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osmocom-analog/src/cnetz/dsp.c

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/* C-Netz audio processing
*
* (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/>.
*/
#define CHAN cnetz->sender.kanal
#include <stdio.h>
#include <stdint.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include <errno.h>
#include "../libsample/sample.h"
#include "../libdebug/debug.h"
#include "../libmobile/call.h"
#include "cnetz.h"
#include "sysinfo.h"
#include "telegramm.h"
#include "dsp.h"
/* test function to mirror received audio from ratio back to radio */
//#define TEST_SCRAMBLE
/* test the audio quality after cascading two scramblers (TEST_SCRAMBLE must be defined) */
//#define TEST_UNSCRAMBLE
#define PI M_PI
#define MAX_DEVIATION 4000.0
#define MAX_MODULATION 3000.0
#define DBM0_DEVIATION 4000.0 /* deviation of dBm0 at 1 kHz */
#define COMPANDOR_0DB 1.0 /* A level of 0dBm (1.0) shall be unaccected */
#define FSK_DEVIATION (2500.0 / DBM0_DEVIATION) /* no emphasis */
#define MAX_DISPLAY 1.4 /* something above dBm0, no emphasis */
#define BITRATE 5280.0 /* bits per second */
#define BLOCK_BITS 198 /* duration of one time slot including pause at beginning and end */
#define CUT_OFF_OFFSET 300.0 /* cut off frequency for offset filter (level correction between subsequent audio chunks) */
#ifdef TEST_SCRAMBLE
jitter_t scrambler_test_jb;
scrambler_t scrambler_test_scrambler1;
scrambler_t scrambler_test_scrambler2;
#endif
static sample_t ramp_up[256], ramp_down[256];
void dsp_init(void)
{
}
static void dsp_init_ramp(cnetz_t *cnetz)
{
double c;
int i;
PDEBUG(DDSP, DEBUG_DEBUG, "Generating smooth ramp table.\n");
for (i = 0; i < 256; i++) {
c = cos((double)i / 256.0 * PI);
/* use square-root of cosine ramp. tests showed that phones are more
* happy with that. */
if (c < 0)
c = -sqrt(-c);
else
c = sqrt(c);
ramp_down[i] = c * (double)cnetz->fsk_deviation;
ramp_up[i] = -ramp_down[i];
}
}
/* Init transceiver instance. */
int dsp_init_sender(cnetz_t *cnetz, int measure_speed, double clock_speed[2], enum demod_type demod)
{
int rc = 0;
double size;
double RC, dt;
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Init FSK for 'Sender'.\n");
/* set modulation parameters */
sender_set_fm(&cnetz->sender, MAX_DEVIATION, MAX_MODULATION, DBM0_DEVIATION, MAX_DISPLAY);
if (measure_speed) {
cnetz->measure_speed = measure_speed;
cant_recover = 1;
}
if (clock_speed[0] > 1000 || clock_speed[0] < -1000 || clock_speed[1] > 1000 || clock_speed[1] < -1000) {
PDEBUG(DDSP, DEBUG_ERROR, "Clock speed %.1f,%.1f ppm out of range! Plese use range between +-1000 ppm!\n", clock_speed[0], clock_speed[1]);
return -EINVAL;
}
PDEBUG(DDSP, DEBUG_INFO, "Using clock speed of %.1f ppm (RX) and %.1f ppm (TX) to correct sound card's clock.\n", clock_speed[0], clock_speed[1]);
cnetz->fsk_bitduration = (double)cnetz->sender.samplerate / ((double)BITRATE / (1.0 + clock_speed[1] / 1000000.0));
cnetz->fsk_tx_bitstep = 1.0 / cnetz->fsk_bitduration;
PDEBUG(DDSP, DEBUG_DEBUG, "Use %.4f samples for one bit duration @ %d.\n", cnetz->fsk_bitduration, cnetz->sender.samplerate);
size = cnetz->fsk_bitduration * (double)BLOCK_BITS * 16.0; /* 16 blocks for distributed frames */
cnetz->fsk_tx_buffer_size = size * 1.1; /* more to compensate clock speed */
cnetz->fsk_tx_buffer = calloc(sizeof(sample_t), cnetz->fsk_tx_buffer_size);
if (!cnetz->fsk_tx_buffer) {
PDEBUG(DDSP, DEBUG_DEBUG, "No memory!\n");
rc = -ENOMEM;
goto error;
}
/* create devation and ramp */
cnetz->fsk_deviation = FSK_DEVIATION;
dsp_init_ramp(cnetz);
/* init low pass filter for received signal */
iir_lowpass_init(&cnetz->lp, MAX_MODULATION, cnetz->sender.samplerate, 2);
/* create speech buffer */
cnetz->dsp_speech_buffer = calloc(sizeof(sample_t), (int)(cnetz->fsk_bitduration * 70.0)); /* more to compensate clock speed. we just need it to fill 62 bits (60 bits, including pause bits). */
if (!cnetz->dsp_speech_buffer) {
PDEBUG(DDSP, DEBUG_DEBUG, "No memory!\n");
rc = -ENOMEM;
goto error;
}
/* reinit the sample rate to shrink/expand audio */
init_samplerate(&cnetz->sender.srstate, 8000.0, (double)cnetz->sender.samplerate / 1.1, 3300.0); /* 66 <-> 60 */
rc = fsk_fm_init(&cnetz->fsk_demod, cnetz, cnetz->sender.samplerate, (double)BITRATE / (1.0 + clock_speed[0] / 1000000.0), demod);
if (rc < 0)
goto error;
/* init scrambler for shrinked audio */
scrambler_setup(&cnetz->scrambler_tx, (double)cnetz->sender.samplerate / 1.1);
scrambler_setup(&cnetz->scrambler_rx, (double)cnetz->sender.samplerate / 1.1);
/* reinit jitter buffer for 8000 kHz */
jitter_destroy(&cnetz->sender.dejitter);
rc = jitter_create(&cnetz->sender.dejitter, 8000 / 5);
if (rc < 0)
goto error;
/* init compandor, according to C-Netz specs, attack and recovery time
* shall not exceed according to ITU G.162 */
init_compandor(&cnetz->cstate, 8000, 5.0, 22.5, COMPANDOR_0DB);
/* use this filter to compensate level changes between two subsequent audio chunks */
RC = 1.0 / (CUT_OFF_OFFSET * 2.0 *3.14);
dt = 1.0 / cnetz->sender.samplerate;
cnetz->offset_factor = RC / (RC + dt);
#ifdef TEST_SCRAMBLE
rc = jitter_create(&scrambler_test_jb, cnetz->sender.samplerate / 5);
if (rc < 0) {
PDEBUG(DDSP, DEBUG_ERROR, "Failed to init jitter buffer for scrambler test!\n");
exit(0);
}
scrambler_setup(&scrambler_test_scrambler1, cnetz->sender.samplerate);
scrambler_setup(&scrambler_test_scrambler2, cnetz->sender.samplerate);
#endif
return 0;
error:
dsp_cleanup_sender(cnetz);
return rc;
}
void dsp_cleanup_sender(cnetz_t *cnetz)
{
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Cleanup FSK for 'Sender'.\n");
if (cnetz->fsk_tx_buffer) {
free(cnetz->fsk_tx_buffer);
cnetz->fsk_tx_buffer = NULL;
}
if (cnetz->dsp_speech_buffer) {
free(cnetz->dsp_speech_buffer);
cnetz->dsp_speech_buffer = NULL;
}
fsk_fm_exit(&cnetz->fsk_demod);
}
/* receive sample time and calculate speed against system clock
* tx: indicates transmit stream
* result: if set the actual signal speed is used (instead of sample rate) */
void calc_clock_speed(cnetz_t *cnetz, double samples, int tx, int result)
{
struct clock_speed *cs = &cnetz->clock_speed;
double ti;
double speed_ppm_avg[4];
int index = (result << 1) | tx;
int i;
if (!cnetz->measure_speed)
return;
ti = get_time();
/* skip some time to avoid false mesurement due to filling of buffers */
if (cs->meas_ti == 0.0) {
cs->meas_ti = ti + 1.0;
return;
}
if (cs->meas_ti > ti)
return;
/* start sample counting */
if (cs->start_ti[index] == 0.0) {
cs->start_ti[index] = ti;
cs->spl_count[index] = 0.0;
return;
}
/* add elapsed time */
cs->last_ti[index] = ti;
cs->spl_count[index] += samples;
/* only calculate speed, if one second has elapsed */
if (ti - cs->meas_ti <= 2.0)
return;
cs->meas_ti += 2.0;
/* add to ring buffer */
if (!cs->spl_count[0] || !cs->spl_count[1] || !cs->spl_count[2] || !cs->spl_count[3])
return;
for (index = 0; index < 4; index++) {
cs->speed_ppm[index][cs->idx[index] & 0xff] = (cs->spl_count[index] / (double)cnetz->sender.samplerate) / (cs->last_ti[index] - cs->start_ti[index]) * 1000000.0 - 1000000.0;
cs->idx[index]++;
if (cs->num[index]++ > 30)
cs->num[index] = 30;
speed_ppm_avg[index] = 0.0;
for (i = 0; i < cs->num[index]; i++)
speed_ppm_avg[index] += cs->speed_ppm[index][(cs->idx[index] - i - 1) & 0xff];
speed_ppm_avg[index] /= (double)cs->num[index];
}
PDEBUG_CHAN(DDSP, DEBUG_NOTICE, "Clock: RX=%.3f TX=%.3f; Signal: RX=%.3f TX=%.3f ppm\n", speed_ppm_avg[0], speed_ppm_avg[1], speed_ppm_avg[2], speed_ppm_avg[3]);
}
static int fsk_nothing_encode(cnetz_t *cnetz)
{
sample_t *spl;
double phase, bitstep;
int i, count;
spl = cnetz->fsk_tx_buffer;
phase = cnetz->fsk_tx_phase;
bitstep = cnetz->fsk_tx_bitstep * 256.0;
/* add 198 bits of no power (silence) */
for (i = 0; i < 198; i++) {
do {
*spl++ = -10.0; /* marker for power off */
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
}
/* depending on the number of samples, return the number */
count = ((uintptr_t)spl - (uintptr_t)cnetz->fsk_tx_buffer) / sizeof(*spl);
cnetz->fsk_tx_phase = phase;
cnetz->fsk_tx_buffer_length = count;
return count;
}
/* encode one data block into samples
* input: 184 data bits (including barker code)
* output: samples
* return number of samples */
static int fsk_block_encode(cnetz_t *cnetz, const char *bits, int ogk)
{
/* alloc samples, add 1 in case there is a rest */
sample_t *spl;
double phase, bitstep, deviation;
int i, count;
char last;
deviation = cnetz->fsk_deviation;
spl = cnetz->fsk_tx_buffer;
phase = cnetz->fsk_tx_phase;
bitstep = cnetz->fsk_tx_bitstep * 256.0;
/* add 7 bits of pause */
for (i = 0; i < 7; i++) {
do {
*spl++ = 0.0;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
}
/* add 184 bits */
last = ' ';
for (i = 0; i < 184; i++) {
switch (last) {
case ' ':
if (bits[i] == '1') {
/* ramp up from 0 */
do {
*spl++ = ramp_up[(uint8_t)phase] / 2 + deviation / 2;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
} else {
/* ramp down from 0 */
do {
*spl++ = ramp_down[(uint8_t)phase] / 2 - deviation / 2;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
}
break;
case '1':
if (bits[i] == '1') {
/* stay up */
do {
*spl++ = deviation;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
} else {
/* ramp down */
do {
*spl++ = ramp_down[(uint8_t)phase];
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
}
break;
case '0':
if (bits[i] == '1') {
/* ramp up */
do {
*spl++ = ramp_up[(uint8_t)phase];
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
} else {
/* stay down */
do {
*spl++ = -deviation;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
}
break;
}
last = bits[i];
}
/* add 7 bits of pause */
if (last == '0') {
/* ramp up to 0 */
do {
*spl++ = ramp_up[(uint8_t)phase] / 2 - deviation / 2;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
} else {
/* ramp down to 0 */
do {
*spl++ = ramp_down[(uint8_t)phase] / 2 + deviation / 2;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
}
for (i = 1; i < 7; i++) {
/* turn off power for OgK */
if (ogk) {
do {
*spl++ = -10.0; /* marker for power off */
phase += bitstep;
} while (phase < 256.0);
} else {
do {
*spl++ = 0.0;
phase += bitstep;
} while (phase < 256.0);
}
phase -= 256.0;
}
/* depending on the number of samples, return the number */
count = ((uintptr_t)spl - (uintptr_t)cnetz->fsk_tx_buffer) / sizeof(*spl);
cnetz->fsk_tx_phase = phase;
cnetz->fsk_tx_buffer_length = count;
return count;
}
/* encode one distributed data block into samples
* input: 184 data bits (including barker code)
* output: samples
* if a sample contains a marker, it indicates where to insert speech block
* return number of samples
*
* the marker is placed in the middle of the 6th bit.
* because we have a transition (ramp) in the middle of each bit.
* the phone will see the position of the marker as start of the 6th bit.
* the marker marks the pont where the speech is ramped up, so the phone
* will see the speech completely ramped up after the 6th bit
*/
static int fsk_distributed_encode(cnetz_t *cnetz, const char *bits)
{
/* alloc samples, add 1 in case there is a rest */
sample_t *spl, *marker;
double phase, bitstep, deviation;
int i, j, count;
char last;
deviation = cnetz->fsk_deviation;
spl = cnetz->fsk_tx_buffer;
phase = cnetz->fsk_tx_phase;
bitstep = cnetz->fsk_tx_bitstep * 256.0;
/* add 2 * (1+4+1 + 60) bits of pause / for speech */
for (i = 0; i < 2; i++) {
for (j = 0; j < 6; j++) {
do {
*spl++ = 0;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
}
marker = spl - (int)(cnetz->fsk_bitduration / 2.0); /* in the middle of the 6th bit */
for (j = 0; j < 60; j++) {
do {
*spl++ = 0;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
}
*marker += 10.0; /* marker for inserting speech */
}
/* add 46 * (1+4+1 + 60) bits */
for (i = 0; i < 46; i++) {
/* unmodulated bit */
do {
*spl++ = 0;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
last = ' ';
for (j = 0; j < 4; j++) {
switch (last) {
case ' ':
if (bits[i * 4 + j] == '1') {
/* ramp up from 0 */
do {
*spl++ = ramp_up[(uint8_t)phase] / 2 + deviation / 2;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
} else {
/* ramp down from 0 */
do {
*spl++ = ramp_down[(uint8_t)phase] / 2 - deviation / 2;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
}
break;
case '1':
if (bits[i * 4 + j] == '1') {
/* stay up */
do {
*spl++ = deviation;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
} else {
/* ramp down */
do {
*spl++ = ramp_down[(uint8_t)phase];
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
}
break;
case '0':
if (bits[i * 4 + j] == '1') {
/* ramp up */
do {
*spl++ = ramp_up[(uint8_t)phase];
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
} else {
/* stay down */
do {
*spl++ = -deviation;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
}
break;
}
last = bits[i * 4 + j];
}
/* ramp down */
if (last == '0') {
/* ramp up to 0 */
do {
*spl++ = ramp_up[(uint8_t)phase] / 2 - deviation / 2;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
} else {
/* ramp down to 0 */
do {
*spl++ = ramp_down[(uint8_t)phase] / 2 + deviation / 2;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
}
marker = spl - (int)(cnetz->fsk_bitduration / 2.0); /* in the middle of the 6th bit */
for (j = 0; j < 60; j++) {
do {
*spl++ = 0;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
}
*marker += 10.0; /* marker for inserting speech */
}
/* depending on the number of samples, return the number */
count = ((uintptr_t)spl - (uintptr_t)cnetz->fsk_tx_buffer) / sizeof(*spl);
cnetz->fsk_tx_phase = phase;
cnetz->fsk_tx_buffer_length = count;
return count;
}
/* decode samples and hut for bit changes
* use deviation to find greatest slope of the signal (bit change)
*/
void sender_receive(sender_t *sender, sample_t *samples, int length, double __attribute__((unused)) rf_level_db)
{
cnetz_t *cnetz = (cnetz_t *) sender;
/* measure rx sample speed */
calc_clock_speed(cnetz, length, 0, 0);
#ifdef TEST_SCRAMBLE
#ifdef TEST_UNSCRAMBLE
scrambler(&scrambler_test_scrambler1, samples, length);
#endif
jitter_save(&scrambler_test_jb, samples, length);
return;
#endif
if (cnetz->dsp_mode != DSP_MODE_OFF) {
iir_process(&cnetz->lp, samples, length);
fsk_fm_demod(&cnetz->fsk_demod, samples, length);
}
return;
}
/* shrink audio segment from 12.5 ms to the duration of 60 bits */
static int shrink_speech(cnetz_t *cnetz, sample_t *speech_buffer)
{
int speech_length;
jitter_load(&cnetz->sender.dejitter, speech_buffer, 100);
/* 1. compress dynamics */
compress_audio(&cnetz->cstate, speech_buffer, 100);
/* 2. upsample */
speech_length = samplerate_upsample(&cnetz->sender.srstate, speech_buffer, 100, speech_buffer);
/* 3. scramble */
if (cnetz->scrambler)
scrambler(&cnetz->scrambler_tx, speech_buffer, speech_length);
/* 4. pre-emphasis is done by cnetz code, not by common code */
/* pre-emphasis is only used when scrambler is off, see FTZ 171 TR 60 Clause 4 */
if (cnetz->pre_emphasis && !cnetz->scrambler)
pre_emphasis(&cnetz->estate, speech_buffer, speech_length);
return speech_length;
}
static int fsk_telegramm(cnetz_t *cnetz, sample_t *samples, uint8_t *power, int length)
{
int count = 0, pos, copy, i, speech_length, speech_pos;
sample_t *spl, *speech_buffer;
const char *bits;
speech_buffer = cnetz->dsp_speech_buffer;
speech_length = cnetz->dsp_speech_length;
speech_pos = cnetz->dsp_speech_pos;
again:
/* there must be length, otherwise we would skip blocks */
if (count == length)
return count;
pos = cnetz->fsk_tx_buffer_pos;
spl = cnetz->fsk_tx_buffer + pos;
/* start new telegramm, so we generate one */
if (pos == 0) {
/* a new hyper frame starts */
if (cnetz->sched_ts == 0 && cnetz->sched_r_m == 0) {
/* measure actual signal speed */
calc_clock_speed(cnetz, (double)cnetz->sender.samplerate * 2.4, 1, 1);
/* sync TX (might not be required, if there is no error in math calculation) */
if (!cnetz->sender.master) { /* if no link to a master, we are master */
/* we are master, so we store sample count and phase */
cnetz->frame_last_scount = cnetz->fsk_tx_scount;
cnetz->frame_last_phase = cnetz->fsk_tx_phase;
} else {
/* we are slave, so we sync to phase */
cnetz_t *master = (cnetz_t *)cnetz->sender.master;
/* it may happen that the sample count does not match with the master,
* because one has a phase wrap before and the other after a sample.
* then we do it next hyper frame cycle */
if (master->frame_last_scount == cnetz->fsk_tx_scount) {
PDEBUG(DDSP, DEBUG_DEBUG, "Sync phase of slave to master: master=%.15f, slave=%.15f, diff=%.15f\n", master->frame_last_phase, cnetz->fsk_tx_phase, master->frame_last_phase - cnetz->fsk_tx_phase);
cnetz->fsk_tx_phase = master->frame_last_phase;
} else {
PDEBUG(DDSP, DEBUG_DEBUG, "Not sync phase of slave to master: Sample counts during frame change are different, ignoring this time!\n");
}
}
}
/* switch to speech channel */
if (cnetz->sched_switch_mode && cnetz->sched_r_m == 0) {
if (--cnetz->sched_switch_mode == 0) {
/* OgK / SpK(K) / SpK(V) */
PDEBUG_CHAN(DDSP, DEBUG_INFO, "Switching channel (mode)\n");
cnetz_set_dsp_mode(cnetz, cnetz->sched_dsp_mode);
}
}
switch (cnetz->dsp_mode) {
case DSP_MODE_OGK:
/* if automatic cell selection is used, toggle between
* two cells until a response for one cell is received
*/
if (cnetz->cell_auto)
cnetz->cell_nr = (cnetz->sched_ts & 7) >> 2;
/* send on timeslots depending on the cell_nr:
* cell 0: 0, 8, 16, 24
* cell 1: 4, 12, 20, 28
*/
if (((cnetz->sched_ts & 7) == 0 && cnetz->cell_nr == 0)
|| ((cnetz->sched_ts & 7) == 4 && cnetz->cell_nr == 1)) {
if (cnetz->sched_r_m == 0) {
/* set last time slot, so we can match received message from mobile station */
cnetz->sched_last_ts[cnetz->cell_nr] = cnetz->sched_ts;
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Transmitting 'Rufblock' at timeslot %d\n", cnetz->sched_ts);
bits = cnetz_encode_telegramm(cnetz);
} else {
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Transmitting 'Meldeblock' at timeslot %d\n", cnetz->sched_ts);
bits = cnetz_encode_telegramm(cnetz);
}
fsk_block_encode(cnetz, bits, 1);
} else {
fsk_nothing_encode(cnetz);
}
break;
case DSP_MODE_SPK_K:
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Transmitting 'Konzentrierte Signalisierung'\n");
bits = cnetz_encode_telegramm(cnetz);
fsk_block_encode(cnetz, bits, 0);
break;
case DSP_MODE_SPK_V:
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Transmitting 'Verteilte Signalisierung'\n");
bits = cnetz_encode_telegramm(cnetz);
fsk_distributed_encode(cnetz, bits);
break;
case DSP_MODE_OFF:
default:
fsk_nothing_encode(cnetz);
}
if (cnetz->dsp_mode == DSP_MODE_SPK_V) {
/* count sub frame */
cnetz->sched_ts += 8;
} else {
/* count slot */
if (cnetz->sched_r_m == 0)
cnetz->sched_r_m = 1;
else {
cnetz->sched_r_m = 0;
cnetz->sched_ts++;
}
}
if (cnetz->sched_ts == 32)
cnetz->sched_ts = 0;
}
copy = cnetz->fsk_tx_buffer_length - pos;
if (length - count < copy)
copy = length - count;
for (i = 0; i < copy; i++) {
if (*spl > 5.0) { /* speech marker found */
int begin, end, j;
/* correct marker (not the best way) */
*spl -= 10.0;
begin = (int)cnetz->fsk_bitduration;
end = (int)(cnetz->fsk_bitduration * 61.0);
/* get audio */
speech_length = shrink_speech(cnetz, speech_buffer + begin);
/* ramp before speech */
for (j = 0; j < begin; j++) {
/* ramp up from 0 to speech level */
speech_buffer[j] = speech_buffer[begin] * (ramp_up[j * 256 / begin] / cnetz->fsk_deviation / 2.0 + 0.5);
}
speech_length += begin; /* add one bit duration before speech*/
/* ramp after speech */
while (speech_length < end) {
speech_buffer[speech_length] = speech_buffer[speech_length - 1];
speech_length++;
}
speech_length = end; /* shorten 'speech_length', if greater than 'end' */
for (j = 0; j < begin; j++) {
/* ramp down from speech level to 0 */
speech_buffer[end + j] = speech_buffer[end - 1] * (ramp_down[j * 256 / begin] / cnetz->fsk_deviation / 2.0 + 0.5);
}
speech_length += begin; /* add one bit duration after speech */
speech_pos = 0;
}
if (*spl < -5.0) { /* power off marker found */
/* correct marker (not the best way) */
*spl += 10.0;
*power = 0;
} else
*power = 1;
/* add speech as long as we have something left in buffer */
if (speech_pos < speech_length)
*samples++ = *spl + speech_buffer[speech_pos++];
else
*samples++ = *spl;
spl++;
power++;
}
cnetz->dsp_speech_length = speech_length;
cnetz->dsp_speech_pos = speech_pos;
cnetz->fsk_tx_scount += copy;
pos += copy;
count += copy;
if (pos == cnetz->fsk_tx_buffer_length) {
cnetz->fsk_tx_buffer_pos = 0;
goto again;
}
cnetz->fsk_tx_buffer_pos = pos;
return count;
}
/* Provide stream of audio toward radio unit */
void sender_send(sender_t *sender, sample_t *samples, uint8_t *power, int length)
{
cnetz_t *cnetz = (cnetz_t *) sender;
int count;
/* measure tx sample speed */
calc_clock_speed(cnetz, length, 1, 0);
#ifdef TEST_SCRAMBLE
jitter_load(&scrambler_test_jb, samples, length);
scrambler(&scrambler_test_scrambler2, samples, length);
return;
#endif
count = fsk_telegramm(cnetz, samples, power, length);
if (count < length) {
printf("length=%d < count=%d\n", length, count);
printf("this shall not happen, so please fix!\n");
exit(0);
}
}
/* unshrink audio segment from the duration of 60 bits to 12.5 ms */
void unshrink_speech(cnetz_t *cnetz, sample_t *speech_buffer, int count)
{
sample_t *spl;
int pos, i;
double x, y, x_last, y_last, factor;
/* check if we still have a transaction
* this might not be true, if we just released transaction, but still
* get a complete frame before we already switched back to OgK.
*/
if (!cnetz->trans_list)
return;
/* fix offset between speech blocks by using high pass filter */
/* use first sample as previous sample, so we don't have a level jump between two subsequent audio chunks */
x_last = speech_buffer[0];
y_last = cnetz->offset_y_last;
factor = cnetz->offset_factor;
for (i = 0; i < count; i++) {
/* change level */
x = speech_buffer[i];
/* high-pass to remove low level frequencies, caused by level jump between audio chunks */
y = factor * (y_last + x - x_last);
x_last = x;
y_last = y;
speech_buffer[i] = y;
}
cnetz->offset_y_last = y_last;
/* 4. de-emphasis is done by cnetz code, not by common code */
/* de-emphasis is only used when scrambler is off, see FTZ 171 TR 60 Clause 4 */
if (cnetz->de_emphasis)
dc_filter(&cnetz->estate, speech_buffer, count);
if (cnetz->de_emphasis && !cnetz->scrambler)
de_emphasis(&cnetz->estate, speech_buffer, count);
/* 3. descramble */
if (cnetz->scrambler)
scrambler(&cnetz->scrambler_rx, speech_buffer, count);
/* 2. decompress time */
count = samplerate_downsample(&cnetz->sender.srstate, speech_buffer, count);
/* 1. expand dynamics */
expand_audio(&cnetz->cstate, speech_buffer, count);
/* to call control */
spl = cnetz->sender.rxbuf;
pos = cnetz->sender.rxbuf_pos;
for (i = 0; i < count; i++) {
spl[pos++] = speech_buffer[i];
if (pos == 160) {
call_up_audio(cnetz->trans_list->callref, spl, 160);
pos = 0;
}
}
cnetz->sender.rxbuf_pos = pos;
}
const char *cnetz_dsp_mode_name(enum dsp_mode mode)
{
static char invalid[16];
switch (mode) {
case DSP_SCHED_NONE:
return "SCHED_NONE";
case DSP_MODE_OFF:
return "OFF";
case DSP_MODE_OGK:
return "OGK";
case DSP_MODE_SPK_K:
return "SPK_K";
case DSP_MODE_SPK_V:
return "SPK_V";
}
sprintf(invalid, "invalid(%d)", mode);
return invalid;
}
void cnetz_set_dsp_mode(cnetz_t *cnetz, enum dsp_mode mode)
{
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "DSP mode %s -> %s\n", cnetz_dsp_mode_name(cnetz->dsp_mode), cnetz_dsp_mode_name(mode));
cnetz->dsp_mode = mode;
/* we must get rid of partly received frame */
fsk_demod_reset(&cnetz->fsk_demod);
}
void cnetz_set_sched_dsp_mode(cnetz_t *cnetz, enum dsp_mode mode, int frames_ahead)
{
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, " Schedule DSP mode %s -> %s in %d frames\n", cnetz_dsp_mode_name(cnetz->dsp_mode), cnetz_dsp_mode_name(mode), frames_ahead);
cnetz->sched_dsp_mode = mode;
cnetz->sched_switch_mode = frames_ahead;
}
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