Add 'osmoradio', an analog radio (FM/AM)

This radio can be a receiver or a transmitter or both simultaniously.
pull/1/head
Andreas Eversberg 5 years ago
parent c4d4e7feda
commit 49050eff90
  1. 1
      .gitignore
  2. 1
      configure.ac
  3. 1
      src/Makefile.am
  4. 1
      src/libdebug/debug.c
  5. 1
      src/libdebug/debug.h
  6. 44
      src/radio/Makefile.am
  7. 488
      src/radio/main.c
  8. 705
      src/radio/radio.c
  9. 85
      src/radio/radio.h

1
.gitignore vendored

@ -60,6 +60,7 @@ src/jtacs/jtacs
src/r2000/radiocom2000
src/jolly/jollycom
src/tv/osmotv
src/radio/osmoradio
sim/cnetz_sim
src/test/test_filter
src/test/test_sendevolumenregler

@ -85,6 +85,7 @@ AC_OUTPUT(
src/r2000/Makefile
src/jolly/Makefile
src/tv/Makefile
src/radio/Makefile
src/test/Makefile
src/Makefile
sim/Makefile

@ -47,5 +47,6 @@ SUBDIRS += \
r2000 \
jolly \
tv \
radio \
test

@ -62,6 +62,7 @@ struct debug_cat {
{ "uhd", "\033[1;35m" },
{ "soapy", "\033[1;35m" },
{ "wave", "\033[1;33m" },
{ "radio", "\033[1;34m" },
{ NULL, NULL }
};

@ -25,6 +25,7 @@
#define DUHD 18
#define DSOAPY 19
#define DWAVE 20
#define DRADIO 21
void get_win_size(int *w, int *h);

@ -0,0 +1,44 @@
AM_CPPFLAGS = -Wall -Wextra -g $(all_includes)
if HAVE_SDR
bin_PROGRAMS = \
osmoradio
osmoradio_SOURCES = \
radio.c \
main.c
osmoradio_LDADD = \
$(COMMON_LA) \
$(top_builddir)/src/libdebug/libdebug.a \
$(top_builddir)/src/libwave/libwave.a \
$(top_builddir)/src/libsample/libsample.a \
$(top_builddir)/src/libsdr/libsdr.a \
$(top_builddir)/src/libclipper/libclipper.a \
$(top_builddir)/src/libfm/libfm.a \
$(top_builddir)/src/libam/libam.a \
$(top_builddir)/src/libemphasis/libemphasis.a \
$(top_builddir)/src/libsamplerate/libsamplerate.a \
$(top_builddir)/src/libjitter/libjitter.a \
$(top_builddir)/src/libfilter/libfilter.a \
$(top_builddir)/src/libdisplay/libdisplay.a \
$(top_builddir)/src/libfft/libfft.a \
$(top_builddir)/src/libtimer/libtimer.a \
$(UHD_LIBS) \
$(SOAPY_LIBS) \
-lm
if HAVE_ALSA
osmoradio_LDADD += \
$(top_builddir)/src/libsound/libsound.a \
$(ALSA_LIBS)
endif
if HAVE_ALSA
AM_CPPFLAGS += -DHAVE_ALSA
endif
AM_CPPFLAGS += -DHAVE_SDR
endif

@ -0,0 +1,488 @@
/* 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/>.
*/
enum paging_signal;
#include <stdio.h>
#include <stdint.h>
#include <string.h>
#include <unistd.h>
#include <stdlib.h>
#include <getopt.h>
#include <signal.h>
#include <math.h>
#include <termios.h>
#include <unistd.h>
#include "../libsample/sample.h"
#include "../libdebug/debug.h"
#include "../libsdr/sdr_config.h"
#include "../libsdr/sdr.h"
#include "../libdisplay/display.h"
#include "radio.h"
#define DEFAULT_LO_OFFSET -1000000.0
void *sender_head = NULL;
int use_sdr = 0;
int num_kanal = 1; /* only one channel used for debugging */
void *get_sender_by_empfangsfrequenz() { return NULL; }
static double frequency = 0.0;
static int samplerate = 100000;
static int latency = 30;
static const char *tx_wave_file = NULL;
static const char *rx_wave_file = NULL;
static const char *tx_audiodev = NULL;
static const char *rx_audiodev = NULL;
static enum modulation modulation = MODULATION_NONE;
static int rx = 0, tx = 0;
static double bandwidth_am = 4500.0;
static double bandwidth_fm = 15000.0;
static double bandwidth = 0.0;
static double deviation = 75000.0;
static double modulation_index = 1.0;
static double time_constant_us = 50.0;
static int stereo = 0;
static int rds = 0;
static int rds2 = 0;
/* global variable to quit main loop */
int quit = 0;
void sighandler(int sigset)
{
if (sigset == SIGHUP)
return;
if (sigset == SIGPIPE)
return;
// clear_console_text();
printf("Signal received: %d\n", sigset);
quit = 1;
}
static int get_char()
{
struct timeval tv = {0, 0};
fd_set fds;
char c = 0;
int __attribute__((__unused__)) rc;
FD_ZERO(&fds);
FD_SET(0, &fds);
select(0+1, &fds, NULL, NULL, &tv);
if (FD_ISSET(0, &fds)) {
rc = read(0, &c, 1);
return c;
} else
return -1;
}
void print_help(const char *arg0)
{
printf("Usage: %s --sdr-soapy|--sdr-uhd <sdr options> -f <frequency> -M <modulation> -R|-T [options]\n", arg0);
/* - - */
printf("\noptions:\n");
printf(" -f --frequency <frequency>\n");
printf(" Give frequency in Hertz.\n");
printf(" -s --samplerate <sample rate>\n");
printf(" Give signal processing sample rate in Hz. (default = %d)\n", samplerate);
printf(" This sample rate must be high enough for the signal's spectrum to fit.\n");
printf(" I will inform you, if this bandwidth is too low.\n");
printf(" -r --tx-wave-file <filename>\n");
printf(" Input transmitted audio from wave file\n");
printf(" -w --rx-wave-file <filename>\n");
printf(" Output received audio to wave file\n");
printf(" -a --audio-device hw:<card>,<device>\n");
printf(" Input audio from sound card's device number\n");
printf(" -M --modulation fm | am | usb | lsb\n");
printf(" fm = Frequency modulation to be used for VHF.\n");
printf(" am = Amplitude modulation to be used for long/medium/short wave.\n");
printf(" usb = Amplitude modulation with upper side band only.\n");
printf(" lsb = Amplitude modulation with lower side band only.\n");
printf(" -R --rx\n");
printf(" Receive radio signal.\n");
printf(" -T --tx\n");
printf(" Transmit radio signal.\n");
printf(" -B --bandwidth\n");
printf(" Give bandwidth of audio frequency. (default AM=%.0f FM=%.0f)\n", bandwidth_am, bandwidth_fm);
printf(" -D --deviation\n");
printf(" Give deviation of frequency modulated signal. (default %.0f)\n", deviation);
printf(" -I --modulation-index 0..1\n");
printf(" Give modulation index of amplitude modulated signal. (default %.0f)\n", deviation);
printf(" -E --emphasis <uS> | 0\n");
printf(" Use given time constant of pre- and de-emphasis for frequency\n");
printf(" modulation. Give 0 to disbale. (default = %.0f uS)\n", time_constant_us);
printf(" VHF broadcast 50 uS in Europe and 75 uS in the United States.\n");
printf(" Other radio FM should use 530 uS, to cover complete speech spectrum.\n");
printf(" -S --stereo\n");
printf(" Enables stereo carrier for frequency modulated UHF broadcast.\n");
printf(" It uses the 'Pilot-tone' system.\n");
sdr_config_print_help();
}
static struct option long_options_common[] = {
{"help", 0, 0, 'h'},
{"frequency", 1, 0, 'f'},
{"samplerate", 1, 0, 's'},
{"tx-wave-file", 1, 0, 'r'},
{"rx-wave-file", 1, 0, 'w'},
{"audio-device", 1, 0, 'a'},
{"modulation", 1, 0, 'M'},
{"rx", 0, 0, 'R'},
{"tx", 0, 0, 'T'},
{"bandwidth", 1, 0, 'B'},
{"deviation", 1, 0, 'D'},
{"modulation-index", 1, 0, 'I'},
{"emphasis", 1, 0, 'E'},
{"stereo", 0, 0, 'S'},
{0, 0, 0, 0}
};
static const char *optstring_common = "hf:s:r:w:a:M:RTB:D:I:E:S";
struct option *long_options;
char *optstring;
static void check_duplicate_option(int num, struct option *option)
{
int i;
for (i = 0; i < num; i++) {
if (long_options[i].val == option->val) {
fprintf(stderr, "Duplicate option %d. Please fix!\n", option->val);
abort();
}
}
}
void set_options_common(void)
{
int i = 0, j;
long_options = calloc(sizeof(*long_options), 256);
for (j = 0; long_options_common[i].name; i++, j++) {
check_duplicate_option(i, &long_options_common[j]);
memcpy(&long_options[i], &long_options_common[j], sizeof(*long_options));
}
for (j = 0; sdr_config_long_options[j].name; i++, j++) {
check_duplicate_option(i, &sdr_config_long_options[j]);
memcpy(&long_options[i], &sdr_config_long_options[j], sizeof(*long_options));
}
optstring = calloc(256, 2);
strcpy(optstring, optstring_common);
strcat(optstring, sdr_config_optstring);
}
static int handle_options(int argc, char **argv)
{
int skip_args = 0;
int rc;
set_options_common();
while (1) {
int option_index = 0, c;
c = getopt_long(argc, argv, optstring, long_options, &option_index);
if (c == -1)
break;
switch (c) {
case 'h':
print_help(argv[0]);
exit(0);
case 'f':
frequency = atof(optarg);
skip_args += 2;
break;
case 's':
samplerate = atof(optarg);
skip_args += 2;
break;
case 'r':
tx_wave_file = strdup(optarg);
skip_args += 2;
break;
case 'w':
rx_wave_file = strdup(optarg);
skip_args += 2;
break;
case 'a':
tx_audiodev = strdup(optarg);
rx_audiodev = strdup(optarg);
skip_args += 2;
break;
case 'M':
if (!strcasecmp(optarg, "fm"))
modulation = MODULATION_FM;
else
if (!strcasecmp(optarg, "am"))
modulation = MODULATION_AM_DSB;
else
if (!strcasecmp(optarg, "usb"))
modulation = MODULATION_AM_USB;
else
if (!strcasecmp(optarg, "lsb"))
modulation = MODULATION_AM_LSB;
else
{
fprintf(stderr, "Invalid modulation option, see help!\n");
exit(0);
}
skip_args += 2;
break;
case 'R':
rx = 1;
skip_args += 1;
break;
case 'T':
tx = 1;
skip_args += 1;
break;
case 'B':
bandwidth = atof(optarg);
skip_args += 2;
break;
case 'D':
deviation = atof(optarg);
skip_args += 2;
break;
case 'I':
modulation_index = atof(optarg);
if (modulation_index < 0.0 || modulation_index > 1.0) {
fprintf(stderr, "Invalid modulation index, see help!\n");
exit(0);
}
skip_args += 2;
break;
case 'E':
time_constant_us = atof(optarg);
skip_args += 2;
break;
case 'S':
stereo = 1;
skip_args += 1;
break;
default:
rc = sdr_config_opt_switch(c, &skip_args);
if (rc < 0)
exit(0);
break;
}
}
return skip_args;
}
int main(int argc, char *argv[])
{
int skip_args;
int rc;
const char *arg0 = argv[0];
radio_t radio;
struct termios term, term_orig;
int c;
int latspl;
debuglevel = 0;
sdr_config_init(DEFAULT_LO_OFFSET);
skip_args = handle_options(argc, argv);
argc -= skip_args + 1;
argv += skip_args + 1;
if (frequency == 0.0) {
printf("No frequency given, I suggest to use 100000000 (100 MHz) and FM\n\n");
print_help(arg0);
exit(0);
}
rc = sdr_configure(samplerate);
if (rc < 0)
return rc;
if (rc == 0) {
fprintf(stderr, "Please select SDR, see help!\n");
exit(0);
}
if (modulation == MODULATION_NONE) {
fprintf(stderr, "Please select modulation, see help!\n");
exit(0);
}
if (bandwidth == 0) {
if (modulation == MODULATION_FM)
bandwidth = bandwidth_fm;
else
bandwidth = bandwidth_am;
}
if (stereo && modulation != MODULATION_FM) {
fprintf(stderr, "Stereo works with FM only, see help!\n");
exit(0);
}
if (!rx && !tx) {
fprintf(stderr, "You need to specify --rx (receiver) and/or --tx (transmitter), see help!\n");
exit(0);
}
if (stereo && bandwidth != 15000.0) {
fprintf(stderr, "Warning: Stereo works with bandwidth of 15 KHz only, using this bandwidth!\n");
}
if (stereo && time_constant_us != 75.0 && time_constant_us != 50.0) {
fprintf(stderr, "Stereo works with time constant of 50 uS or 75 uS only, see help!\n");
exit(0);
}
/* now we have latency and sample rate */
latspl = samplerate * latency / 1000;
rc = radio_init(&radio, latspl, samplerate, tx_wave_file, rx_wave_file, (tx) ? tx_audiodev : NULL, (rx) ? rx_audiodev : NULL, modulation, bandwidth, deviation, modulation_index, time_constant_us, stereo, rds, rds2);
if (rc < 0) {
fprintf(stderr, "Failed to initialize radio with given options, exitting!\n");
exit(0);
}
void *sdr = NULL;
float *sendbuff = NULL;
sendbuff = calloc(latspl * 2, sizeof(*sendbuff));
if (!sendbuff) {
fprintf(stderr, "No mem!\n");
goto error;
}
double tx_frequencies[1], rx_frequencies[1];
tx_frequencies[0] = frequency;
rx_frequencies[0] = frequency;
sdr = sdr_open(NULL, tx_frequencies, rx_frequencies, 1, 0.0, samplerate, latspl, 0.0, 0.0);
if (!sdr)
goto error;
sdr_start(sdr);
/* prepare terminal */
tcgetattr(0, &term_orig);
term = term_orig;
term.c_lflag &= ~(ISIG|ICANON|ECHO);
term.c_cc[VMIN]=1;
term.c_cc[VTIME]=2;
tcsetattr(0, TCSANOW, &term);
/* catch signals */
signal(SIGINT, sighandler);
signal(SIGHUP, sighandler);
signal(SIGTERM, sighandler);
signal(SIGPIPE, sighandler);
printf("Starting radio...\n");
rc = radio_start(&radio);
if (rc < 0) {
fprintf(stderr, "Failed to start radio's streaming, exitting!\n");
goto error;
}
int tosend, got;
while (!quit) {
usleep(1000);
got = sdr_read(sdr, (void *)sendbuff, latspl, 0, NULL);
if (rx) {
got = radio_rx(&radio, sendbuff, got);
if (got < 0)
break;
}
if (tx) {
tosend = sdr_get_tosend(sdr, latspl);
if (tosend > latspl / 10)
tosend = latspl / 10;
if (tosend == 0) {
continue;
}
/* perform radio modulation */
tosend = radio_tx(&radio, sendbuff, tosend);
if (tosend < 0)
break;
/* write to SDR */
sdr_write(sdr, (void *)sendbuff, NULL, tosend, NULL, NULL, 0);
}
/* process keyboard input */
next_char:
c = get_char();
switch (c) {
case 3:
/* quit */
// if (clear_console_text)
// clear_console_text();
printf("CTRL+c received, quitting!\n");
quit = 1;
goto next_char;
#if 0
- carrier frequency
- deviation
- modulation index
- stereo pilot
case 'm':
/* toggle measurements display */
display_iq_on(0);
display_spectrum_on(0);
display_measurements_on(-1);
goto next_char;
#endif
case 'q':
/* toggle IQ display */
display_measurements_on(0);
display_spectrum_on(0);
display_iq_on(-1);
goto next_char;
case 's':
/* toggle spectrum display */
display_measurements_on(0);
display_iq_on(0);
display_spectrum_on(-1);
goto next_char;
case 'B':
calibrate_bias();
goto next_char;
}
}
/* reset signals */
signal(SIGINT, SIG_DFL);
signal(SIGHUP, SIG_DFL);
signal(SIGTERM, SIG_DFL);
signal(SIGPIPE, SIG_DFL);
/* reset terminal */
tcsetattr(0, TCSANOW, &term_orig);
error:
free(sendbuff);
if (sdr)
sdr_close(sdr);
radio_exit(&radio);
return 0;
}

@ -0,0 +1,705 @@
/* 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, int stereo, int rds, int rds2)
{
int rc = -EINVAL;
clipper_init(CLIP_POINT);
memset(radio, 0, sizeof(*radio));
radio->latspl = latspl;
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 = 4800;
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
}
/* 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);
/* 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]);
/* substract 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);
/* 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;
}

@ -0,0 +1,85 @@
#include "../libwave/wave.h"
#include "../libsamplerate/samplerate.h"
#include "../libemphasis/emphasis.h"
#include "../libjitter/jitter.h"
#include "../libfm/fm.h"
#include "../libam/am.h"
enum modulation {
MODULATION_NONE = 0,
MODULATION_FM,
MODULATION_AM_DSB,
MODULATION_AM_USB,
MODULATION_AM_LSB,
};
enum audio_mode {
AUDIO_MODE_NONE = 0,
AUDIO_MODE_WAVEFILE,
AUDIO_MODE_AUDIODEV,
AUDIO_MODE_TESTTONE,
};
typedef struct radio {
/* modes */
int latspl; /* maximum number of samples */
enum modulation modulation; /* modulation type */
enum audio_mode tx_audio_mode; /* mode for audio source */
enum audio_mode rx_audio_mode; /* mode for audio sink */
int stereo; /* use stere FM */
int rds, rds2; /* use RDS */
/* audio stage */
double tx_audio_samplerate; /* sample rate of audio source */
double rx_audio_samplerate; /* sample rate of audio sink */
int tx_audio_channels; /* number of channels of audio source */
int rx_audio_channels; /* number of channels of audio sink */
double audio_bandwidth; /* audio bandwidth */
const char *tx_wave_file; /* wave file name of source */
const char *rx_wave_file; /* wave file name of sink */
wave_play_t wave_tx_play; /* wave playback process */
wave_rec_t wave_rx_rec; /* wave record process */