447 lines
12 KiB
C
447 lines
12 KiB
C
/* B-Netz signal processing
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*
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* (C) 2016 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 bnetz->sender.kanal
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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 "../common/timer.h"
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#include "../common/call.h"
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#include "bnetz.h"
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#include "dsp.h"
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#define PI 3.1415927
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/* Notes on TX_PEAK_TONE level:
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*
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* At 2000 Hz the deviation shall be 4 kHz, so with emphasis the deviation
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* at 1000 Hz would be theoretically 2 kHz. This is factor 0.714 below
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* 2.8 kHz deviation we want at dBm0.
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*/
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/* signaling */
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#define MAX_DEVIATION 4000.0
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#define MAX_MODULATION 3000.0
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#define DBM0_DEVIATION 2800.0 /* deviation of dBm0 at 1 kHz */
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#define TX_PEAK_TONE (4000.0 / 2000.0 * 1000.0 / DBM0_DEVIATION)
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#define MAX_DISPLAY 1.4 /* something above dBm0 */
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#define BIT_DURATION 0.010 /* bit length: 10 ms */
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#define FILTER_STEP 0.001 /* step every 1 ms */
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#define METERING_HZ 2900 /* metering pulse frequency */
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#define TONE_DETECT_TH 70 /* 70 milliseconds to detect continuous tone */
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/* carrier loss detection */
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#define LOSS_INTERVAL 1000 /* filter steps (milliseconds) for one second interval */
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#define LOSS_TIME 12 /* duration of signal loss before release */
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/* two signaling tones */
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static double fsk_bits[2] = {
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2070.0,
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1950.0,
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};
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/* table for fast sine generation */
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static sample_t dsp_sine[65536];
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/* global init for FSK */
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void dsp_init(void)
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{
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int i;
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PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table.\n");
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for (i = 0; i < 65536; i++) {
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dsp_sine[i] = sin((double)i / 65536.0 * 2.0 * PI) * TX_PEAK_TONE;
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}
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}
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/* Init transceiver instance. */
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int dsp_init_sender(bnetz_t *bnetz)
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{
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sample_t *spl;
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int i;
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if ((bnetz->sender.samplerate % 1000)) {
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PDEBUG(DDSP, DEBUG_ERROR, "Samples rate must be a multiple of 1000 bits per second.\n");
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return -EINVAL;
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}
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PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Init DSP for 'Sender'.\n");
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/* set modulation parameters */
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sender_set_fm(&bnetz->sender, MAX_DEVIATION, MAX_MODULATION, DBM0_DEVIATION, MAX_DISPLAY);
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audio_init_loss(&bnetz->sender.loss, LOSS_INTERVAL, bnetz->sender.loss_volume, LOSS_TIME);
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bnetz->samples_per_bit = bnetz->sender.samplerate * BIT_DURATION;
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PDEBUG(DDSP, DEBUG_DEBUG, "Using %d samples per bit duration.\n", bnetz->samples_per_bit);
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bnetz->fsk_filter_step = bnetz->sender.samplerate * FILTER_STEP;
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PDEBUG(DDSP, DEBUG_DEBUG, "Using %d samples per filter step.\n", bnetz->fsk_filter_step);
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spl = calloc(16, bnetz->samples_per_bit * sizeof(*spl));
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if (!spl) {
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PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
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return -ENOMEM;
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}
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bnetz->telegramm_spl = spl;
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spl = calloc(1, bnetz->samples_per_bit * sizeof(*spl));
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if (!spl) {
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PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
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return -ENOMEM;
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}
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bnetz->fsk_filter_spl = spl;
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bnetz->fsk_filter_bit = -1;
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bnetz->tone_detected = -1;
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/* count symbols */
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for (i = 0; i < 2; i++) {
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audio_goertzel_init(&bnetz->fsk_goertzel[i], fsk_bits[i], bnetz->sender.samplerate);
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bnetz->phaseshift65536[i] = 65536.0 / ((double)bnetz->sender.samplerate / fsk_bits[i]);
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PDEBUG(DDSP, DEBUG_DEBUG, "phaseshift[%d] = %.4f (must be arround 64 at 8000hz)\n", i, bnetz->phaseshift65536[i]);
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}
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return 0;
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}
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/* Cleanup transceiver instance. */
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void dsp_cleanup_sender(bnetz_t *bnetz)
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{
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PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Cleanup DSP for 'Sender'.\n");
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if (bnetz->telegramm_spl) {
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free(bnetz->telegramm_spl);
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bnetz->telegramm_spl = NULL;
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}
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if (bnetz->fsk_filter_spl) {
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free(bnetz->fsk_filter_spl);
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bnetz->fsk_filter_spl = NULL;
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}
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}
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/* Count duration of tone and indicate detection/loss to protocol handler. */
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static void fsk_receive_tone(bnetz_t *bnetz, int bit, int goodtone, double level, double quality)
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{
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/* lost tone because it is not good anymore or has changed */
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if (!goodtone || bit != bnetz->tone_detected) {
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if (bnetz->tone_count >= TONE_DETECT_TH) {
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PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Lost %.0f Hz tone after %d ms.\n", fsk_bits[bnetz->tone_detected], bnetz->tone_count);
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bnetz_receive_tone(bnetz, -1);
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}
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if (goodtone)
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bnetz->tone_detected = bit;
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else
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bnetz->tone_detected = -1;
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bnetz->tone_count = 0;
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return;
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}
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bnetz->tone_count++;
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if (bnetz->tone_count >= TONE_DETECT_TH)
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audio_reset_loss(&bnetz->sender.loss);
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if (bnetz->tone_count == TONE_DETECT_TH) {
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PDEBUG_CHAN(DDSP, DEBUG_INFO, "Detecting continuous tone: %.0f:Level=%3.0f%% Quality=%3.0f%%\n", fsk_bits[bnetz->tone_detected], level * 100.0, quality * 100.0);
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bnetz_receive_tone(bnetz, bnetz->tone_detected);
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}
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}
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/* Collect 16 data bits (digit) and check for sync marc '01110'. */
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static void fsk_receive_bit(bnetz_t *bnetz, int bit, double level, double quality)
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{
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int i;
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bnetz->fsk_filter_telegramm = (bnetz->fsk_filter_telegramm << 1) | bit;
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bnetz->fsk_filter_quality[bnetz->fsk_filter_qualidx] = quality;
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bnetz->fsk_filter_level[bnetz->fsk_filter_qualidx] = level;
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if (++bnetz->fsk_filter_qualidx == 16)
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bnetz->fsk_filter_qualidx = 0;
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/* check if pattern 01110xxxxxxxxxxx matches */
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if ((bnetz->fsk_filter_telegramm & 0xf800) != 0x7000)
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return;
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/* get worst bit and average level */
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level = 0;
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for (i = 0; i < 16; i++) {
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if (bnetz->fsk_filter_quality[i] < quality)
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quality = bnetz->fsk_filter_quality[i];
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level = bnetz->fsk_filter_level[i];
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}
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/* send telegramm */
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bnetz_receive_telegramm(bnetz, bnetz->fsk_filter_telegramm, level, quality);
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}
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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 10ms. This window slides over audio stream
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* and is processed every 1ms. (one step) */
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static inline void fsk_decode_step(bnetz_t *bnetz, 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 = bnetz->samples_per_bit;
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spl = bnetz->fsk_filter_spl;
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level = audio_level(spl, max);
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if (audio_detect_loss(&bnetz->sender.loss, level))
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bnetz_loss_indication(bnetz);
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audio_goertzel(bnetz->fsk_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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/* scale it, since both filters overlap by some percent */
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#define MIN_QUALITY 0.08
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softbit = (softbit - MIN_QUALITY) / (0.850 - MIN_QUALITY - MIN_QUALITY);
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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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#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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#endif
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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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// 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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// FIXME: better threshold
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/* adjust level, so we get peak of sine curve */
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if (level / 0.63 > 0.05 && (softbit > 0.75 || softbit < 0.25)) {
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fsk_receive_tone(bnetz, bit, 1, level / 0.63662 / TX_PEAK_TONE, quality);
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} else
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fsk_receive_tone(bnetz, bit, 0, level / 0.63662 / TX_PEAK_TONE, quality);
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if (bnetz->fsk_filter_bit != bit) {
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/* if we have a bit change, reset sample counter to one half bit duration */
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bnetz->fsk_filter_bit = bit;
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bnetz->fsk_filter_sample = 5;
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} else if (--bnetz->fsk_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_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 we get peak of sine curve */
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fsk_receive_bit(bnetz, bit, level / 0.63662 / TX_PEAK_TONE, quality);
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bnetz->fsk_filter_sample = 10;
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}
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}
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/* Process received audio stream from radio unit. */
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void sender_receive(sender_t *sender, sample_t *samples, int length)
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{
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bnetz_t *bnetz = (bnetz_t *) sender;
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sample_t *spl;
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int max, pos, step;
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int i;
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/* write received samples to decode buffer */
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max = bnetz->samples_per_bit;
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pos = bnetz->fsk_filter_pos;
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step = bnetz->fsk_filter_step;
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spl = bnetz->fsk_filter_spl;
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for (i = 0; i < length; i++) {
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spl[pos++] = samples[i];
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if (pos == max)
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pos = 0;
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/* if filter step has been reched */
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if (!(pos % step)) {
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fsk_decode_step(bnetz, pos);
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}
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}
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bnetz->fsk_filter_pos = pos;
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if (bnetz->dsp_mode == DSP_MODE_AUDIO && bnetz->callref) {
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int count;
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count = samplerate_downsample(&bnetz->sender.srstate, samples, length);
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spl = bnetz->sender.rxbuf;
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pos = bnetz->sender.rxbuf_pos;
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for (i = 0; i < count; i++) {
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spl[pos++] = samples[i];
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if (pos == 160) {
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call_tx_audio(bnetz->callref, spl, 160);
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pos = 0;
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}
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}
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bnetz->sender.rxbuf_pos = pos;
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} else
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bnetz->sender.rxbuf_pos = 0;
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}
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static void fsk_tone(bnetz_t *bnetz, sample_t *samples, int length, int tone)
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{
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double phaseshift, phase;
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int i;
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phase = bnetz->phase65536;
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phaseshift = bnetz->phaseshift65536[tone];
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for (i = 0; i < length; i++) {
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*samples++ = dsp_sine[(uint16_t)phase];
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phase += phaseshift;
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if (phase >= 65536)
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phase -= 65536;
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}
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bnetz->phase65536 = phase;
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}
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static int fsk_telegramm(bnetz_t *bnetz, sample_t *samples, int length)
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{
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sample_t *spl;
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const char *telegramm;
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int i, j;
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double phaseshift, phase;
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int count, max;
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next_telegramm:
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if (!bnetz->telegramm) {
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/* request telegramm */
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// PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Request new 'Telegramm'.\n");
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telegramm = bnetz_get_telegramm(bnetz);
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if (!telegramm) {
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PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Stop sending 'Telegramm'.\n");
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return length;
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}
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bnetz->telegramm = 1;
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bnetz->telegramm_pos = 0;
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spl = bnetz->telegramm_spl;
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/* render telegramm */
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phase = bnetz->phase65536;
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for (i = 0; i < 16; i++) {
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phaseshift = bnetz->phaseshift65536[telegramm[i] == '1'];
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for (j = 0; j < bnetz->samples_per_bit; j++) {
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*spl++ = dsp_sine[(uint16_t)phase];
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phase += phaseshift;
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if (phase >= 65536)
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phase -= 65536;
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}
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}
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bnetz->phase65536 = phase;
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}
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/* send audio from telegramm */
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max = bnetz->samples_per_bit * 16;
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count = max - bnetz->telegramm_pos;
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if (count > length)
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count = length;
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spl = bnetz->telegramm_spl + bnetz->telegramm_pos;
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for (i = 0; i < count; i++)
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*samples++ = *spl++;
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length -= count;
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bnetz->telegramm_pos += count;
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/* check for end of telegramm */
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if (bnetz->telegramm_pos == max) {
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bnetz->telegramm = 0;
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/* we need more ? */
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if (length)
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goto next_telegramm;
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}
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return length;
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}
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/* Provide stream of audio toward radio unit */
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void sender_send(sender_t *sender, sample_t *samples, int length)
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{
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bnetz_t *bnetz = (bnetz_t *) sender;
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int len;
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again:
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switch (bnetz->dsp_mode) {
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case DSP_MODE_SILENCE:
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memset(samples, 0, length * sizeof(*samples));
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break;
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case DSP_MODE_AUDIO:
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jitter_load(&bnetz->sender.dejitter, samples, length);
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break;
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case DSP_MODE_0:
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fsk_tone(bnetz, samples, length, 0);
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break;
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case DSP_MODE_1:
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fsk_tone(bnetz, samples, length, 1);
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break;
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case DSP_MODE_TELEGRAMM:
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/* Encode telegramm into audio stream. If telegramms have
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* stopped, process again for rest of stream. */
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len = fsk_telegramm(bnetz, samples, length);
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if (len) {
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samples += length - len;
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length = len;
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goto again;
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}
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break;
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}
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}
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const char *bnetz_dsp_mode_name(enum dsp_mode mode)
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{
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static char invalid[16];
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switch (mode) {
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case DSP_MODE_SILENCE:
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return "SILENCE";
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case DSP_MODE_AUDIO:
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return "AUDIO";
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case DSP_MODE_0:
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return "TONE 0";
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case DSP_MODE_1:
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return "TONE 1";
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case DSP_MODE_TELEGRAMM:
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return "TELEGRAMM";
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}
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sprintf(invalid, "invalid(%d)", mode);
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return invalid;
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}
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void bnetz_set_dsp_mode(bnetz_t *bnetz, enum dsp_mode mode)
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{
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/* reset telegramm */
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if (mode == DSP_MODE_TELEGRAMM && bnetz->dsp_mode != mode)
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bnetz->telegramm = 0;
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PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "DSP mode %s -> %s\n", bnetz_dsp_mode_name(bnetz->dsp_mode), bnetz_dsp_mode_name(mode));
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bnetz->dsp_mode = mode;
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}
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