osmocom-analog/src/mpt1327/dsp.c

350 lines
10 KiB
C

/* audio processing
*
* (C) 2021 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 mpt1327->sender.kanal
#include <stdio.h>
#include <stdint.h>
#include <stdlib.h>
#include <string.h>
#include <errno.h>
#include <math.h>
#include "../libsample/sample.h"
#include "../libmobile/call.h"
#include "../libdebug/debug.h"
#include "../libtimer/timer.h"
#include "mpt1327.h"
#include "dsp.h"
#include "message.h"
#define PI M_PI
/* signaling */
#define MAX_DEVIATION 2500.0
#define MAX_MODULATION 2550.0
#define SPEECH_DEVIATION 1500.0 /* deviation of speech (no emphasis) */
#define TX_PEAK_FSK (1500.0 / SPEECH_DEVIATION)
#define BIT_RATE 1200.0
#define BIT_ADJUST 0.1 /* how much do we adjust bit clock on frequency change */
#define F0 1800.0
#define F1 1200.0
#define MAX_DISPLAY 1.4 /* something above speech level */
/* carrier loss detection */
#define MUTE_TIME 0.1 /* time to mute after loosing signal */
void dsp_init(void)
{
}
static int fsk_send_bit(void *inst);
static void fsk_receive_bit(void *inst, int bit, double quality, double level);
/* Init FSK of transceiver */
int dsp_init_sender(mpt1327_t *mpt1327, double squelch_db)
{
int rc;
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Init DSP for Transceiver.\n");
/* init squelch */
squelch_init(&mpt1327->squelch, mpt1327->sender.kanal, squelch_db, MUTE_TIME, MUTE_TIME);
/* set modulation parameters */
sender_set_fm(&mpt1327->sender, MAX_DEVIATION, MAX_MODULATION, SPEECH_DEVIATION, MAX_DISPLAY);
PDEBUG(DDSP, DEBUG_DEBUG, "Using FSK level of %.3f (%.3f KHz deviation)\n", TX_PEAK_FSK, SPEECH_DEVIATION * TX_PEAK_FSK / 1e3);
/* init fsk */
if (fsk_mod_init(&mpt1327->fsk_mod, mpt1327, fsk_send_bit, mpt1327->sender.samplerate, BIT_RATE, F0, F1, TX_PEAK_FSK, 1, 0) < 0) {
PDEBUG_CHAN(DDSP, DEBUG_ERROR, "FSK init failed!\n");
return -EINVAL;
}
if (fsk_demod_init(&mpt1327->fsk_demod, mpt1327, fsk_receive_bit, mpt1327->sender.samplerate, BIT_RATE, F0, F1, BIT_ADJUST) < 0) {
PDEBUG_CHAN(DDSP, DEBUG_ERROR, "FSK init failed!\n");
return -EINVAL;
}
mpt1327->dmp_frame_level = display_measurements_add(&mpt1327->sender.dispmeas, "Frame Level", "%.1f %% (last)", DISPLAY_MEAS_LAST, DISPLAY_MEAS_LEFT, 0.0, 150.0, 100.0);
mpt1327->dmp_frame_quality = display_measurements_add(&mpt1327->sender.dispmeas, "Frame Quality", "%.1f %% (last)", DISPLAY_MEAS_LAST, DISPLAY_MEAS_LEFT, 0.0, 100.0, 100.0);
/* repeater */
rc = jitter_create(&mpt1327->repeater_dejitter, mpt1327->sender.samplerate / 5);
if (rc < 0) {
PDEBUG(DDSP, DEBUG_ERROR, "Failed to create and init repeater buffer!\n");
goto error;
}
return 0;
error:
dsp_cleanup_sender(mpt1327);
return rc;
}
/* Cleanup transceiver instance. */
void dsp_cleanup_sender(mpt1327_t *mpt1327)
{
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Cleanup DSP for Transceiver.\n");
fsk_mod_cleanup(&mpt1327->fsk_mod);
fsk_demod_cleanup(&mpt1327->fsk_demod);
jitter_destroy(&mpt1327->repeater_dejitter);
}
/* Check for SYNC bits, then collect data bits */
static void fsk_receive_bit(void *inst, int bit, double quality, double level)
{
mpt1327_t *mpt1327 = (mpt1327_t *)inst;
int i;
/* normalize FSK level */
level /= TX_PEAK_FSK;
// printf("bit=%d quality=%.4f\n", bit, quality);
if (!mpt1327->rx_in_sync) {
mpt1327->rx_sync = (mpt1327->rx_sync << 1) | bit;
/* level and quality */
mpt1327->rx_level[mpt1327->rx_count & 0xff] = level;
mpt1327->rx_quality[mpt1327->rx_count & 0xff] = quality;
mpt1327->rx_count++;
/* check if sync pattern match */
if (mpt1327->rx_sync != mpt1327->sync_word)
return;
/* average level and quality */
level = quality = 0;
for (i = 0; i < 16; i++) {
level += mpt1327->rx_level[(mpt1327->rx_count - 1 - i) & 0xff];
quality += mpt1327->rx_quality[(mpt1327->rx_count - 1 - i) & 0xff];
}
level /= 16.0; quality /= 16.0;
// printf("sync (level = %.2f, quality = %.2f\n", level, quality);
/* do not accept garbage */
if (quality < 0.65)
return;
/* rest sync register */
mpt1327->rx_sync = 0;
mpt1327->rx_in_sync = 1;
mpt1327->rx_count = 0;
/* mute audio from now on */
mpt1327->rx_mute = 1;
return;
}
/* read bits */
mpt1327->rx_bits = (mpt1327->rx_bits << 1) | (bit & 1);
mpt1327->rx_level[mpt1327->rx_count] = level;
mpt1327->rx_quality[mpt1327->rx_count] = quality;
if (++mpt1327->rx_count != 64)
return;
/* check parity */
if (mpt1327_checkbits(mpt1327->rx_bits, NULL) != (mpt1327->rx_bits & 0xffff)) {
PDEBUG(DDSP, DEBUG_NOTICE, "Received corrupt codeword or noise.\n");
mpt1327->rx_in_sync = 0;
mpt1327->rx_mute = 0;
return;
}
/* reset counter for next frame */
mpt1327->rx_count = 0;
/* average level and quality */
level = quality = 0;
for (i = 0; i < 64; i++) {
level += mpt1327->rx_level[i];
quality += mpt1327->rx_quality[i];
}
level /= 64.0; quality /= 64.0;
/* update measurements */
display_measurements_update(mpt1327->dmp_frame_level, level * 100.0, 0.0);
display_measurements_update(mpt1327->dmp_frame_quality, quality * 100.0, 0.0);
/* convert level so that received level at TX_PEAK_FSK results in 1.0 (100%) */
mpt1327_receive_codeword(mpt1327, mpt1327->rx_bits, quality, level);
}
/* Process received audio stream from radio unit. */
void sender_receive(sender_t *sender, sample_t *samples, int length, double __attribute__((unused)) rf_level_db)
{
mpt1327_t *mpt1327 = (mpt1327_t *) sender;
sample_t *spl;
int pos;
int i;
int was_mute = mpt1327->rx_mute; /* remember, so always mute whole chunk */
int was_pressel_on = mpt1327->pressel_on;
/* if channel is off, do nothing */
if (mpt1327->dsp_mode == DSP_MODE_OFF) {
/* measure squelch even if channel is turned off */
if (!isinf(mpt1327->squelch.threshold_db))
squelch(&mpt1327->squelch, rf_level_db, (double)length / (double)mpt1327->sender.samplerate);
return;
}
/* fsk signal */
fsk_demod_receive(&mpt1327->fsk_demod, samples, length);
/* on traffic channel mute and indicate signal strength */
if (mpt1327->dsp_mode == DSP_MODE_TRAFFIC) {
/* process signal mute/loss, also for signalling tone */
if (!isinf(mpt1327->squelch.threshold_db)) {
/* use squelch to unmute and reset call timer */
switch (squelch(&mpt1327->squelch, rf_level_db, (double)length / (double)mpt1327->sender.samplerate)) {
case SQUELCH_LOSS:
case SQUELCH_MUTE:
memset(samples, 0, sizeof(*samples) * length);
break;
default:
mpt1327_signal_indication(mpt1327);
}
} else {
/* muting audio while pressel is off */
if (!was_pressel_on || !mpt1327->pressel_on)
memset(samples, 0, sizeof(*samples) * length);
}
/* muting audio while receiving frame */
if (was_mute || mpt1327->rx_mute)
memset(samples, 0, sizeof(*samples) * length);
}
if (mpt1327->dsp_mode == DSP_MODE_TRAFFIC) {
/* if repeater mode, store sample in jitter buffer */
if (mpt1327->repeater)
jitter_save(&mpt1327->repeater_dejitter, samples, length);
if (mpt1327->unit && mpt1327->unit->callref) {
int count;
count = samplerate_downsample(&mpt1327->sender.srstate, samples, length);
spl = mpt1327->sender.rxbuf;
pos = mpt1327->sender.rxbuf_pos;
for (i = 0; i < count; i++) {
spl[pos++] = samples[i];
if (pos == 160) {
call_up_audio(mpt1327->unit->callref, spl, 160);
pos = 0;
}
}
mpt1327->sender.rxbuf_pos = pos;
} else
mpt1327->sender.rxbuf_pos = 0;
} else
mpt1327->sender.rxbuf_pos = 0;
}
static int fsk_send_bit(void *inst)
{
mpt1327_t *mpt1327 = (mpt1327_t *)inst;
/* send frame bit (prio) */
if (!mpt1327->tx_bit_num || mpt1327->tx_count == mpt1327->tx_bit_num) {
/* request frame */
mpt1327->tx_bit_num = mpt1327_send_codeword(mpt1327, &mpt1327->tx_bits);
if (mpt1327->tx_bit_num == 0) {
return -1;
}
mpt1327->tx_count = 0;
}
return (mpt1327->tx_bits >> (63 - mpt1327->tx_count++)) & 1;
return -1;
}
/* Provide stream of audio toward radio unit */
void sender_send(sender_t *sender, sample_t *samples, uint8_t *power, int length)
{
mpt1327_t *mpt1327 = (mpt1327_t *) sender;
if (mpt1327->dsp_mode == DSP_MODE_OFF) {
memset(power, 0, length);
memset(samples, 0, sizeof(*samples) * length);
return;
}
memset(power, 1, length);
if (mpt1327->dsp_mode == DSP_MODE_TRAFFIC) {
jitter_load(&mpt1327->sender.dejitter, samples, length);
/* if repeater mode, sum samples from jitter buffer to samples */
if (mpt1327->repeater) {
sample_t uplink[length];
int i;
jitter_load(&mpt1327->repeater_dejitter, uplink, length);
for (i = 0; i < length; i++)
samples[i] += uplink[i];
}
} else
memset(samples, 0, sizeof(*samples) * length);
/* If there is something to modulate (pending TX frame),
* overwrite audio with FSK audio. */
fsk_mod_send(&mpt1327->fsk_mod, samples, length, 0);
}
const char *mpt1327_dsp_mode_name(enum dsp_mode mode)
{
static char invalid[16];
switch (mode) {
case DSP_MODE_OFF:
return "OFF";
case DSP_MODE_TRAFFIC:
return "TRAFFIC";
case DSP_MODE_CONTROL:
return "CONTROL";
}
sprintf(invalid, "invalid(%d)", mode);
return invalid;
}
void mpt1327_set_dsp_mode(mpt1327_t *mpt1327, enum dsp_mode mode, int repeater)
{
//NOTE: DO NOT RESET FRAME, because mode may change before frame has been sent!
if (mode == DSP_MODE_CONTROL)
mpt1327->sync_word = 0xc4d7;
if (mode == DSP_MODE_TRAFFIC)
mpt1327->sync_word = 0x3b28;
if (repeater)
jitter_reset(&mpt1327->repeater_dejitter);
mpt1327->repeater = repeater;
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "DSP mode %s -> %s\n", mpt1327_dsp_mode_name(mpt1327->dsp_mode), mpt1327_dsp_mode_name(mode));
mpt1327->dsp_mode = mode;
}
void mpt1327_reset_sync(mpt1327_t *mpt1327)
{
mpt1327->rx_in_sync = 0;
mpt1327->rx_sync = 0;
mpt1327->rx_mute = 0;
}