560 lines
17 KiB
C
560 lines
17 KiB
C
/* NMT audio 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 nmt->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 "../libsample/sample.h"
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#include "../liblogging/logging.h"
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#include "nmt.h"
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#include "transaction.h"
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#include "dsp.h"
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#define PI M_PI
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/* Notes on TX_PEAK_FSK level:
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*
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* This deviation is -2.2db below the speech deviation.
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*
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* At 1800 Hz the deviation shall be 4.2 kHz, so with emphasis the deviation
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* at 1000 Hz would be theoretically 2.333 kHz. This is factor 0.777 below
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* 3 kHz deviation we want at speech.
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*/
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/* Notes on TX_PEAK_SUPER (supervisory signal) level:
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*
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* This level has 0.3 kHz deviation at 4015 Hz.
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*
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* Same calculation as above, but now we want 0.3 kHz deviation after emphasis,
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* so we calculate what we would need at 1000 Hz in relation to 3 kHz
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* deviation.
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*/
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/* signaling */
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#define MAX_DEVIATION 4700.0
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#define MAX_MODULATION 4055.0
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#define SPEECH_DEVIATION 3000.0 /* deviation of speech at 1 kHz */
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#define TX_PEAK_FSK (4200.0 / 1800.0 * 1000.0 / SPEECH_DEVIATION)
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#define TX_PEAK_SUPER (300.0 / 4015.0 * 1000.0 / SPEECH_DEVIATION)
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#define BIT_RATE 1200.0
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#define BIT_ADJUST 0.1 /* how much do we adjust bit clock on frequency change */
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#define F0 1800.0
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#define F1 1200.0
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#define MAX_DISPLAY 1.4 /* something above speech level */
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#define DIALTONE_HZ 425.0 /* dial tone frequency */
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#define TX_PEAK_DIALTONE 1.0 /* dial tone peak FIXME: Not found in the specs! */
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#define SUPER_BANDWIDTH 30.0 /* distance between two SAT tones, also bandwidth for goertzel filter */
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#define SUPER_PRINT 2 /* print supervisory signal measurement every 0.5 seconds */
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#define SUPER_LOST_COUNT 4 /* number of measures to loose supervisory signal */
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#define SUPER_DETECT_COUNT 6 /* number of measures to detect supervisory signal */
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#define MUTE_DURATION 0.280 /* a tiny bit more than two frames */
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/* two supervisory tones */
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static double super_freq[5] = {
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3955.0, /* 0-Signal 1 */
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3985.0, /* 0-Signal 2 */
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4015.0, /* 0-Signal 3 */
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4045.0, /* 0-Signal 4 */
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3895.0, /* noise level to check against */
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};
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/* table for fast sine generation */
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static sample_t dsp_sine_super[65536];
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static sample_t dsp_sine_dialtone[65536];
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/* global init for dsp */
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void dsp_init(void)
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{
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int i;
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double s;
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LOGP(DDSP, LOGL_DEBUG, "Generating sine table for supervisory signal and dial tone.\n");
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for (i = 0; i < 65536; i++) {
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s = sin((double)i / 65536.0 * 2.0 * PI);
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/* supervisor sine */
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dsp_sine_super[i] = s * TX_PEAK_SUPER;
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/* dialtone sine */
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dsp_sine_dialtone[i] = s * TX_PEAK_DIALTONE;
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}
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compandor_init();
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}
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static int fsk_send_bit(void *inst);
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static void fsk_receive_bit(void *inst, int bit, double quality, double level);
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/* Init FSK of transceiver */
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int dsp_init_sender(nmt_t *nmt, double deviation_factor)
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{
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sample_t *spl;
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int i;
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/* attack (3ms) and recovery time (13.5ms) according to NMT specs */
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setup_compandor(&nmt->cstate, 8000, 3.0, 13.5);
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LOGP_CHAN(DDSP, LOGL_DEBUG, "Init DSP for Transceiver.\n");
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/* set modulation parameters */
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sender_set_fm(&nmt->sender, MAX_DEVIATION * deviation_factor, MAX_MODULATION * deviation_factor, SPEECH_DEVIATION * deviation_factor, MAX_DISPLAY);
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LOGP(DDSP, LOGL_DEBUG, "Using FSK level of %.3f (%.3f KHz deviation @ 1500 Hz)\n", TX_PEAK_FSK * deviation_factor, 3.5 * deviation_factor);
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LOGP(DDSP, LOGL_DEBUG, "Using Supervisory level of %.3f (%.3f KHz deviation @ 4015 Hz)\n", TX_PEAK_SUPER * deviation_factor, 0.3 * deviation_factor);
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/* init fsk */
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if (fsk_mod_init(&nmt->fsk_mod, nmt, fsk_send_bit, nmt->sender.samplerate, BIT_RATE, F0, F1, TX_PEAK_FSK, 1, 0) < 0) {
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LOGP_CHAN(DDSP, LOGL_ERROR, "FSK init failed!\n");
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return -EINVAL;
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}
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if (fsk_demod_init(&nmt->fsk_demod, nmt, fsk_receive_bit, nmt->sender.samplerate, BIT_RATE, F0, F1, BIT_ADJUST) < 0) {
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LOGP_CHAN(DDSP, LOGL_ERROR, "FSK init failed!\n");
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return -EINVAL;
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}
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/* allocate ring buffer for SAT signal detection
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* the bandwidth of the Goertzel filter is the reciprocal of the duration
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* we half our bandwidth, so that other supervisory signals will be canceled out completely by goertzel filter
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*/
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nmt->super_samples = (int)((double)nmt->sender.samplerate * (1.0 / (SUPER_BANDWIDTH / 2)) + 0.5);
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spl = calloc(1, nmt->super_samples * sizeof(*spl));
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if (!spl) {
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LOGP(DDSP, LOGL_ERROR, "No memory!\n");
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return -ENOMEM;
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}
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nmt->super_filter_spl = spl;
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/* count supervidory tones */
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for (i = 0; i < 5; i++) {
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audio_goertzel_init(&nmt->super_goertzel[i], super_freq[i], nmt->sender.samplerate);
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if (i < 4)
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nmt->super_phaseshift65536[i] = 65536.0 / ((double)nmt->sender.samplerate / super_freq[i]);
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}
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super_reset(nmt);
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/* dial tone */
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nmt->dial_phaseshift65536 = 65536.0 / ((double)nmt->sender.samplerate / DIALTONE_HZ);
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/* dtmf, generate tone relative to speech level */
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dtmf_encode_init(&nmt->dtmf, 8000, 1.0 / SPEECH_LEVEL);
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nmt->dmp_frame_level = display_measurements_add(&nmt->sender.dispmeas, "Frame Level", "%.1f %% (last)", DISPLAY_MEAS_LAST, DISPLAY_MEAS_LEFT, 0.0, 150.0, 100.0);
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nmt->dmp_frame_quality = display_measurements_add(&nmt->sender.dispmeas, "Frame Quality", "%.1f %% (last)", DISPLAY_MEAS_LAST, DISPLAY_MEAS_LEFT, 0.0, 100.0, 100.0);
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if (nmt->sysinfo.chan_type == CHAN_TYPE_TC || nmt->sysinfo.chan_type == CHAN_TYPE_AC_TC || nmt->sysinfo.chan_type == CHAN_TYPE_CC_TC) {
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nmt->dmp_super_level = display_measurements_add(&nmt->sender.dispmeas, "Super Level", "%.1f %%", DISPLAY_MEAS_AVG, DISPLAY_MEAS_LEFT, 0.0, 150.0, 100.0);
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nmt->dmp_super_quality = display_measurements_add(&nmt->sender.dispmeas, "Super Quality", "%.1f %%", DISPLAY_MEAS_AVG, DISPLAY_MEAS_LEFT, 0.0, 100.0, 100.0);
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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(nmt_t *nmt)
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{
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LOGP_CHAN(DDSP, LOGL_DEBUG, "Cleanup DSP for Transceiver.\n");
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fsk_mod_cleanup(&nmt->fsk_mod);
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fsk_demod_cleanup(&nmt->fsk_demod);
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if (nmt->super_filter_spl) {
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free(nmt->super_filter_spl);
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nmt->super_filter_spl = NULL;
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}
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}
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/* Check for SYNC bits, then collect data bits */
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static void fsk_receive_bit(void *inst, int bit, double quality, double level)
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{
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nmt_t *nmt = (nmt_t *)inst;
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uint64_t frames_elapsed;
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int i;
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/* normalize FSK level */
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level /= TX_PEAK_FSK;
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nmt->rx_bits_count++;
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if (nmt->trans && nmt->trans->dms_call)
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fsk_receive_bit_dms(nmt, bit, quality, level);
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// printf("bit=%d quality=%.4f\n", bit, quality);
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if (!nmt->rx_in_sync) {
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nmt->rx_sync = (nmt->rx_sync << 1) | bit;
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/* level and quality */
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nmt->rx_level[nmt->rx_count & 0xff] = level;
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nmt->rx_quality[nmt->rx_count & 0xff] = quality;
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nmt->rx_count++;
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/* check if pattern 1010111100010010 matches */
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if (nmt->rx_sync != 0xaf12)
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return;
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/* average level and quality */
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level = quality = 0;
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for (i = 0; i < 16; i++) {
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level += nmt->rx_level[(nmt->rx_count - 1 - i) & 0xff];
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quality += nmt->rx_quality[(nmt->rx_count - 1 - i) & 0xff];
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}
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level /= 16.0; quality /= 16.0;
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// printf("sync (level = %.2f, quality = %.2f\n", level, quality);
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/* do not accept garbage */
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if (quality < 0.65)
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return;
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/* sync time */
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nmt->rx_bits_count_last = nmt->rx_bits_count_current;
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nmt->rx_bits_count_current = nmt->rx_bits_count - 26.0;
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/* rest sync register */
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nmt->rx_sync = 0;
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nmt->rx_in_sync = 1;
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nmt->rx_count = 0;
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/* set muting of receive path */
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nmt->rx_mute = (int)((double)nmt->sender.samplerate * MUTE_DURATION);
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return;
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}
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/* read bits */
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nmt->rx_frame[nmt->rx_count] = bit + '0';
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nmt->rx_level[nmt->rx_count] = level;
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nmt->rx_quality[nmt->rx_count] = quality;
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if (++nmt->rx_count != 140)
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return;
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/* end of frame */
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nmt->rx_frame[140] = '\0';
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nmt->rx_in_sync = 0;
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/* average level and quality */
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level = quality = 0;
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for (i = 0; i < 140; i++) {
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level += nmt->rx_level[i];
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quality += nmt->rx_quality[i];
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}
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level /= 140.0; quality /= 140.0;
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/* update measurements */
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display_measurements_update(nmt->dmp_frame_level, level * 100.0, 0.0);
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display_measurements_update(nmt->dmp_frame_quality, quality * 100.0, 0.0);
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/* send telegramm */
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frames_elapsed = (nmt->rx_bits_count_current - nmt->rx_bits_count_last + 83) / 166; /* round to nearest frame */
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/* convert level so that received level at TX_PEAK_FSK results in 1.0 (100%) */
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nmt_receive_frame(nmt, nmt->rx_frame, quality, level, frames_elapsed);
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}
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/* compare supervisory signal against noise floor around 3895 Hz */
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static void super_decode(nmt_t *nmt, sample_t *samples, int length)
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{
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double result[2], level, quality;
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audio_goertzel(&nmt->super_goertzel[nmt->supervisory - 1], samples, length, 0, &result[0], 1);
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audio_goertzel(&nmt->super_goertzel[4], samples, length, 0, &result[1], 1); /* noise floor detection */
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/* normalize supervisory level */
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level = result[0] / TX_PEAK_SUPER;
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quality = (result[0] - result[1]) / result[0];
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if (quality < 0)
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quality = 0;
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if (nmt->state == STATE_ACTIVE) {
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if (++nmt->super_print == SUPER_PRINT) {
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nmt->super_print = 0;
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LOGP_CHAN(DDSP, LOGL_NOTICE, "Supervisory level %.0f%% quality %.0f%%\n", level * 100.0, quality * 100.0);
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}
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/* update measurements (if dmp_* params are NULL, we omit this) */
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display_measurements_update(nmt->dmp_super_level, level * 100.0, 0.0);
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display_measurements_update(nmt->dmp_super_quality, quality * 100.0, 0.0);
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}
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if (quality > 0.7) {
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if (nmt->super_detected == 0) {
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nmt->super_detect_count++;
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if (nmt->super_detect_count == SUPER_DETECT_COUNT) {
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nmt->super_detected = 1;
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nmt->super_detect_count = 0;
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LOGP_CHAN(DDSP, LOGL_DEBUG, "Supervisory signal detected with level=%.0f%%, quality=%.0f%%.\n", result[0] / 0.63662 / TX_PEAK_SUPER * 100.0, quality * 100.0);
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nmt_rx_super(nmt, 1, quality);
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}
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} else
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nmt->super_detect_count = 0;
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} else {
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if (nmt->super_detected == 1) {
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nmt->super_detect_count++;
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if (nmt->super_detect_count == SUPER_LOST_COUNT) {
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nmt->super_detected = 0;
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nmt->super_detect_count = 0;
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LOGP_CHAN(DDSP, LOGL_DEBUG, "Supervisory signal lost.\n");
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nmt_rx_super(nmt, 0, 0.0);
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}
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} else
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nmt->super_detect_count = 0;
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}
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}
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/* Reset supervisory detection states, so ongoing tone will be detected again. */
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void super_reset(nmt_t *nmt)
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{
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LOGP_CHAN(DDSP, LOGL_DEBUG, "Supervisory detector reset.\n");
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nmt->super_detected = 0;
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nmt->super_detect_count = 0;
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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, double __attribute__((unused)) rf_level_db)
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{
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nmt_t *nmt = (nmt_t *) sender;
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sample_t *spl;
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int max, pos;
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int i;
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/* write received samples to decode buffer */
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max = nmt->super_samples;
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spl = nmt->super_filter_spl;
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pos = nmt->super_filter_pos;
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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 (nmt->supervisory)
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super_decode(nmt, spl, max);
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}
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}
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nmt->super_filter_pos = pos;
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/* fsk signal */
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fsk_demod_receive(&nmt->fsk_demod, samples, length);
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/* muting audio while receiving frame */
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for (i = 0; i < length; i++) {
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if (nmt->rx_mute && !nmt->sender.loopback) {
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samples[i] = 0;
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nmt->rx_mute--;
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}
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}
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if ((nmt->dsp_mode == DSP_MODE_AUDIO || nmt->dsp_mode == DSP_MODE_DTMF)
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&& nmt->trans && nmt->trans->callref) {
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int len, count;
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len = samplerate_downsample(&nmt->sender.srstate, samples, length);
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if (nmt->compandor)
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expand_audio(&nmt->cstate, samples, len);
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if (nmt->dsp_mode == DSP_MODE_DTMF) {
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/* encode and fill with silence after finish */
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count = dtmf_encode(&nmt->dtmf, samples, len);
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if (count < len)
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memset(samples + count, 0, sizeof(*samples) * (len - count));
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}
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spl = nmt->sender.rxbuf;
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pos = nmt->sender.rxbuf_pos;
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for (i = 0; i < len; i++) {
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spl[pos++] = samples[i];
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if (pos == 160) {
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call_up_audio(nmt->trans->callref, spl, 160);
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pos = 0;
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}
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}
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nmt->sender.rxbuf_pos = pos;
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} else
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nmt->sender.rxbuf_pos = 0;
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}
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static int fsk_send_bit(void *inst)
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{
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nmt_t *nmt = (nmt_t *)inst;
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const char *frame;
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/* send frame bit (prio) */
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if (nmt->dsp_mode == DSP_MODE_FRAME) {
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if (!nmt->tx_frame_length || nmt->tx_frame_pos == nmt->tx_frame_length) {
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/* request frame */
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frame = nmt_get_frame(nmt);
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if (!frame) {
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nmt->tx_frame_length = 0;
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LOGP_CHAN(DDSP, LOGL_DEBUG, "Stop sending frames.\n");
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return -1;
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}
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memcpy(nmt->tx_frame, frame, 166);
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nmt->tx_frame_length = 166;
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nmt->tx_frame_pos = 0;
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}
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return nmt->tx_frame[nmt->tx_frame_pos++];
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}
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/* send dms bit */
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return dms_send_bit(nmt);
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}
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/* Generate audio stream with supervisory signal. Keep phase for next call of function. */
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static void super_encode(nmt_t *nmt, sample_t *samples, int length)
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{
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double phaseshift, phase;
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int i;
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phaseshift = nmt->super_phaseshift65536[nmt->supervisory - 1];
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phase = nmt->super_phase65536;
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for (i = 0; i < length; i++) {
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*samples++ += dsp_sine_super[(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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nmt->super_phase65536 = phase;
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}
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/* Generate audio stream from dial tone. Keep phase for next call of function. */
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static void dial_tone(nmt_t *nmt, sample_t *samples, int length)
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{
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double phaseshift, phase;
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int i;
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phaseshift = nmt->dial_phaseshift65536;
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phase = nmt->dial_phase65536;
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for (i = 0; i < length; i++) {
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*samples++ = dsp_sine_dialtone[(uint16_t)phase];
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phase += phaseshift;
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if (phase >= 65536)
|
|
phase -= 65536;
|
|
}
|
|
|
|
nmt->dial_phase65536 = phase;
|
|
}
|
|
|
|
/* Provide stream of audio toward radio unit */
|
|
void sender_send(sender_t *sender, sample_t *samples, uint8_t *power, int length)
|
|
{
|
|
nmt_t *nmt = (nmt_t *) sender;
|
|
int count, input_num;
|
|
|
|
memset(power, 1, length);
|
|
|
|
again:
|
|
switch (nmt->dsp_mode) {
|
|
case DSP_MODE_AUDIO:
|
|
case DSP_MODE_DTMF:
|
|
input_num = samplerate_upsample_input_num(&sender->srstate, length);
|
|
{
|
|
int16_t spl[input_num];
|
|
jitter_load_samples(&sender->dejitter, (uint8_t *)spl, input_num, sizeof(*spl), jitter_conceal_s16, NULL);
|
|
int16_to_samples_speech(samples, spl, input_num);
|
|
}
|
|
if (nmt->compandor)
|
|
compress_audio(&nmt->cstate, samples, input_num);
|
|
samplerate_upsample(&sender->srstate, samples, input_num, samples, length);
|
|
/* send after dejitter, so audio is flushed */
|
|
if (nmt->dms.tx_frame_valid) {
|
|
fsk_mod_send(&nmt->fsk_mod, samples, length, 0);
|
|
break;
|
|
}
|
|
if (nmt->supervisory)
|
|
super_encode(nmt, samples, length);
|
|
break;
|
|
case DSP_MODE_DIALTONE:
|
|
dial_tone(nmt, samples, length);
|
|
break;
|
|
case DSP_MODE_SILENCE:
|
|
memset(samples, 0, length * sizeof(*samples));
|
|
break;
|
|
case DSP_MODE_FRAME:
|
|
/* Encode frame into audio stream. If frames have
|
|
* stopped, process again for rest of stream. */
|
|
count = fsk_mod_send(&nmt->fsk_mod, samples, length, 0);
|
|
/* special case: add supervisory signal to frame at loop test */
|
|
if (nmt->sender.loopback && nmt->supervisory)
|
|
super_encode(nmt, samples, count);
|
|
samples += count;
|
|
length -= count;
|
|
if (length)
|
|
goto again;
|
|
break;
|
|
}
|
|
}
|
|
|
|
static const char *nmt_dsp_mode_name(enum dsp_mode mode)
|
|
{
|
|
static char invalid[16];
|
|
|
|
switch (mode) {
|
|
case DSP_MODE_SILENCE:
|
|
return "SILENCE";
|
|
case DSP_MODE_DIALTONE:
|
|
return "DIALTONE";
|
|
case DSP_MODE_AUDIO:
|
|
return "AUDIO";
|
|
case DSP_MODE_FRAME:
|
|
return "FRAME";
|
|
case DSP_MODE_DTMF:
|
|
return "DTMF";
|
|
}
|
|
|
|
sprintf(invalid, "invalid(%d)", mode);
|
|
return invalid;
|
|
}
|
|
|
|
void nmt_set_dsp_mode(nmt_t *nmt, enum dsp_mode mode)
|
|
{
|
|
/* reset frame */
|
|
if (mode == DSP_MODE_FRAME && nmt->dsp_mode != mode) {
|
|
fsk_mod_reset(&nmt->fsk_mod);
|
|
nmt->tx_frame_length = 0;
|
|
}
|
|
|
|
LOGP_CHAN(DDSP, LOGL_DEBUG, "DSP mode %s -> %s\n", nmt_dsp_mode_name(nmt->dsp_mode), nmt_dsp_mode_name(mode));
|
|
if ((mode == DSP_MODE_AUDIO || mode == DSP_MODE_DTMF) && (nmt->dsp_mode != DSP_MODE_AUDIO && nmt->dsp_mode != DSP_MODE_DTMF))
|
|
jitter_reset(&nmt->sender.dejitter);
|
|
|
|
nmt->dsp_mode = mode;
|
|
}
|
|
|
|
/* Receive audio from call instance. */
|
|
void call_down_audio(void *decoder, void *decoder_priv, int callref, uint16_t sequence, uint8_t marker, uint32_t timestamp, uint32_t ssrc, uint8_t *payload, int payload_len)
|
|
{
|
|
transaction_t *trans;
|
|
nmt_t *nmt;
|
|
|
|
trans = get_transaction_by_callref(callref);
|
|
if (!trans)
|
|
return;
|
|
nmt = trans->nmt;
|
|
if (!nmt)
|
|
return;
|
|
|
|
if (nmt->dsp_mode == DSP_MODE_AUDIO || nmt->dsp_mode == DSP_MODE_DTMF) {
|
|
jitter_frame_t *jf;
|
|
jf = jitter_frame_alloc(decoder, decoder_priv, payload, payload_len, marker, sequence, timestamp, ssrc);
|
|
if (jf)
|
|
jitter_save(&nmt->sender.dejitter, jf);
|
|
}
|
|
}
|
|
|
|
void call_down_clock(void) {}
|
|
|