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

733 lines
26 KiB

/* main function
*
* (C) 2018 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/>.
*/
#include <stdio.h>
#include <stdint.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include <errno.h>
#include <pthread.h>
#include "../libsample/sample.h"
#include "../libdebug/debug.h"
#include "../libsound/sound.h"
#include "../libclipper/clipper.h"
#include "radio.h"
#define CLIP_POINT 0.85
#define DC_CUTOFF 30.0 // Wikipedia: UKW-Rundfunk
#define STEREO_BW 15000.0
#define PILOT_FREQ 19000.0
#define PILOT_BW 5.0
int radio_init(radio_t *radio, int latspl, int samplerate, const char *tx_wave_file, const char *rx_wave_file, const char *tx_audiodev, const char *rx_audiodev, enum modulation modulation, double bandwidth, double deviation, double modulation_index, double time_constant_us, double volume, int stereo, int rds, int rds2)
{
int rc = -EINVAL;
clipper_init(CLIP_POINT);
memset(radio, 0, sizeof(*radio));
radio->latspl = latspl;
radio->volume = volume;
radio->stereo = stereo;
radio->rds = rds;
radio->rds2 = rds2;
radio->tx_wave_file = tx_wave_file;
radio->modulation = modulation;
radio->signal_samplerate = samplerate;
radio->audio_bandwidth = bandwidth;
switch (radio->modulation) {
case MODULATION_FM:
radio->fm_deviation = deviation;
radio->signal_bandwidth = deviation + bandwidth;
if (radio->stereo) {
radio->signal_bandwidth = deviation + 53000.0;
radio->audio_bandwidth = STEREO_BW;
}
if (radio->rds)
radio->signal_bandwidth = deviation + 60000.0;
if (radio->rds2)
radio->signal_bandwidth = deviation + 80000.0;
break;
case MODULATION_AM_DSB:
case MODULATION_AM_USB:
case MODULATION_AM_LSB:
/* level is 1.0, which is full amplitude */
radio->signal_bandwidth = bandwidth;
break;
case MODULATION_NONE:
PDEBUG(DRADIO, DEBUG_ERROR, "Wrong modulation, plese fix!\n");
goto error;
}
if (tx_wave_file) {
/* open wave file */
int _samplerate = 0;
radio->tx_audio_channels = 0;
rc = wave_create_playback(&radio->wave_tx_play, tx_wave_file, &_samplerate, &radio->tx_audio_channels, 1.0);
if (rc < 0) {
PDEBUG(DRADIO, DEBUG_ERROR, "Failed to create WAVE playback instance!\n");
goto error;
}
if (radio->tx_audio_channels != 1 && radio->tx_audio_channels != 2)
{
PDEBUG(DRADIO, DEBUG_ERROR, "WAVE file must have one or two channels!\n");
goto error;
}
radio->tx_audio_samplerate = _samplerate;
radio->tx_audio_mode = AUDIO_MODE_WAVEFILE;
} else if (tx_audiodev) {
#ifdef HAVE_ALSA
/* open audio device */
radio->tx_audio_samplerate = 48000;
radio->tx_audio_channels = (stereo) ? 2 : 1;
radio->tx_sound = sound_open(tx_audiodev, NULL, NULL, radio->tx_audio_channels, 0.0, radio->tx_audio_samplerate, radio->latspl, 1.0, 0.0);
if (!radio->tx_sound) {
rc = -EIO;
PDEBUG(DRADIO, DEBUG_ERROR, "Failed to open sound device!\n");
goto error;
}
jitter_create(&radio->tx_dejitter[0], radio->tx_audio_samplerate / 5);
jitter_create(&radio->tx_dejitter[1], radio->tx_audio_samplerate / 5);
radio->tx_audio_mode = AUDIO_MODE_AUDIODEV;
#else
rc = -ENOTSUP;
PDEBUG(DRADIO, DEBUG_ERROR, "No sound card support compiled in!\n");
goto error;
#endif
} else {
int i;
double phase;
/* use built-in sample sound */
radio->tx_audio_samplerate = samplerate;
radio->tx_audio_channels = (radio->stereo) ? 2 : 1;
radio->testtone_length = radio->tx_audio_samplerate;
radio->testtone[0] = calloc(radio->testtone_length * 2, sizeof(sample_t));
if (!radio->testtone[0]) {
rc = -ENOMEM;
PDEBUG(DRADIO, DEBUG_ERROR, "Failed to allocate test sound buffer!\n");
goto error;
}
radio->testtone[1] = radio->testtone[0] + radio->testtone_length;
/* generate tone */
phase = 2.0 * M_PI * 1000.0 / radio->tx_audio_samplerate;
if (radio->stereo) {
for (i = 0; i < radio->testtone_length / 2; i++) {
radio->testtone[0][i] = sin(i * phase);
radio->testtone[1][i] = 0.0;
}
for (; i < radio->testtone_length; i++) {
radio->testtone[0][i] = 0.0;
radio->testtone[1][i] = sin(i * phase);
}
} else {
for (i = 0; i < radio->testtone_length; i++) {
radio->testtone[0][i] = sin(i * phase);
}
}
radio->tx_audio_mode = AUDIO_MODE_TESTTONE;
}
if (rx_wave_file) {
/* open wave file */
radio->rx_audio_samplerate = 48000;
radio->rx_audio_channels = (radio->stereo) ? 2 : 1;
rc = wave_create_record(&radio->wave_rx_rec, rx_wave_file, radio->rx_audio_samplerate, radio->rx_audio_channels, 1.0);
if (rc < 0) {
PDEBUG(DRADIO, DEBUG_ERROR, "Failed to create WAVE record instance!\n");
goto error;
}
radio->rx_audio_mode = AUDIO_MODE_WAVEFILE;
} else if (rx_audiodev) {
#ifdef HAVE_ALSA
/* open audio device */
radio->rx_audio_samplerate = 48000;
radio->rx_audio_channels = (stereo) ? 2 : 1;
/* check if we use same device */
if (radio->tx_sound && !strcmp(tx_audiodev, rx_audiodev))
radio->rx_sound = radio->tx_sound;
else
radio->rx_sound = sound_open(rx_audiodev, NULL, NULL, radio->rx_audio_channels, 0.0, radio->rx_audio_samplerate, radio->latspl, 1.0, 0.0);
if (!radio->rx_sound) {
rc = -EIO;
PDEBUG(DRADIO, DEBUG_ERROR, "Failed to open sound device!\n");
goto error;
}
jitter_create(&radio->rx_dejitter[0], radio->rx_audio_samplerate / 5);
jitter_create(&radio->rx_dejitter[1], radio->rx_audio_samplerate / 5);
radio->rx_audio_mode = AUDIO_MODE_AUDIODEV;
#else
rc = -ENOTSUP;
PDEBUG(DRADIO, DEBUG_ERROR, "No sound card support compiled in!\n");
goto error;
#endif
#if 0
} else {
rc = -ENOTSUP;
PDEBUG(DRADIO, DEBUG_ERROR, "No RX audio sink is selected, try \"--audio-device default\"!\n");
goto error;
#endif
}
/* check if sample rate is too low */
if (radio->tx_audio_samplerate > radio->signal_samplerate) {
rc = -EINVAL;
PDEBUG(DRADIO, DEBUG_ERROR, "You have selected a signal processing sample rate of %.0f. Your audio sample rate is %.0f.\n", radio->signal_samplerate, radio->tx_audio_samplerate);
PDEBUG(DRADIO, DEBUG_ERROR, "Please select a sample rate that is higher or equal the audio sample rate!\n");
goto error;
}
if (radio->rx_audio_samplerate > radio->signal_samplerate) {
rc = -EINVAL;
PDEBUG(DRADIO, DEBUG_ERROR, "You have selected a signal processing sample rate of %.0f. Your audio sample rate is %.0f.\n", radio->signal_samplerate, radio->rx_audio_samplerate);
PDEBUG(DRADIO, DEBUG_ERROR, "Please select a sample rate that is higher or equal the audio sample rate!\n");
goto error;
}
if (radio->signal_samplerate < radio->signal_bandwidth * 2 / 0.75) {
rc = -EINVAL;
PDEBUG(DRADIO, DEBUG_ERROR, "You have selected a signal processing sample rate of %.0f. Your signal's bandwidth %.0f.\n", radio->signal_samplerate, radio->signal_bandwidth);
PDEBUG(DRADIO, DEBUG_ERROR, "Your signal processing sample rate must be at least one third greater than the signal's double bandwidth. Use at least %.0f.\n", radio->signal_bandwidth * 2.0 / 0.75);
goto error;
}
iir_highpass_init(&radio->tx_dc_removal[0], DC_CUTOFF, radio->tx_audio_samplerate, 1);
iir_highpass_init(&radio->tx_dc_removal[1], DC_CUTOFF, radio->tx_audio_samplerate, 1);
/* stereo pilot tone phase */
radio->pilot_phasestep = 2.0 * M_PI * PILOT_FREQ / radio->signal_samplerate;
/* stere decoding filters */
iir_lowpass_init(&radio->rx_lp_pilot_I, PILOT_BW, radio->signal_samplerate, 2);
iir_lowpass_init(&radio->rx_lp_pilot_Q, PILOT_BW, radio->signal_samplerate, 2);
iir_lowpass_init(&radio->rx_lp_sum, STEREO_BW, radio->signal_samplerate, 2);
iir_lowpass_init(&radio->rx_lp_diff, STEREO_BW, radio->signal_samplerate, 2);
/* init sample rate conversion, use complete bandwidth for resample filter */
rc = init_samplerate(&radio->tx_resampler[0], radio->tx_audio_samplerate, radio->signal_samplerate, radio->tx_audio_samplerate / 2.0);
if (rc < 0)
goto error;
rc = init_samplerate(&radio->tx_resampler[1], radio->tx_audio_samplerate, radio->signal_samplerate, radio->tx_audio_samplerate / 2.0);
if (rc < 0)
goto error;
rc = init_samplerate(&radio->rx_resampler[0], radio->rx_audio_samplerate, radio->signal_samplerate, radio->rx_audio_samplerate / 2.0);
if (rc < 0)
goto error;
rc = init_samplerate(&radio->rx_resampler[1], radio->rx_audio_samplerate, radio->signal_samplerate, radio->rx_audio_samplerate / 2.0);
if (rc < 0)
goto error;
/* init filters (using signal sample rate) */
switch (radio->modulation) {
case MODULATION_FM:
/* time constant */
PDEBUG(DRADIO, DEBUG_INFO, "Using emphasis cut-off at %.0f Hz.\n", timeconstant2cutoff(time_constant_us));
rc = init_emphasis(&radio->fm_emphasis[0], radio->signal_samplerate, timeconstant2cutoff(time_constant_us), DC_CUTOFF, radio->audio_bandwidth);
if (rc < 0)
goto error;
rc = init_emphasis(&radio->fm_emphasis[1], radio->signal_samplerate, timeconstant2cutoff(time_constant_us), DC_CUTOFF, radio->audio_bandwidth);
if (rc < 0)
goto error;
rc = fm_mod_init(&radio->fm_mod, radio->signal_samplerate, 0.0, 1.0);
if (rc < 0)
goto error;
rc = fm_demod_init(&radio->fm_demod, radio->signal_samplerate, 0.0, 2 * radio->signal_bandwidth);
if (rc < 0)
goto error;
break;
case MODULATION_AM_DSB:
iir_lowpass_init(&radio->tx_am_bw_limit, radio->audio_bandwidth, radio->signal_samplerate, 1);
/* modulation index 0.0 = no envelope, bias 1.0
* modulation index 1.0 = envelope +-0.5, bias 0.5
* modulation index 0.5 = envelope +-0.25, bias 0.75
*/
double gain = modulation_index / 2.0;
double bias = 1.0 - gain;
rc = am_mod_init(&radio->am_mod, radio->signal_samplerate, 0.0, gain, bias);
if (rc < 0)
goto error;
rc = am_demod_init(&radio->am_demod, radio->signal_samplerate, 0.0, radio->signal_bandwidth, 1.0 / modulation_index);
if (rc < 0)
goto error;
break;
case MODULATION_AM_USB:
iir_lowpass_init(&radio->tx_am_bw_limit, radio->audio_bandwidth, radio->signal_samplerate, 1);
rc = am_mod_init(&radio->am_mod, radio->signal_samplerate, 0.0, 1.0, 0.0);
if (rc < 0)
goto error;
break;
case MODULATION_AM_LSB:
iir_lowpass_init(&radio->tx_am_bw_limit, radio->audio_bandwidth, radio->signal_samplerate, 1);
rc = am_mod_init(&radio->am_mod, radio->signal_samplerate, 0.0, 1.0, 0.0);
if (rc < 0)
goto error;
break;
default:
break;
}
if (radio->tx_audio_mode)
PDEBUG(DRADIO, DEBUG_INFO, "Bandwidth of audio source is %.0f Hz.\n", radio->tx_audio_samplerate / 2.0);
if (radio->rx_audio_mode)
PDEBUG(DRADIO, DEBUG_INFO, "Bandwidth of audio sink is %.0f Hz.\n", radio->rx_audio_samplerate / 2.0);
PDEBUG(DRADIO, DEBUG_INFO, "Bandwidth of audio signal is %.0f Hz.\n", radio->audio_bandwidth);
PDEBUG(DRADIO, DEBUG_INFO, "Bandwidth of modulated signal is %.0f Hz.\n", radio->signal_bandwidth);
if (radio->tx_audio_mode)
PDEBUG(DRADIO, DEBUG_INFO, "Sample rate of audio source is %.0f Hz.\n", radio->tx_audio_samplerate);
if (radio->rx_audio_mode)
PDEBUG(DRADIO, DEBUG_INFO, "Sample rate of audio sink is %.0f Hz.\n", radio->rx_audio_samplerate);
PDEBUG(DRADIO, DEBUG_INFO, "Sample rate of signal is %.0f Hz.\n", radio->signal_samplerate);
/* one or two audio channels */
if (radio->tx_audio_channels != 1 && radio->tx_audio_channels != 2)
{
PDEBUG(DRADIO, DEBUG_ERROR, "Wrong number of audio channels, please fix!\n");
goto error;
}
/* audio buffers: how many sample for audio (rounded down) */
int tx_size = (int)((double)latspl / radio->tx_resampler[0].factor);
int rx_size = (int)((double)latspl / radio->rx_resampler[0].factor);
if (tx_size > rx_size)
radio->audio_buffer_size = tx_size;
else
radio->audio_buffer_size = rx_size;
radio->audio_buffer = calloc(radio->audio_buffer_size * 2, sizeof(*radio->audio_buffer));
if (!radio->audio_buffer) {
PDEBUG(DRADIO, DEBUG_ERROR, "No memory!!\n");
rc = -ENOMEM;
goto error;
}
/* signal buffers */
radio->signal_buffer_size = latspl;
radio->signal_buffer = calloc(radio->signal_buffer_size * 3, sizeof(*radio->signal_buffer));
radio->signal_power_buffer = calloc(radio->signal_buffer_size, sizeof(*radio->signal_power_buffer));
if (!radio->signal_buffer || !radio->signal_power_buffer) {
PDEBUG(DRADIO, DEBUG_ERROR, "No memory!!\n");
rc = -ENOMEM;
goto error;
}
/* termporary I/Q/carrier buffers, used while demodulating */
radio->I_buffer = calloc(latspl, sizeof(*radio->I_buffer));
radio->Q_buffer = calloc(latspl, sizeof(*radio->Q_buffer));
radio->carrier_buffer = calloc(latspl, sizeof(*radio->carrier_buffer));
if (!radio->I_buffer || !radio->Q_buffer || !radio->carrier_buffer) {
PDEBUG(DRADIO, DEBUG_ERROR, "No memory!!\n");
rc = -ENOMEM;
goto error;
}
return 0;
error:
radio_exit(radio);
return rc;
}
void radio_exit(radio_t *radio)
{
if (radio->audio_buffer) {
free(radio->audio_buffer);
radio->audio_buffer = NULL;
}
if (radio->signal_buffer) {
free(radio->signal_buffer);
radio->signal_buffer = NULL;
}
if (radio->signal_power_buffer) {
free(radio->signal_power_buffer);
radio->signal_power_buffer = NULL;
}
if (radio->I_buffer) {
free(radio->I_buffer);
radio->I_buffer = NULL;
}
if (radio->Q_buffer) {
free(radio->Q_buffer);
radio->Q_buffer = NULL;
}
if (radio->carrier_buffer) {
free(radio->carrier_buffer);
radio->carrier_buffer = NULL;
}
if (radio->tx_audio_mode == AUDIO_MODE_WAVEFILE) {
wave_destroy_playback(&radio->wave_tx_play);
radio->tx_audio_mode = AUDIO_MODE_NONE;
}
if (radio->rx_audio_mode == AUDIO_MODE_WAVEFILE) {
wave_destroy_record(&radio->wave_rx_rec);
radio->rx_audio_mode = AUDIO_MODE_NONE;
}
#ifdef HAVE_ALSA
if (radio->tx_sound) {
sound_close(radio->tx_sound);
/* if same device was used */
if (radio->tx_sound == radio->rx_sound)
radio->rx_sound = NULL;
radio->tx_sound = NULL;
radio->tx_audio_mode = AUDIO_MODE_NONE;
}
if (radio->rx_sound) {
sound_close(radio->rx_sound);
radio->rx_sound = NULL;
radio->rx_audio_mode = AUDIO_MODE_NONE;
}
#endif
jitter_destroy(&radio->tx_dejitter[0]);
jitter_destroy(&radio->tx_dejitter[1]);
jitter_destroy(&radio->rx_dejitter[0]);
jitter_destroy(&radio->rx_dejitter[1]);
if (radio->tx_audio_mode == AUDIO_MODE_TESTTONE) {
free(radio->testtone[0]);
radio->tx_audio_mode = AUDIO_MODE_NONE;
}
if (radio->modulation == MODULATION_FM)
fm_mod_exit(&radio->fm_mod);
else
am_mod_exit(&radio->am_mod);
}
int radio_start(radio_t __attribute__((unused)) *radio)
{
int rc = 0;
#ifdef HAVE_ALSA
/* start rx sound */
if (radio->rx_sound)
rc = sound_start(radio->rx_sound);
/* start tx sound, if different device */
if (radio->tx_sound && radio->tx_sound != radio->rx_sound)
rc = sound_start(radio->tx_sound);
#endif
return rc;
}
int radio_tx(radio_t *radio, float *baseband, int signal_num)
{
int i;
int __attribute__((unused)) rc;
int audio_num;
sample_t *audio_samples[2];
sample_t *signal_samples[3];
uint8_t *signal_power;
if (signal_num > radio->latspl) {
PDEBUG(DRADIO, DEBUG_ERROR, "signal_num > latspl, please fix!.\n");
abort();
}
/* audio buffers: how many sample for audio (rounded down) */
audio_num = (int)((double)signal_num / radio->tx_resampler[0].factor);
if (audio_num > radio->audio_buffer_size) {
PDEBUG(DRADIO, DEBUG_ERROR, "audio_num > audio_buffer_size, please fix!.\n");
abort();
}
audio_samples[0] = radio->audio_buffer;
audio_samples[1] = radio->audio_buffer + radio->audio_buffer_size;
/* signal buffers: a bit more samples to be safe */
signal_num = (int)((double)audio_num * radio->tx_resampler[0].factor + 0.5) + 10;
if (signal_num > radio->signal_buffer_size) {
PDEBUG(DRADIO, DEBUG_ERROR, "signal_num > signal_buffer_size, please fix!.\n");
abort();
}
signal_samples[0] = radio->signal_buffer;
signal_samples[1] = radio->signal_buffer + radio->signal_buffer_size;
signal_samples[2] = radio->signal_buffer + radio->signal_buffer_size * 2;
signal_power = radio->signal_power_buffer;
/* get audio to be sent */
switch (radio->tx_audio_mode) {
case AUDIO_MODE_WAVEFILE:
wave_read(&radio->wave_tx_play, audio_samples, audio_num);
if (!radio->wave_tx_play.left) {
int rc;
int _samplerate = 0;
wave_destroy_playback(&radio->wave_tx_play);
rc = wave_create_playback(&radio->wave_tx_play, radio->tx_wave_file, &_samplerate, &radio->tx_audio_channels, 1.0);
if (rc < 0) {
PDEBUG(DRADIO, DEBUG_ERROR, "Failed to re-open wave file.\n");
return rc;
}
}
break;
#ifdef HAVE_ALSA
case AUDIO_MODE_AUDIODEV:
rc = sound_read(radio->tx_sound, audio_samples, radio->audio_buffer_size, radio->tx_audio_channels, NULL);
if (rc < 0) {
PDEBUG(DRADIO, DEBUG_ERROR, "Failed to read from sound device (rc = %d)!\n", audio_num);
if (rc == -EPIPE)
PDEBUG(DRADIO, DEBUG_ERROR, "Trying to recover.\n");
else
return 0;
}
jitter_save(&radio->tx_dejitter[0], audio_samples[0], rc);
jitter_load(&radio->tx_dejitter[0], audio_samples[0], audio_num);
if (radio->tx_audio_channels == 2) {
jitter_save(&radio->tx_dejitter[1], audio_samples[1], rc);
jitter_load(&radio->tx_dejitter[1], audio_samples[1], audio_num);
}
break;
#endif
case AUDIO_MODE_TESTTONE:
for (i = 0; i < audio_num; i++) {
audio_samples[0][i] = radio->testtone[0][radio->testtone_pos];
audio_samples[1][i] = radio->testtone[1][radio->testtone_pos];
radio->testtone_pos = (radio->testtone_pos + 1) % radio->testtone_length;
}
break;
default:
PDEBUG(DRADIO, DEBUG_ERROR, "Wrong audio mode, plese fix!\n");
return -EINVAL;
}
/* convert mono/stereo, generate differential signal */
if (radio->stereo && radio->tx_audio_channels == 1) {
/* mono to stereo: sum is 90%, differential signal is 0 */
for (i = 0; i < audio_num; i++) {
audio_samples[0][i] = 0.9;
audio_samples[1][i] = 0.0;
}
}
if (radio->stereo && radio->tx_audio_channels == 2) {
/* stereo: sum is 90%, diffential is 90% */
double left, right;
for (i = 0; i < audio_num; i++) {
left = audio_samples[0][i];
right = audio_samples[1][i];
audio_samples[0][i] = (left + right) * 0.45;
audio_samples[1][i] = (left - right) * 0.45;
}
}
if (!radio->stereo && radio->tx_audio_channels == 2) {
/* stereo to mono: sum both channel */
for (i = 0; i < audio_num; i++)
audio_samples[0][i] = (audio_samples[0][i] + audio_samples[1][i]) / 2.0;
}
/* remove DC */
iir_process(&radio->tx_dc_removal[0], audio_samples[0], audio_num);
if (radio->stereo)
iir_process(&radio->tx_dc_removal[1], audio_samples[1], audio_num);
/* gain volume */
if (radio->volume != 1.0) {
for (i = 0; i < audio_num; i++)
audio_samples[0][i] *= radio->volume;
if (radio->stereo) {
for (i = 0; i < audio_num; i++)
audio_samples[1][i] *= radio->volume;
}
}
/* upsample */
signal_num = samplerate_upsample(&radio->tx_resampler[0], audio_samples[0], audio_num, signal_samples[0]);
if (radio->stereo)
samplerate_upsample(&radio->tx_resampler[1], audio_samples[1], audio_num, signal_samples[1]);
/* prepare baseband */
memset(baseband, 0, sizeof(float) * 2 * signal_num);
/* filter audio (remove DC, remove high frequencies, pre-emphasis)
* and modulate */
switch (radio->modulation) {
case MODULATION_FM:
memset(signal_power, 1, signal_num);
pre_emphasis(&radio->fm_emphasis[0], signal_samples[0], signal_num);
clipper_process(signal_samples[0], signal_num);
if (radio->stereo) {
pre_emphasis(&radio->fm_emphasis[1], signal_samples[1], signal_num);
clipper_process(signal_samples[1], signal_num);
/* add pilot tone */
double phasestep = radio->pilot_phasestep;
double phase = radio->tx_pilot_phase;
for (i = 0; i < signal_num; i++) {
signal_samples[0][i] += sin(phase) * 0.1;
signal_samples[0][i] += signal_samples[1][i] * sin(phase * 2);
phase += phasestep;
if (phase >= 2.0 * M_PI)
phase -= 2.0 * M_PI;
}
radio->tx_pilot_phase = phase;
}
for (i = 0; i < signal_num; i++)
signal_samples[0][i] *= radio->fm_deviation;
fm_modulate_complex(&radio->fm_mod, signal_samples[0], signal_power, signal_num, baseband);
break;
case MODULATION_AM_DSB:
/* also clip to prevent overshooting after audio filtering */
clipper_process(signal_samples[0], signal_num);
iir_process(&radio->tx_am_bw_limit, signal_samples[0], signal_num);
am_modulate_complex(&radio->am_mod, signal_samples[0], signal_num, baseband);
break;
case MODULATION_AM_USB:
case MODULATION_AM_LSB:
/* also clip to prevent overshooting after audio filtering */
clipper_process(signal_samples[0], signal_num);
iir_process(&radio->tx_am_bw_limit, signal_samples[0], signal_num);
am_modulate_complex(&radio->am_mod, signal_samples[0], signal_num, baseband);
break;
default:
break;
}
return signal_num;
}
int radio_rx(radio_t *radio, float *baseband, int signal_num)
{
int i;
int audio_num;
sample_t *samples[3];
double p;
if (signal_num > radio->latspl) {
PDEBUG(DRADIO, DEBUG_ERROR, "signal_num > latspl, please fix!.\n");
abort();
}
if (signal_num > radio->signal_buffer_size) {
PDEBUG(DRADIO, DEBUG_ERROR, "signal_num > signal_buffer_size, please fix!.\n");
abort();
}
samples[0] = radio->signal_buffer;
samples[1] = radio->signal_buffer + radio->signal_buffer_size;
samples[2] = radio->signal_buffer + radio->signal_buffer_size * 2;
switch (radio->modulation) {
case MODULATION_FM:
fm_demodulate_complex(&radio->fm_demod, samples[0], signal_num, baseband, radio->I_buffer, radio->Q_buffer);
for (i = 0; i < signal_num; i++)
samples[0][i] /= radio->fm_deviation;
if (radio->stereo) {
/* filter pilot tone */
p = radio->rx_pilot_phase; /* don't increment in radio structure, will be done later */
for (i = 0; i < signal_num; i++) {
samples[1][i] = samples[0][i] * cos(p); /* I */
samples[2][i] = samples[0][i] * sin(p); /* Q */
p += radio->pilot_phasestep;
if (p >= 2.0 * M_PI)
p -= 2.0 * M_PI;
}
iir_process(&radio->rx_lp_pilot_I, samples[1], signal_num);
iir_process(&radio->rx_lp_pilot_Q, samples[2], signal_num);
/* mix pilot tone (double phase) with differential signal */
for (i = 0; i < signal_num; i++) {
p = atan2(samples[2][i], samples[1][i]);
/* subtract measured phase difference (use double amplitude, because we filter later) */
samples[1][i] = samples[0][i] * sin((radio->rx_pilot_phase - p) * 2.0) * 2.0;
radio->rx_pilot_phase += radio->pilot_phasestep;
if (radio->rx_pilot_phase >= 2.0 * M_PI)
radio->rx_pilot_phase -= 2.0 * M_PI;
}
/* filter to match bandwidth */
iir_process(&radio->rx_lp_sum, samples[0], signal_num);
iir_process(&radio->rx_lp_diff, samples[1], signal_num);
}
dc_filter(&radio->fm_emphasis[0], samples[0], signal_num);
de_emphasis(&radio->fm_emphasis[0], samples[0], signal_num);
if (radio->stereo) {
dc_filter(&radio->fm_emphasis[1], samples[1], signal_num);
de_emphasis(&radio->fm_emphasis[1], samples[1], signal_num);
}
break;
case MODULATION_AM_DSB:
am_demodulate_complex(&radio->am_demod, samples[0], signal_num, baseband, radio->I_buffer, radio->Q_buffer, radio->carrier_buffer);
break;
case MODULATION_AM_USB:
case MODULATION_AM_LSB:
am_demodulate_complex(&radio->am_demod, samples[0], signal_num, baseband, radio->I_buffer, radio->Q_buffer, radio->carrier_buffer);
break;
default:
break;
}
/* downsample */
audio_num = samplerate_downsample(&radio->rx_resampler[0], samples[0], signal_num);
if (radio->stereo)
samplerate_downsample(&radio->rx_resampler[1], samples[1], signal_num);
/* dampen volume */
if (radio->volume != 1.0) {
for (i = 0; i < audio_num; i++)
samples[0][i] /= radio->volume;
if (radio->stereo) {
for (i = 0; i < audio_num; i++)
samples[1][i] /= radio->volume;
}
}
/* convert mono/stereo, (from differential signal) */
if (radio->stereo && radio->rx_audio_channels == 1) {
/* stereo to mono */
for (i = 0; i < audio_num; i++) {
samples[0][i] = (samples[0][i] + samples[1][i]) / 2.0;
}
}
if (radio->stereo && radio->rx_audio_channels == 2) {
/* stereo from differential */
double sum, diff;
for (i = 0; i < audio_num; i++) {
sum = samples[0][i];
diff = samples[1][i];
samples[0][i] = sum + diff / 2.0;
samples[1][i] = sum - diff / 2.0;
}
}
if (!radio->stereo && radio->rx_audio_channels == 2) {
/* mono to stereo: clone channel */
for (i = 0; i < audio_num; i++)
samples[1][i] = samples[0][i];
}
/* store received audio */
switch (radio->rx_audio_mode) {
case AUDIO_MODE_WAVEFILE:
wave_write(&radio->wave_rx_rec, samples, audio_num);
break;
#ifdef HAVE_ALSA
case AUDIO_MODE_AUDIODEV:
jitter_save(&radio->rx_dejitter[0], samples[0], audio_num);
if (radio->rx_audio_channels == 2)
jitter_save(&radio->rx_dejitter[1], samples[1], audio_num);
audio_num = sound_get_tosend(radio->rx_sound, radio->signal_buffer_size);
jitter_load(&radio->rx_dejitter[0], samples[0], audio_num);
if (radio->rx_audio_channels == 2)
jitter_load(&radio->rx_dejitter[1], samples[1], audio_num);
audio_num = sound_write(radio->rx_sound, samples, NULL, audio_num, NULL, NULL, radio->rx_audio_channels);
if (audio_num < 0) {
PDEBUG(DRADIO, DEBUG_ERROR, "Failed to write to sound device (rc = %d)!\n", audio_num);
if (audio_num == -EPIPE)
PDEBUG(DRADIO, DEBUG_ERROR, "Trying to recover.\n");
else
return 0;
}
break;
#endif
default:
PDEBUG(DRADIO, DEBUG_ERROR, "Wrong audio mode, plese fix!\n");
return -EINVAL;
}
return signal_num;
}