An implementation of Analog cellular networks like A-Netz, B-Netz, C-Netz, NMT, AMPS, TACS, JTACS, Radiocom 2000, IMTS, MPT1327, Eurosignal and more http://osmocom-analog.eversberg.eu/
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osmocom-analog/src/cnetz/fsk_demod.c

665 lines
22 KiB

/* FSK decoder of carrier FSK signals received by simple FM receiver
*
* (C) 2016 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/>.
*/
/* How does it work:
* -----------------
*
* C-Netz modulates the carrier frequency. If it is 2.4 kHz above, it is high
* level, if it is 2.4 kHz below, it is low level. Look at FTZ 171 TR 60
* Chapter 5 (data exchange) for closer information.
*
* Detecting level change (from SDR):
*
* Whenever we cross zero, we detect a level change. Also we know the level
* of the bit then. If we don't get another level change within 1.5 of bit
* duration, we will sample the next bit with the current level. From then
* we will sample the next bit 1.0 bit duration later, if there is still no
* level change. If we get another level change, we take that bit and wait
* 1.5 bit duration for next change...
*
* Detect level change (from analog radio):
*
* We don't just look for high/low level, because we don't know what the actual
* 0-level of the phone's transmitter is. (level of carrier frequency) Also we
* use receiver and sound card that cause any level to return to 0 after some
* time, Even if the transmitter still transmits a level above or below the
* carrier frequnecy. Insted we look at the change of the received signal. An
* upward change indicates 1. An downward change indicates 0. (This may also be
* reversed, if we find out, that we received a sync sequence in reversed
* polarity.) If there is no significant change in level, we keep the value of
* last change, regardless of what level we actually receive.
*
* To determine a change from noise, we use a theshold. This is set to half of
* the level of last received change. This means that the next change may be
* down to a half lower. There is a special case during distributed signaling.
* The first level change of each data chunk raises or falls from 0-level
* (unmodulated carrier), so the threshold for this bit is only a quarter of the
* last received change.
*
* While searching for a sync sequence, the threshold for the next change is set
* after each change. After synchronization, the the threshold is locked to half
* of the average change level of the sync sequence.
*
* Search window
*
* We use a window of one bit length (9 samples at 48 kHz sample rate) and look
* for a change that is higher than the threshold and has its highest slope in
* the middle of the window. To determine the level, the min and max value
* inside the window is searched. The differece is the change level. To
* determine the highest slope, the highest difference between subsequent
* samples is used. For every sample we move the window one bit to the right
* (next sample), check if change level matches the threshold and highest slope
* is in the middle and so forth. Only if the highes slope is exactly in the
* middle, we declare a change. This means that we detect a slope about half of
* a bit duration later.
*
* When we are not synced:
*
* For every change we record a bit. A positive change is 1 and a negative 0. If
* it turns out that the receiver or sound card is reversed, we reverse bits.
* After every change we wait up to 1.5 bit duration for next change. If there
* is a change, we record our next bit. If there is no change, we record the
* state of the last bit. After we had no change, we wait 1 bit duration, since
* we already 0.5 behind the start of the recently recorded bit.
*
* When we are synced:
*
* After we recorded the time of all level changes during the sync sequence, we
* calculate an average and use it as a time base for sampling the subsequent 150
* bit of a message. From now on, a bit change does not cause any resync. We
* just remember what change we received. Later we use it for sampling the 150
* bits.
*
* We wait a duration of 1.5 bits after the sync sequence and the start of the
* bit that follows the sync sequence. We record what we received as last
* change. For all following 149 bits we wait 1 bit duration and record what we
* received as last change.
*
* Sync clock
*
* Because we transmit and receive chunks of sample from buffers of different
* drivers, we cannot determine the exact latency between received and
* transmitted samples. Also some sound cards may have different RX and TX
* speed. One (pure software) solution is to sync ourself to the mobile phone,
* since the mobile phone is perfectly synced to us.
*
* After receiving and decoding of a frame, we use the time of received sync
* sequence to synchronize the receiver to the mobile phone. If we receive a
* message on the OgK (control channel), we know that this is a response to a
* message of a specific time slot we recently sent. Then we can fully sync the
* receiver's clock. For any other frame, we cannot determine the absolute
* clock. We just correct the receiver's clock, as the clock differs only
* slightly from the time the message was received.
*
*/
/* Words on debugging:
*
* A debug file can be written. It will show the current sample that is right
* in the middle of the search window. Additional information is shown right
* of the sample graph.
*/
/* use to debug decoder
* if debug is set to 0, debugging will start from SPK_V signalling,
* if debug is set to 1, debugging will start at program start
*/
//#define DEBUG_DECODER
//static int debug = 0;
#include <stdio.h>
#include <stdint.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include "../libsample/sample.h"
#include "../libdebug/debug.h"
#include "cnetz.h"
#include "dsp.h"
#include "telegramm.h"
int fsk_fm_init(fsk_fm_demod_t *fsk, cnetz_t *cnetz, int samplerate, double bitrate, enum demod_type demod)
{
int len, half;
memset(fsk, 0, sizeof(*fsk));
if (samplerate < 48000) {
PDEBUG(DDSP, DEBUG_ERROR, "Sample rate must be at least 48000 Hz!\n");
return -1;
}
fsk->cnetz = cnetz;
fsk->demod_type = demod;
switch (demod) {
case FSK_DEMOD_SLOPE:
PDEBUG(DDSP, DEBUG_INFO, "Detecting level change by looking at slope (good for sound cards)\n");
break;
case FSK_DEMOD_LEVEL:
PDEBUG(DDSP, DEBUG_INFO, "Detecting level change by looking zero crosssing (good for SDR)\n");
break;
default:
PDEBUG(DDSP, DEBUG_ERROR, "Wrong demod type, please fix!\n");
abort();
}
len = (int)((double)samplerate / bitrate + 0.5);
half = (int)((double)samplerate / bitrate / 2.0 + 0.5);
fsk->bit_buffer_spl = calloc(sizeof(fsk->bit_buffer_spl[0]), len);
if (!fsk->bit_buffer_spl) {
PDEBUG(DDSP, DEBUG_ERROR, "No mem!\n");
goto error;
}
fsk->bit_buffer_len = len;
fsk->bit_buffer_half = half;
fsk->bits_per_sample = bitrate / (double)samplerate;
fsk->speech_size = samplerate * 60 / bitrate + 10; /* 60 bits duration, add 10 to be safe */
fsk->speech_buffer = calloc(sizeof(fsk->speech_buffer[0]), fsk->speech_size);
if (!fsk->speech_buffer) {
PDEBUG(DDSP, DEBUG_ERROR, "No mem!\n");
goto error;
}
fsk->level_threshold = 0.1;
#ifdef DEBUG_DECODER
char debug_filename[256];
sprintf(debug_filename, "/tmp/debug_decoder_channel_%d.txt", cnetz->sender.kanal);
fsk->debug_fp = fopen(debug_filename, "w");
if (!fsk->debug_fp) {
fprintf(stderr, "Failed to open decoder debug file '%s'!\n", debug_filename);
exit(0);
} else
printf("**** Writing decoder debug file '%s' ****\n", debug_filename);
#endif
fsk->dmp_frame_level = display_measurements_add(&cnetz->sender.dispmeas, "Frame Level", "%.1f %% (last)", DISPLAY_MEAS_LAST, DISPLAY_MEAS_LEFT, 0.0, 150.0, 100.0);
fsk->dmp_frame_stddev = display_measurements_add(&cnetz->sender.dispmeas, "Frame Stddev", "%.1f %% (last)", DISPLAY_MEAS_LAST, DISPLAY_MEAS_LEFT, 0.0, 100.0, 100.0);
return 0;
error:
fsk_fm_exit(fsk);
return -1;
}
void fsk_fm_exit(fsk_fm_demod_t *fsk)
{
if (fsk->bit_buffer_spl) {
free(fsk->bit_buffer_spl);
fsk->bit_buffer_spl = NULL;
}
if (fsk->speech_buffer) {
free(fsk->speech_buffer);
fsk->speech_buffer = NULL;
}
#ifdef DEBUG_DECODER
if (fsk->debug_fp) {
fclose(fsk->debug_fp);
fsk->debug_fp = NULL;
}
#endif
}
/* get levels, sync time and jitter/stddev from sync sequence or frame data */
static inline void get_levels(fsk_fm_demod_t *fsk, double *_min, double *_max, double *_avg, int *_probes, int num, double *_time, double *_jitter, double *_stddev)
{
int count = 0;
double min = 0, max = 0, avg = 0, level;
double time = 0, t, sync_average, sync_time, jitter = 0.0, stddev = 0.0;
int bit_offset;
int i;
/* get levels an the average receive time */
for (i = 0; i < num; i++) {
level = fsk->change_levels[(fsk->change_pos - 1 - i) & 0xff];
if (level <= 0.0)
continue;
/* in spk mode, we skip the voice part (62 bits) */
if (fsk->cnetz->dsp_mode == DSP_MODE_SPK_V)
bit_offset = i + ((i + 2) >> 2) * 62;
else
bit_offset = i;
t = fmod(fsk->change_when[(fsk->change_pos - 1 - i) & 0xff] - fsk->bit_time + (double)bit_offset + BITS_PER_SUPERFRAME, BITS_PER_SUPERFRAME);
if (t > BITS_PER_SUPERFRAME / 2)
t -= BITS_PER_SUPERFRAME;
//if (fsk->cnetz->dsp_mode == DSP_MODE_SPK_V)
// printf("%d: level=%.0f%% @%.2f difference=%.2f\n", bit_offset, level * 100, fsk->change_when[(fsk->change_pos - 1 - i) & 0xff], t);
time += t;
if (i == 0 || level < min)
min = level;
if (i == 0 || level > max)
max = level;
avg += level;
count++;
}
avg /= (double)count;
time /= (double)count;
/* should never happen */
if (!count) {
*_min = *_max = *_avg = 0.0;
return;
}
/* when did we received the sync?
* sync_average is the average about how early (negative) or
* late (positive) we received the sync relative to current bit_time.
* sync_time is the absolute time within the super frame.
*/
sync_average = time;
sync_time = fmod(sync_average + fsk->bit_time + BITS_PER_SUPERFRAME, BITS_PER_SUPERFRAME);
*_probes = count;
*_min = min;
*_max = max;
*_avg = avg;
if (_time) {
// if (fsk->cnetz->dsp_mode == DSP_MODE_SPK_V)
// printf("sync at distributed mode\n");
// printf("sync at bit_time=%.2f (sync_average = %.2f)\n", sync_time, sync_average);
/* if our average sync is later (greater) than the current
* bit_time, we must wait longer (next_bit above 1.5)
* for the time to sample the bit.
* if sync is earlier, bit_time is already too late, so
* we must wait less than 1.5 bits */
fsk->next_bit = 1.5 + sync_average;
*_time = sync_time;
}
if (_jitter) {
/* get jitter of received changes */
for (i = 0; i < num; i++) {
level = fsk->change_levels[(fsk->change_pos - 1 - i) & 0xff];
if (level <= 0.0)
continue;
/* in spk mode, we skip the voice part (62 bits) */
if (fsk->cnetz->dsp_mode == DSP_MODE_SPK_V)
bit_offset = i + ((i + 2) >> 2) * 62;
else
bit_offset = i;
t = fmod(fsk->change_when[(fsk->change_pos - 1 - i) & 0xff] - sync_time + (double)bit_offset + BITS_PER_SUPERFRAME, BITS_PER_SUPERFRAME);
if (t > BITS_PER_SUPERFRAME / 2)
t = BITS_PER_SUPERFRAME - t; /* turn negative into positive */
jitter += t;
}
*_jitter = jitter / (double)count;
}
if (_stddev) {
/* get standard deviation of level */
for (i = 0; i < num; i++) {
level = fsk->change_levels[(fsk->change_pos - 1 - i) & 0xff];
if (level <= 0.0)
continue;
stddev += (level - avg) * (level - avg);
}
*_stddev = sqrt(stddev / (double)count);
}
}
static inline void got_bit(fsk_fm_demod_t *fsk, int bit, double change_level)
{
int probes;
double min, max, avg;
/* count bits, but do not exceed 4 bits per SPK block */
if (fsk->cnetz->dsp_mode == DSP_MODE_SPK_V) {
/* for first bit, we have only half of the modulation deviation, so we multiply level by two */
if (fsk->bit_count == 0)
change_level *= 2.0;
if (fsk->bit_count >= 4)
return;
}
fsk->bit_count++;
fsk->change_levels[fsk->change_pos] = change_level;
fsk->change_when[fsk->change_pos++] = fsk->bit_time;
switch (fsk->sync) {
case FSK_SYNC_NONE:
fsk->rx_sync = (fsk->rx_sync << 1) | bit;
/* use half level of last change for threshold change detection.
* if there is no change detected for 5 bits, set theshold to
* 1 percent, so the 7 pause bits before a frame will make sure
* that the change is below noise level, so the first sync
* bit is detected. then the change is set and adjusted
* for all other bits in the sync sequence.
* after sync, the theshold is set to half of the average of
* all changes in the sync sequence */
if (change_level > 0.0) {
fsk->level_threshold = change_level / 2.0;
} else if ((fsk->rx_sync & 0x1f) == 0x00 || (fsk->rx_sync & 0x1f) == 0x1f) {
if (fsk->cnetz->dsp_mode != DSP_MODE_SPK_V)
fsk->level_threshold = 0.01;
}
if (detect_sync(fsk->rx_sync)) {
fsk->sync = FSK_SYNC_POSITIVE;
got_sync:
#ifdef DEBUG_DECODER
if (debug)
fprintf(fsk->debug_fp, " SYNC!");
#endif
get_levels(fsk, &min, &max, &avg, &probes, 30, &fsk->sync_time, NULL, &fsk->sync_stddev);
fsk->sync_level = avg;
if (fsk->sync == FSK_SYNC_NEGATIVE)
fsk->sync_level = -fsk->sync_level;
// printf("sync (change min=%.0f%% max=%.0f%% avg=%.0f%% sync_time=%.2f stddev=%.0f%% probes=%d)\n", min * 100, max * 100, avg * 100, fsk->sync_time, fsk->sync_stddev / avg, probes);
fsk->level_threshold = (double)avg;
fsk->rx_sync = 0;
fsk->rx_buffer_count = 0;
break;
}
if (detect_sync(fsk->rx_sync ^ 0xfffffffff)) {
fsk->sync = FSK_SYNC_NEGATIVE;
goto got_sync;
}
break;
case FSK_SYNC_NEGATIVE:
bit = 1 - bit;
/* FALLTHRU */
case FSK_SYNC_POSITIVE:
fsk->rx_buffer[fsk->rx_buffer_count] = bit + '0';
if (++fsk->rx_buffer_count == 150) {
fsk->sync = FSK_SYNC_NONE;
#ifdef DEBUG_DECODER
if (debug)
fprintf(fsk->debug_fp, " FRAME DONE!");
#endif
if (fsk->cnetz->dsp_mode != DSP_MODE_SPK_V) {
/* received 40 bits after start of block */
fsk->sync_time = fmod(fsk->sync_time - (7+33) + BITS_PER_SUPERFRAME, BITS_PER_SUPERFRAME);
} else {
/* received 662 bits after start of block (10 SPK blocks + 1 bit (== 2 level changes)) */
fsk->sync_time = fmod(fsk->sync_time - (66*10+2) + BITS_PER_SUPERFRAME, BITS_PER_SUPERFRAME);
}
/* update measurements */
display_measurements_update(fsk->dmp_frame_level, fabs(fsk->sync_level) / fsk->cnetz->fsk_deviation * 100.0, 0.0);
display_measurements_update(fsk->dmp_frame_stddev, fsk->sync_stddev / fabs(fsk->sync_level) * 100.0, 0.0);
/* receive frame */
cnetz_decode_telegramm(fsk->cnetz, fsk->rx_buffer, fsk->sync_level, fsk->sync_time, fsk->sync_stddev);
}
break;
}
}
/* find bit change by checking slope within a window */
static inline void find_change_slope(fsk_fm_demod_t *fsk)
{
sample_t level_min = 0, level_max = 0, change_max = -1;
int change_at = -1, change_positive = -1;
sample_t s, last_s = 0;
sample_t threshold;
int i;
#ifdef DEBUG_DECODER
/* show deviation of middle sample in windows (in a range of bandwidth) */
if (debug) {
fprintf(fsk->debug_fp, "%s",
debug_amplitude(
fsk->bit_buffer_spl[(fsk->bit_buffer_pos + fsk->bit_buffer_half) % fsk->bit_buffer_len]
)
);
}
#endif
/* get level range (level_min and level_max) and also
* get maximum slope (change_max) and where it was
* (change_at) and what direction it went (change_positive)
*/
for (i = 0; i < fsk->bit_buffer_len; i++) {
last_s = s;
s = fsk->bit_buffer_spl[fsk->bit_buffer_pos++];
if (fsk->bit_buffer_pos == fsk->bit_buffer_len)
fsk->bit_buffer_pos = 0;
if (i > 0) {
if (s - last_s > change_max) {
change_max = s - last_s;
change_at = i;
change_positive = 1;
} else if (last_s - s > change_max) {
change_max = last_s - s;
change_at = i;
change_positive = 0;
}
}
if (i == 0 || s > level_max)
level_max = s;
if (i == 0 || s < level_min)
level_min = s;
}
/* for first bit, we have only half of the modulation deviation, so we divide the threshold by two */
if (fsk->cnetz->dsp_mode == DSP_MODE_SPK_V && fsk->bit_count == 0)
threshold = fsk->level_threshold / 2.0;
else
threshold = fsk->level_threshold;
/* if we are not in sync, for every detected change we set
* next_bit to 1.5, so we wait 1.5 bits for next change
* if it is not received within this time, there is no change,
* so the bit does not change.
* if we are in sync, we remember last change. after 1.5
* bits after sync average, we measure the first bit
* and then all subsequent bits after 1.0 bits */
if (level_max - level_min > threshold && change_at == fsk->bit_buffer_half) {
#ifdef DEBUG_DECODER
if (debug) {
fprintf(fsk->debug_fp, " CHANGE %d->%d (level=%.3f, threshold=%.3f)",
fsk->last_change_positive,
change_positive,
level_max - level_min,
threshold);
}
#endif
fsk->last_change_positive = change_positive;
if (!fsk->sync) {
fsk->next_bit = 1.5;
got_bit(fsk, change_positive, (level_max - level_min) / 2.0);
}
}
if (fsk->next_bit <= 0.0) {
#ifdef DEBUG_DECODER
if (debug)
fprintf(fsk->debug_fp, " SAMPLING %d", fsk->last_change_positive);
#endif
fsk->next_bit += 1.0;
#ifdef DEBUG_DECODER
if (debug && fsk->cnetz->dsp_mode == DSP_MODE_SPK_V && fsk->bit_count >= 4)
fprintf(fsk->debug_fp, " (ignoring)");
#endif
got_bit(fsk, fsk->last_change_positive, 0.0);
}
fsk->next_bit -= fsk->bits_per_sample;
#ifdef DEBUG_DECODER
if (debug)
fprintf(fsk->debug_fp, "\n");
#endif
}
/* find bit change by looking at zero crossing */
static inline void find_change_level(fsk_fm_demod_t *fsk)
{
int change_positive = -1;
sample_t s;
/* get bit in the middle of the buffer */
s = fsk->bit_buffer_spl[(fsk->bit_buffer_pos + fsk->bit_buffer_half) % fsk->bit_buffer_len];
#ifdef DEBUG_DECODER
/* show deviation */
if (debug)
fprintf(fsk->debug_fp, "%s", debug_amplitude(s));
#endif
/* just sample first bit in distributed mode */
if (fsk->cnetz->dsp_mode == DSP_MODE_SPK_V && fsk->bit_count == 0) {
if (fmod(fsk->bit_time, BITS_PER_SPK_BLOCK) < 1.5)
goto done;
#ifdef DEBUG_DECODER
if (debug)
fprintf(fsk->debug_fp, " (First bit of data chunk)");
#endif
/* use current level for first bit to sample */
fsk->last_change_positive = (s > 0);
fsk->next_bit = 0.0;
} else {
/* see if we have a level change */
if (!fsk->last_change_positive && s > 0)
change_positive = 1;
if (fsk->last_change_positive && s < 0)
change_positive = 0;
}
/* if we are not in sync, for every detected change we set
* next_bit to 1.5, so we wait 1.5 bits for next change
* if it is not received within this time, there is no change,
* so the bit does not change.
* if we are in sync, we remember last change. after 1.5
* bits after sync average, we measure the first bit
* and then all subsequent bits after 1.0 bits */
if (change_positive >= 0) {
#ifdef DEBUG_DECODER
if (debug)
fprintf(fsk->debug_fp, " CHANGE %d->%d", fsk->last_change_positive, change_positive);
#endif
fsk->last_change_positive = change_positive;
if (!fsk->sync) {
fsk->next_bit = 1.5;
/* if bit change is inside window, we can get level from borders of window */
s = fsk->bit_buffer_spl[fsk->bit_buffer_pos];
s -= fsk->bit_buffer_spl[(fsk->bit_buffer_pos + fsk->bit_buffer_len - 1) % fsk->bit_buffer_len];
got_bit(fsk, change_positive, fabs(s / 2.0));
}
}
if (fsk->next_bit <= 0.0) {
#ifdef DEBUG_DECODER
if (debug)
fprintf(fsk->debug_fp, " SAMPLING %d", fsk->last_change_positive);
#endif
fsk->next_bit += 1.0;
#ifdef DEBUG_DECODER
if (debug && fsk->cnetz->dsp_mode == DSP_MODE_SPK_V && fsk->bit_count >= 4)
fprintf(fsk->debug_fp, " (ignoring)");
#endif
got_bit(fsk, fsk->last_change_positive, 0.0);
}
fsk->next_bit -= fsk->bits_per_sample;
done:
#ifdef DEBUG_DECODER
if (debug)
fprintf(fsk->debug_fp, "\n");
#endif
return;
}
/* receive FM signal from receiver */
void fsk_fm_demod(fsk_fm_demod_t *fsk, sample_t *samples, int length)
{
int i;
double t;
/* process signaling block, sample by sample */
for (i = 0; i < length; i++) {
fsk->bit_buffer_spl[fsk->bit_buffer_pos++] = samples[i];
if (fsk->bit_buffer_pos == fsk->bit_buffer_len)
fsk->bit_buffer_pos = 0;
/* for each sample process buffer */
if (fsk->cnetz->dsp_mode != DSP_MODE_SPK_V) {
if (fsk->demod_type == FSK_DEMOD_SLOPE)
find_change_slope(fsk);
else
find_change_level(fsk);
} else {
#ifdef DEBUG_DECODER
/* start debugging */
debug = 1;
#endif
/* in distributed signaling, measure over 5 bits, but ignore 5th bit.
* also reset next_bit, as soon as we reach the window */
/* note that we start from 0.5, because we detect change 0.5 bits later,
* because the detector of the change is in the middle of the 1 bit
* search window */
t = fmod(fsk->bit_time, BITS_PER_SPK_BLOCK);
if (t < 0.5) {
fsk->next_bit = 1.0 - fsk->bits_per_sample;
#ifdef DEBUG_DECODER
if (debug && fsk->bit_count)
fprintf(fsk->debug_fp, "---- SPK(V) BLOCK START ----\n");
#endif
fsk->bit_count = 0;
} else
if (t >= 0.5 && t < 5.5) {
if (fsk->demod_type == FSK_DEMOD_SLOPE)
find_change_slope(fsk);
else
find_change_level(fsk);
} else
if (t >= 5.5 && t < 65.5) {
/* get audio for the duration of 60 bits */
/* prevent overflow, if speech_size != 0 and SPK_V
* has been restarted. */
if (fsk->speech_count < fsk->speech_size)
fsk->speech_buffer[fsk->speech_count++] = samples[i];
} else
if (t >= 65.5) {
if (fsk->speech_count) {
unshrink_speech(fsk->cnetz, fsk->speech_buffer, fsk->speech_count);
fsk->speech_count = 0;
}
}
}
fsk->bit_time += fsk->bits_per_sample;
if (fsk->bit_time >= BITS_PER_SUPERFRAME) {
fsk->bit_time -= BITS_PER_SUPERFRAME;
}
/* another clock is used to measure actual super frame time */
fsk->bit_time_uncorrected += fsk->bits_per_sample;
if (fsk->bit_time_uncorrected >= BITS_PER_SUPERFRAME) {
fsk->bit_time_uncorrected -= BITS_PER_SUPERFRAME;
calc_clock_speed(fsk->cnetz, (double)fsk->cnetz->sender.samplerate * 2.4, 0, 1);
}
}
}
void fsk_correct_sync(fsk_fm_demod_t *fsk, double offset)
{
fsk->bit_time = fmod(fsk->bit_time - offset + BITS_PER_SUPERFRAME, BITS_PER_SUPERFRAME);
}
/* copy sync from one instance to another (used to sync RX of SpK to OgK */
void fsk_copy_sync(fsk_fm_demod_t *fsk_to, fsk_fm_demod_t *fsk_from)
{
fsk_to->bit_time = fsk_from->bit_time;
}
void fsk_demod_reset(fsk_fm_demod_t *fsk)
{
fsk->sync = FSK_SYNC_NONE;
}