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/* FFSK audio processing (NMT / Radiocom 2000)
*
* (C) 2017 by Andreas Eversberg <jolly@eversberg.eu>
* All Rights Reserved
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
#define CHAN ffsk->channel
#include <stdio.h>
#include <stdint.h>
#include <stdlib.h>
#include <string.h>
#include <errno.h>
#include <math.h>
#include "../common/sample.h"
#include "../common/debug.h"
#include "ffsk.h"
#define PI M_PI
#define BIT_RATE 1200 /* baud rate */
#define FILTER_STEPS 0.1 /* step every 1/12000 sec */
/* two signaling tones */
static double ffsk_freq[2] = {
1800.0,
1200.0,
};
static sample_t dsp_tone_bit[2][2][65536]; /* polarity, bit, phase */
/* global init for FFSK */
void ffsk_global_init(double peak_fsk)
{
int i;
double s;
PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table for FFSK tones.\n");
for (i = 0; i < 65536; i++) {
s = sin((double)i / 65536.0 * 2.0 * PI);
/* bit(1) 1 cycle */
dsp_tone_bit[0][1][i] = s * peak_fsk;
dsp_tone_bit[1][1][i] = -s * peak_fsk;
/* bit(0) 1.5 cycles */
s = sin((double)i / 65536.0 * 3.0 * PI);
dsp_tone_bit[0][0][i] = s * peak_fsk;
dsp_tone_bit[1][0][i] = -s * peak_fsk;
}
}
/* Init FFSK */
int ffsk_init(ffsk_t *ffsk, void *inst, void (*receive_bit)(void *inst, int bit, double quality, double level), int channel, int samplerate)
{
sample_t *spl;
int i;
/* a symbol rate of 1200 Hz, times check interval of FILTER_STEPS */
if (samplerate < (double)BIT_RATE / (double)FILTER_STEPS) {
PDEBUG(DDSP, DEBUG_ERROR, "Sample rate must be at least 12000 Hz to process FSK+supervisory signal.\n");
return -EINVAL;
}
memset(ffsk, 0, sizeof(*ffsk));
ffsk->inst = inst;
ffsk->receive_bit = receive_bit;
ffsk->channel = channel;
ffsk->samplerate = samplerate;
ffsk->samples_per_bit = (double)ffsk->samplerate / (double)BIT_RATE;
ffsk->bits_per_sample = 1.0 / ffsk->samples_per_bit;
PDEBUG(DDSP, DEBUG_DEBUG, "Use %.4f samples for full bit duration @ %d.\n", ffsk->samples_per_bit, ffsk->samplerate);
/* allocate ring buffers, one bit duration */
ffsk->filter_size = floor(ffsk->samples_per_bit); /* buffer holds one bit (rounded down) */
spl = calloc(1, ffsk->filter_size * sizeof(*spl));
if (!spl) {
PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
ffsk_cleanup(ffsk);
return -ENOMEM;
}
ffsk->filter_spl = spl;
ffsk->filter_bit = -1;
/* count symbols */
for (i = 0; i < 2; i++)
audio_goertzel_init(&ffsk->goertzel[i], ffsk_freq[i], ffsk->samplerate);
ffsk->phaseshift65536 = 65536.0 / ffsk->samples_per_bit;
PDEBUG(DDSP, DEBUG_DEBUG, "fsk_phaseshift = %.4f\n", ffsk->phaseshift65536);
return 0;
}
/* Cleanup transceiver instance. */
void ffsk_cleanup(ffsk_t *ffsk)
{
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Cleanup DSP for Transceiver.\n");
if (ffsk->filter_spl) {
free(ffsk->filter_spl);
ffsk->filter_spl = NULL;
}
}
//#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 ffsk_decode_step(ffsk_t *ffsk, int pos)
{
double level, result[2], softbit, quality;
int max;
sample_t *spl;
int bit;
max = ffsk->filter_size;
spl = ffsk->filter_spl;
level = audio_level(spl, max);
/* limit level to prevent division by zero */
if (level < 0.001)
level = 0.001;
audio_goertzel(ffsk->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 (ffsk->filter_bit != bit) {
/* If we have a bit change, move sample counter towards one half bit duration.
* We may have noise, so the bit change may be wrong or not at the correct place.
* This can cause bit slips.
* Therefore we change the sample counter only slightly, so bit slips may not
* happen so quickly.
* */
#ifdef DEBUG_FILTER
puts("bit change");
#endif
ffsk->filter_bit = bit;
if (ffsk->filter_sample < 5)
ffsk->filter_sample++;
if (ffsk->filter_sample > 5)
ffsk->filter_sample--;
} else if (--ffsk->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% */
ffsk->receive_bit(ffsk->inst, bit, quality, level / 0.63662);
ffsk->filter_sample = 10;
}
}
void ffsk_receive(ffsk_t *ffsk, sample_t *sample, int length)
{
sample_t *spl;
int max, pos;
double step, bps;
int i;
/* write received samples to decode buffer */
max = ffsk->filter_size;
pos = ffsk->filter_pos;
step = ffsk->filter_step;
bps = ffsk->bits_per_sample;
spl = ffsk->filter_spl;
for (i = 0; i < length; i++) {
#ifdef DEBUG_MODULATOR
printf("|%s|\n", debug_amplitude((double)samples[i] / 2333.0 /*fsk peak*/ / 2.0));
#endif
/* write into ring buffer */
spl[pos++] = sample[i];
if (pos == max)
pos = 0;
/* if 1/10th of a bit duration is reached, decode buffer */
step += bps;
if (step >= FILTER_STEPS) {
step -= FILTER_STEPS;
ffsk_decode_step(ffsk, pos);
}
}
ffsk->filter_step = step;
ffsk->filter_pos = pos;
}
/* render frame */
int ffsk_render_frame(ffsk_t *ffsk, const char *frame, int length, sample_t *sample)
{
int bit, polarity;
double phaseshift, phase;
int count = 0, i;
polarity = ffsk->polarity;
phaseshift = ffsk->phaseshift65536;
phase = ffsk->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;
}
ffsk->phase65536 = phase;
ffsk->polarity = polarity;
/* return number of samples created for frame */
return count;
}
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