h"
|
||||
#include "../common/dtmf.h"
|
||||
#include "../common/call.h"
|
||||
#include "../common/ffsk.h"
|
||||
#include "dms.h"
|
||||
#include "sms.h"
|
||||
|
||||
|
@ -96,40 +96,29 @@ typedef struct nmt {
|
|||
|
||||
/* dsp states */
|
||||
enum dsp_mode dsp_mode; /* current mode: audio, durable tone 0 or 1, paging */
|
||||
double fsk_samples_per_bit; /* number of samples for one bit (1200 Baud) */
|
||||
double fsk_bits_per_sample; /* fraction of a bit per sample */
|
||||
ffsk_t ffsk; /* ffsk processing */
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int super_samples; /* number of samples in buffer for supervisory detection */
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goertzel_t fsk_goertzel[2]; /* filter for fsk decoding */
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||||
goertzel_t super_goertzel[5]; /* filter for supervisory decoding */
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int fsk_polarity; /* current polarity state of bit */
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||||
sample_t *fsk_filter_spl; /* array to hold ring buffer for bit decoding */
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int fsk_filter_size; /* size of ring buffer */
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int fsk_filter_pos; /* position to write next sample */
|
||||
double fsk_filter_step; /* counts bit duration, to trigger decoding every 10th bit */
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||||
int fsk_filter_bit; /* last bit state, so we detect a bit change */
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||||
int fsk_filter_sample; /* count until it is time to sample bit */
|
||||
uint16_t fsk_filter_sync; /* shift register to detect sync */
|
||||
int fsk_filter_in_sync; /* if we are in sync and receive bits */
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||||
int fsk_filter_mute; /* mute count down after sync */
|
||||
char fsk_filter_frame[141]; /* receive frame (one extra byte to terminate string) */
|
||||
int fsk_filter_count; /* next bit to receive */
|
||||
double fsk_filter_level[256]; /* level infos */
|
||||
double fsk_filter_quality[256];/* quality infos */
|
||||
sample_t *super_filter_spl; /* array with sample buffer for supervisory detection */
|
||||
int super_filter_pos; /* current sample position in filter_spl */
|
||||
double super_phaseshift65536[4];/* how much the phase of sine wave changes per sample */
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||||
double super_phase65536; /* current phase */
|
||||
double dial_phaseshift65536; /* how much the phase of sine wave changes per sample */
|
||||
double dial_phase65536; /* current phase */
|
||||
double fsk_phaseshift65536; /* how much the phase of fsk synbol changes per sample */
|
||||
double fsk_phase65536; /* current phase */
|
||||
uint16_t rx_sync; /* shift register to detect sync */
|
||||
int rx_in_sync; /* if we are in sync and receive bits */
|
||||
int rx_mute; /* mute count down after sync */
|
||||
char rx_frame[141]; /* receive frame (one extra byte to terminate string) */
|
||||
int rx_count; /* next bit to receive */
|
||||
double rx_level[256]; /* level infos */
|
||||
double rx_quality[256]; /* quality infos */
|
||||
sample_t *frame_spl; /* samples to store a complete rendered frame */
|
||||
int frame_size; /* total size of sample buffer */
|
||||
int frame_length; /* current length of data in sample buffer */
|
||||
int frame_pos; /* current sample position in frame_spl */
|
||||
double rx_bits_count; /* sample counter */
|
||||
double rx_bits_count_current; /* sample counter of current frame */
|
||||
double rx_bits_count_last; /* sample counter of last frame */
|
||||
uint64_t rx_bits_count; /* sample counter */
|
||||
uint64_t rx_bits_count_current; /* sample counter of current frame */
|
||||
uint64_t rx_bits_count_last; /* sample counter of last frame */
|
||||
int super_detected; /* current detection state flag */
|
||||
int super_detect_count; /* current number of consecutive detections/losses */
|
||||
|
||||
|
@ -149,7 +138,7 @@ int nmt_create(int nmt_system, const char *country, int channel, enum nmt_chan_t
|
|||
void nmt_check_channels(int nmt_system);
|
||||
void nmt_destroy(sender_t *sender);
|
||||
void nmt_go_idle(nmt_t *nmt);
|
||||
void nmt_receive_frame(nmt_t *nmt, const char *bits, double quality, double level, double frames_elapsed);
|
||||
void nmt_receive_frame(nmt_t *nmt, const char *bits, double quality, double level, int frames_elapsed);
|
||||
const char *nmt_get_frame(nmt_t *nmt);
|
||||
void nmt_rx_super(nmt_t *nmt, int tone, double quality);
|
||||
void timeout_mt_paging(struct transaction *trans);
|
||||
|
|
|
@ -56,14 +56,12 @@ void dms_all_sent(nmt_t *nmt)
|
|||
}
|
||||
|
||||
/* receive bits from DMS */
|
||||
int fsk_render_frame(nmt_t *nmt, const char *frame, int length, sample_t *sample)
|
||||
void test_dms_frame(const char *frame, int length)
|
||||
{
|
||||
printf("(getting %d bits from DMS layer)\n", length);
|
||||
|
||||
memcpy(current_bits, frame, length);
|
||||
current_bit_count = length;
|
||||
|
||||
return nmt->fsk_samples_per_bit * length;
|
||||
}
|
||||
|
||||
nmt_t *alloc_nmt(void)
|
||||
|
@ -71,9 +69,11 @@ nmt_t *alloc_nmt(void)
|
|||
nmt_t *nmt;
|
||||
|
||||
nmt = calloc(sizeof(*nmt), 1);
|
||||
nmt->sender.samplerate = 40 * 1200;
|
||||
dms_init_sender(nmt);
|
||||
nmt->fsk_samples_per_bit = 40;
|
||||
nmt->dms.frame_size = nmt->fsk_samples_per_bit * 127 + 10;
|
||||
ffsk_global_init(1.0);
|
||||
ffsk_init(&nmt->ffsk, nmt, NULL, 1, nmt->sender.samplerate);
|
||||
nmt->dms.frame_size = nmt->ffsk.samples_per_bit * 127 + 10;
|
||||
nmt->dms.frame_spl = calloc(nmt->dms.frame_size, sizeof(nmt->dms.frame_spl[0]));
|
||||
|
||||
dms_reset(nmt);
|
||||
|
|
Loading…
Reference in New Issue