Add support for AM to libmobile and libsdr

pull/1/head
Andreas Eversberg 3 years ago
parent c2f14834e5
commit 150a77b69d
  1. 1
      src/amps/Makefile.am
  2. 1
      src/anetz/Makefile.am
  3. 1
      src/bnetz/Makefile.am
  4. 2
      src/bnetz/dialer.c
  5. 1
      src/cnetz/Makefile.am
  6. 2
      src/datenklo/datenklo.c
  7. 1
      src/imts/Makefile.am
  8. 2
      src/imts/dialer.c
  9. 1
      src/jolly/Makefile.am
  10. 1
      src/jtacs/Makefile.am
  11. 4
      src/libam/am.c
  12. 2
      src/libmncc/mncc_console.c
  13. 25
      src/libmobile/sender.c
  14. 11
      src/libmobile/sender.h
  15. 92
      src/libsdr/sdr.c
  16. 2
      src/libsdr/sdr.h
  17. 2
      src/libsound/sound.h
  18. 2
      src/libsound/sound_alsa.c
  19. 1
      src/nmt/Makefile.am
  20. 1
      src/r2000/Makefile.am
  21. 4
      src/radio/main.c
  22. 4
      src/radio/radio.c
  23. 1
      src/tacs/Makefile.am
  24. 2
      src/test/Makefile.am
  25. 3
      src/tv/Makefile.am
  26. 4
      src/tv/main.c

@ -55,6 +55,7 @@ endif
if HAVE_SDR
amps_LDADD += \
$(top_builddir)/src/libsdr/libsdr.a \
$(top_builddir)/src/libam/libam.a \
$(top_builddir)/src/libfft/libfft.a \
$(UHD_LIBS) \
$(SOAPY_LIBS)

@ -44,6 +44,7 @@ endif
if HAVE_SDR
anetz_LDADD += \
$(top_builddir)/src/libsdr/libsdr.a \
$(top_builddir)/src/libam/libam.a \
$(top_builddir)/src/libfft/libfft.a \
$(UHD_LIBS) \
$(SOAPY_LIBS)

@ -61,6 +61,7 @@ endif
if HAVE_SDR
bnetz_LDADD += \
$(top_builddir)/src/libsdr/libsdr.a \
$(top_builddir)/src/libam/libam.a \
$(top_builddir)/src/libfft/libfft.a \
$(UHD_LIBS) \
$(SOAPY_LIBS)

@ -346,7 +346,7 @@ int main(int argc, char *argv[])
#ifdef HAVE_ALSA
/* init sound */
audio = sound_open(audiodev, NULL, NULL, 1, 0.0, samplerate, latspl, 1.0, 4000.0);
audio = sound_open(audiodev, NULL, NULL, NULL, 1, 0.0, samplerate, latspl, 1.0, 4000.0, 2.0);
if (!audio) {
PDEBUG(DBNETZ, DEBUG_ERROR, "No sound device!\n");
goto exit;

@ -44,6 +44,7 @@ endif
if HAVE_SDR
cnetz_LDADD += \
$(top_builddir)/src/libsdr/libsdr.a \
$(top_builddir)/src/libam/libam.a \
$(top_builddir)/src/libfft/libfft.a \
$(UHD_LIBS) \
$(SOAPY_LIBS)

@ -1395,7 +1395,7 @@ int datenklo_open_audio(datenklo_t *datenklo, const char *audiodev, int latency,
#ifdef HAVE_ALSA
/* init sound */
datenklo->audio = sound_open(audiodev, NULL, NULL, channels, 0.0, datenklo->samplerate, datenklo->latspl, 1.0, 4000.0);
datenklo->audio = sound_open(audiodev, NULL, NULL, NULL, channels, 0.0, datenklo->samplerate, datenklo->latspl, 1.0, 4000.0, 2.0);
if (!datenklo->audio) {
PDEBUG(DDATENKLO, DEBUG_ERROR, "No sound device!\n");
return -EIO;

@ -52,6 +52,7 @@ endif
if HAVE_SDR
imts_LDADD += \
$(top_builddir)/src/libsdr/libsdr.a \
$(top_builddir)/src/libam/libam.a \
$(top_builddir)/src/libfft/libfft.a \
$(UHD_LIBS) \
$(SOAPY_LIBS)

@ -307,7 +307,7 @@ int main(int argc, char *argv[])
#ifdef HAVE_ALSA
/* init sound */
audio = sound_open(audiodev, NULL, NULL, 1, 0.0, samplerate, latspl, 1.0, 4000.0);
audio = sound_open(audiodev, NULL, NULL, NULL, 1, 0.0, samplerate, latspl, 1.0, 4000.0, 2.0);
if (!audio) {
PDEBUG(DBNETZ, DEBUG_ERROR, "No sound device!\n");
goto exit;

@ -38,6 +38,7 @@ endif
if HAVE_SDR
jollycom_LDADD += \
$(top_builddir)/src/libsdr/libsdr.a \
$(top_builddir)/src/libam/libam.a \
$(top_builddir)/src/libfft/libfft.a \
$(UHD_LIBS) \
$(SOAPY_LIBS)

@ -38,6 +38,7 @@ endif
if HAVE_SDR
jtacs_LDADD += \
$(top_builddir)/src/libsdr/libsdr.a \
$(top_builddir)/src/libam/libam.a \
$(top_builddir)/src/libfft/libfft.a \
$(UHD_LIBS) \
$(SOAPY_LIBS)

@ -111,8 +111,8 @@ void am_modulate_complex(am_mod_t *mod, sample_t *amplitude, int num, float *bas
else if (phase >= 65536.0)
phase -= 65536.0;
} else {
*baseband++ = cos(phase) * vector;
*baseband++ = sin(phase) * vector;
*baseband++ += cos(phase) * vector;
*baseband++ += sin(phase) * vector;
phase += rot;
if (phase < 0.0)
phase += 2.0 * M_PI;

@ -299,7 +299,7 @@ int console_open_audio(int __attribute__((unused)) latspl)
#ifdef HAVE_ALSA
/* open sound device for call control */
/* use factor 1.4 of speech level for complete range of sound card */
console.sound = sound_open(console.audiodev, NULL, NULL, 1, 0.0, console.samplerate, latspl, 1.4, 4000.0);
console.sound = sound_open(console.audiodev, NULL, NULL, NULL, 1, 0.0, console.samplerate, latspl, 1.4, 4000.0, 2.0);
if (!console.sound) {
PDEBUG(DSENDER, DEBUG_ERROR, "No sound device!\n");
return -EIO;

@ -183,36 +183,38 @@ int sender_open_audio(int latspl)
channels++;
}
double tx_f[channels], rx_f[channels], paging_frequency = 0.0;
int am[channels];
for (i = 0, inst = master; inst; i++, inst = inst->slave) {
tx_f[i] = inst->sendefrequenz;
rx_f[i] = inst->empfangsfrequenz;
am[i] = inst->am;
if (inst->ruffrequenz)
paging_frequency = inst->ruffrequenz;
}
if (master->write_rx_wave) {
rc = wave_create_record(&master->wave_rx_rec, master->write_rx_wave, master->samplerate, channels, master->max_deviation);
rc = wave_create_record(&master->wave_rx_rec, master->write_rx_wave, master->samplerate, channels, (master->max_deviation) ?: 1.0);
if (rc < 0) {
PDEBUG(DSENDER, DEBUG_ERROR, "Failed to create WAVE recoding instance!\n");
return rc;
}
}
if (master->write_tx_wave) {
rc = wave_create_record(&master->wave_tx_rec, master->write_tx_wave, master->samplerate, channels, master->max_deviation);
rc = wave_create_record(&master->wave_tx_rec, master->write_tx_wave, master->samplerate, channels, (master->max_deviation) ?: 1.0);
if (rc < 0) {
PDEBUG(DSENDER, DEBUG_ERROR, "Failed to create WAVE recoding instance!\n");
return rc;
}
}
if (master->read_rx_wave) {
rc = wave_create_playback(&master->wave_rx_play, master->read_rx_wave, &master->samplerate, &channels, master->max_deviation);
rc = wave_create_playback(&master->wave_rx_play, master->read_rx_wave, &master->samplerate, &channels, (master->max_deviation) ?: 1.0);
if (rc < 0) {
PDEBUG(DSENDER, DEBUG_ERROR, "Failed to create WAVE playback instance!\n");
return rc;
}
}
if (master->read_tx_wave) {
rc = wave_create_playback(&master->wave_tx_play, master->read_tx_wave, &master->samplerate, &channels, master->max_deviation);
rc = wave_create_playback(&master->wave_tx_play, master->read_tx_wave, &master->samplerate, &channels, (master->max_deviation) ?: 1.0);
if (rc < 0) {
PDEBUG(DSENDER, DEBUG_ERROR, "Failed to create WAVE playback instance!\n");
return rc;
@ -220,7 +222,7 @@ int sender_open_audio(int latspl)
}
/* open device */
master->audio = master->audio_open(master->audiodev, tx_f, rx_f, channels, paging_frequency, master->samplerate, latspl, master->max_deviation, master->max_modulation);
master->audio = master->audio_open(master->audiodev, tx_f, rx_f, am, channels, paging_frequency, master->samplerate, latspl, (master->max_deviation) ?: 1.0, master->max_modulation, master->modulation_index);
if (!master->audio) {
PDEBUG(DSENDER, DEBUG_ERROR, "No device for transceiver!\n");
return -EIO;
@ -293,6 +295,19 @@ void sender_set_fm(sender_t *sender, double max_deviation, double max_modulation
PDEBUG_CHAN(DSENDER, DEBUG_DEBUG, "Deviation at speech level: %.1f kHz\n", speech_deviation / 1000.0);
}
/* set amplitude modulation and parameters */
void sender_set_am(sender_t *sender, double max_modulation, double speech_level, double max_display, double modulation_index)
{
sender->am = 1;
sender->max_deviation = 0;
sender->max_modulation = max_modulation;
sender->speech_deviation = speech_level;
sender->max_display = max_display;
sender->modulation_index = modulation_index;
PDEBUG_CHAN(DSENDER, DEBUG_DEBUG, "Modulation degree: %.0f %%, Maximum modulation: %.1f kHz\n", modulation_index / 100.0, max_modulation / 1000.0);
}
static void gain_samples(sample_t *samples, int length, double gain)
{
int i;

@ -33,16 +33,18 @@ typedef struct sender {
double empfangsfrequenz; /* receiver frequency */
double ruffrequenz; /* special paging frequency used for B-Netz */
/* fm levels */
double max_deviation; /* max frequency deviation */
/* FM/AM levels */
int am; /* use AM instead of FM */
double max_deviation; /* max frequency deviation / level */
double max_modulation; /* max frequency modulated */
double speech_deviation; /* deviation of 1000 Hz reference tone at speech level */
double speech_deviation; /* deviation / level of 1000 Hz reference tone at speech level */
double modulation_index; /* AM modulation index */
double max_display; /* level of displaying wave form */
/* audio */
void *audio;
char audiodev[64]; /* audio device name (alsa or sdr) */
void *(*audio_open)(const char *, double *, double *, int, double, int, int, double, double);
void *(*audio_open)(const char *, double *, double *, int *, int, double, int, int, double, double, double);
int (*audio_start)(void *);
void (*audio_close)(void *);
int (*audio_write)(void *, sample_t **, uint8_t **, int, enum paging_signal *, int *, int);
@ -93,6 +95,7 @@ extern int cant_recover;
int sender_create(sender_t *sender, const char *kanal, double sendefrequenz, double empfangsfrequenz, const char *audiodev, int use_sdr, int samplerate, double rx_gain, int pre_emphasis, int de_emphasis, const char *write_rx_wave, const char *write_tx_wave, const char *read_rx_wave, const char *read_tx_wave, int loopback, enum paging_signal paging_signal);
void sender_destroy(sender_t *sender);
void sender_set_fm(sender_t *sender, double max_deviation, double max_modulation, double speech_deviation, double max_display);
void sender_set_am(sender_t *sender, double max_modulation, double speech_deviation, double max_display, double modulation_index);
int sender_open_audio(int latspl);
int sender_start_audio(void);
void process_sender_audio(sender_t *sender, int *quit, int latspl);

@ -30,6 +30,7 @@ enum paging_signal;
#include <unistd.h>
#include "../libsample/sample.h"
#include "../libfm/fm.h"
#include "../libam/am.h"
#include "../libtimer/timer.h"
#include "../libmobile/sender.h"
#include "sdr_config.h"
@ -71,8 +72,11 @@ typedef struct sdr_thread {
typedef struct sdr_chan {
double tx_frequency; /* frequency used */
double rx_frequency; /* frequency used */
fm_mod_t mod; /* modulator instance */
fm_demod_t demod; /* demodulator instance */
int am; /* use AM instead of FM */
fm_mod_t fm_mod; /* modulator instance */
fm_demod_t fm_demod; /* demodulator instance */
am_mod_t am_mod; /* modulator instance */
am_demod_t am_demod; /* demodulator instance */
dispmeasparam_t *dmp_rf_level;
dispmeasparam_t *dmp_freq_offset;
dispmeasparam_t *dmp_deviation;
@ -94,9 +98,10 @@ typedef struct sdr {
wave_rec_t wave_tx_rec;
wave_play_t wave_rx_play;
wave_play_t wave_tx_play;
float *modbuff; /* buffer for FM transmodulation */
float *modbuff; /* buffer for transmodulation */
sample_t *modbuff_I;
sample_t *modbuff_Q;
sample_t *modbuff_carrier;
sample_t *wavespl0; /* sample buffer for wave generation */
sample_t *wavespl1;
} sdr_t;
@ -131,7 +136,7 @@ static void show_spectrum(const char *direction, double halfbandwidth, double ce
PDEBUG(DSDR, DEBUG_INFO, "Frequency P = %.4f MHz (Paging Frequency)\n", paging_frequency / 1e6);
}
void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_frequency, double *rx_frequency, int channels, double paging_frequency, int samplerate, int latspl, double max_deviation, double max_modulation)
void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_frequency, double *rx_frequency, int *am, int channels, double paging_frequency, int samplerate, int latspl, double max_deviation, double max_modulation, double modulation_index)
{
sdr_t *sdr;
int threads = 1, oversample = 1; /* always use threads */
@ -231,6 +236,11 @@ void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_freq
PDEBUG(DSDR, DEBUG_ERROR, "NO MEM!\n");
goto error;
}
sdr->modbuff_carrier = calloc(sdr->latspl, sizeof(*sdr->modbuff_carrier));
if (!sdr->modbuff_carrier) {
PDEBUG(DSDR, DEBUG_ERROR, "NO MEM!\n");
goto error;
}
sdr->wavespl0 = calloc(sdr->latspl, sizeof(*sdr->wavespl0));
if (!sdr->wavespl0) {
PDEBUG(DSDR, DEBUG_ERROR, "NO MEM!\n");
@ -297,7 +307,14 @@ void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_freq
double tx_offset;
tx_offset = sdr->chan[c].tx_frequency - tx_center_frequency;
PDEBUG(DSDR, DEBUG_DEBUG, "Frequency #%d: TX offset: %.6f MHz\n", c, tx_offset / 1e6);
rc = fm_mod_init(&sdr->chan[c].mod, samplerate, tx_offset, sdr->amplitude);
sdr->chan[c].am = am[c];
if (am[c]) {
double gain, bias;
gain = modulation_index / 2.0;
bias = 1.0 - gain;
rc = am_mod_init(&sdr->chan[c].am_mod, samplerate, tx_offset, sdr->amplitude * gain, sdr->amplitude * bias);
} else
rc = fm_mod_init(&sdr->chan[c].fm_mod, samplerate, tx_offset, sdr->amplitude);
if (rc < 0)
goto error;
}
@ -305,7 +322,7 @@ void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_freq
double tx_offset;
tx_offset = sdr->chan[sdr->paging_channel].tx_frequency - tx_center_frequency;
PDEBUG(DSDR, DEBUG_DEBUG, "Paging Frequency: TX offset: %.6f MHz\n", tx_offset / 1e6);
rc = fm_mod_init(&sdr->chan[sdr->paging_channel].mod, samplerate, tx_offset, sdr->amplitude);
rc = fm_mod_init(&sdr->chan[sdr->paging_channel].fm_mod, samplerate, tx_offset, sdr->amplitude);
if (rc < 0)
goto error;
}
@ -400,7 +417,11 @@ void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_freq
double rx_offset;
rx_offset = sdr->chan[c].rx_frequency - rx_center_frequency;
PDEBUG(DSDR, DEBUG_DEBUG, "Frequency #%d: RX offset: %.6f MHz\n", c, rx_offset / 1e6);
rc = fm_demod_init(&sdr->chan[c].demod, samplerate, rx_offset, bandwidth);
sdr->chan[c].am = am[c];
if (am[c])
rc = am_demod_init(&sdr->chan[c].am_demod, samplerate, rx_offset, bandwidth, 1.0 / modulation_index);
else
rc = fm_demod_init(&sdr->chan[c].fm_demod, samplerate, rx_offset, bandwidth / 2.0);
if (rc < 0)
goto error;
}
@ -428,8 +449,10 @@ void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_freq
if (!sender)
continue;
sdr->chan[c].dmp_rf_level = display_measurements_add(&sender->dispmeas, "RF Level", "%.1f dB", DISPLAY_MEAS_AVG, DISPLAY_MEAS_LEFT, -96.0, 0.0, -INFINITY);
sdr->chan[c].dmp_freq_offset = display_measurements_add(&sender->dispmeas, "Freq. Offset", "%+.2f KHz", DISPLAY_MEAS_AVG, DISPLAY_MEAS_CENTER, -max_deviation / 1000.0 * 2.0, max_deviation / 1000.0 * 2.0, 0.0);
sdr->chan[c].dmp_deviation = display_measurements_add(&sender->dispmeas, "Deviation", "%.2f KHz", DISPLAY_MEAS_PEAK2PEAK, DISPLAY_MEAS_LEFT, 0.0, max_deviation / 1000.0 * 1.5, max_deviation / 1000.0);
if (!am[c]) {
sdr->chan[c].dmp_freq_offset = display_measurements_add(&sender->dispmeas, "Freq. Offset", "%+.2f KHz", DISPLAY_MEAS_AVG, DISPLAY_MEAS_CENTER, -max_modulation / 1000.0 * 2.0, max_modulation / 1000.0 * 2.0, 0.0);
sdr->chan[c].dmp_deviation = display_measurements_add(&sender->dispmeas, "Deviation", "%.2f KHz", DISPLAY_MEAS_PEAK2PEAK, DISPLAY_MEAS_LEFT, 0.0, max_deviation / 1000.0 * 1.5, max_deviation / 1000.0);
}
}
}
@ -700,6 +723,7 @@ void sdr_close(void *inst)
free(sdr->modbuff);
free(sdr->modbuff_I);
free(sdr->modbuff_Q);
free(sdr->modbuff_carrier);
free(sdr->wavespl0);
free(sdr->wavespl1);
wave_destroy_record(&sdr->wave_rx_rec);
@ -710,11 +734,13 @@ void sdr_close(void *inst)
int c;
for (c = 0; c < sdr->channels; c++) {
fm_mod_exit(&sdr->chan[c].mod);
fm_demod_exit(&sdr->chan[c].demod);
fm_mod_exit(&sdr->chan[c].fm_mod);
fm_demod_exit(&sdr->chan[c].fm_demod);
am_mod_exit(&sdr->chan[c].am_mod);
am_demod_exit(&sdr->chan[c].am_demod);
}
if (sdr->paging_channel)
fm_mod_exit(&sdr->chan[sdr->paging_channel].mod);
fm_mod_exit(&sdr->chan[sdr->paging_channel].fm_mod);
free(sdr->chan);
}
free(sdr);
@ -747,9 +773,12 @@ int sdr_write(void *inst, sample_t **samples, uint8_t **power, int num, enum pag
for (c = 0; c < channels; c++) {
/* switch to paging channel, if requested */
if (on[c] && sdr->paging_channel)
fm_modulate_complex(&sdr->chan[sdr->paging_channel].mod, samples[c], power[c], num, buff);
else
fm_modulate_complex(&sdr->chan[c].mod, samples[c], power[c], num, buff);
fm_modulate_complex(&sdr->chan[sdr->paging_channel].fm_mod, samples[c], power[c], num, buff);
else if (sdr->chan[c].am) {
if (power[c][0])
am_modulate_complex(&sdr->chan[c].am_mod, samples[c], num, buff);
} else
fm_modulate_complex(&sdr->chan[c].fm_mod, samples[c], power[c], num, buff);
}
} else {
buff = (float *)samples;
@ -914,7 +943,10 @@ int sdr_read(void *inst, sample_t **samples, int num, int channels, double *rf_l
if (channels) {
for (c = 0; c < channels; c++) {
fm_demodulate_complex(&sdr->chan[c].demod, samples[c], count, buff, sdr->modbuff_I, sdr->modbuff_Q);
if (sdr->chan[c].am)
am_demodulate_complex(&sdr->chan[c].am_demod, samples[c], count, buff, sdr->modbuff_I, sdr->modbuff_Q, sdr->modbuff_carrier);
else
fm_demodulate_complex(&sdr->chan[c].fm_demod, samples[c], count, buff, sdr->modbuff_I, sdr->modbuff_Q);
sender_t *sender = get_sender_by_empfangsfrequenz(sdr->chan[c].rx_frequency);
if (!sender || !count)
continue;
@ -928,20 +960,22 @@ int sdr_read(void *inst, sample_t **samples, int num, int channels, double *rf_l
avg = log10(avg) * 20;
display_measurements_update(sdr->chan[c].dmp_rf_level, avg, 0.0);
rf_level_db[c] = avg;
min = 0.0;
max = 0.0;
avg = 0.0;
for (s = 0; s < count; s++) {
avg += samples[c][s];
if (s == 0 || samples[c][s] > max)
max = samples[c][s];
if (s == 0 || samples[c][s] < min)
min = samples[c][s];
if (!sdr->chan[c].am) {
min = 0.0;
max = 0.0;
avg = 0.0;
for (s = 0; s < count; s++) {
avg += samples[c][s];
if (s == 0 || samples[c][s] > max)
max = samples[c][s];
if (s == 0 || samples[c][s] < min)
min = samples[c][s];
}
avg /= (double)count;
display_measurements_update(sdr->chan[c].dmp_freq_offset, avg / 1000.0, 0.0);
/* use half min and max, because we want the deviation above/below (+-) center frequency. */
display_measurements_update(sdr->chan[c].dmp_deviation, min / 2.0 / 1000.0, max / 2.0 / 1000.0);
}
avg /= (double)count;
display_measurements_update(sdr->chan[c].dmp_freq_offset, avg / 1000.0, 0.0);
/* use half min and max, because we want the deviation above/below (+-) center frequency. */
display_measurements_update(sdr->chan[c].dmp_deviation, min / 2.0 / 1000.0, max / 2.0 / 1000.0);
}
}

@ -2,7 +2,7 @@
enum paging_signal;
int sdr_start(void *inst);
void *sdr_open(const char *audiodev, double *tx_frequency, double *rx_frequency, int channels, double paging_frequency, int samplerate, int latspl, double max_deviation, double max_modulation);
void *sdr_open(const char *audiodev, double *tx_frequency, double *rx_frequency, int *am, int channels, double paging_frequency, int samplerate, int latspl, double max_deviation, double max_modulation, double modulation_index);
void sdr_close(void *inst);
int sdr_write(void *inst, sample_t **samples, uint8_t **power, int num, enum paging_signal *paging_signal, int *on, int channels);
int sdr_read(void *inst, sample_t **samples, int num, int channels, double *rf_level_db);

@ -1,7 +1,7 @@
enum paging_signal;
void *sound_open(const char *audiodev, double *tx_frequency, double *rx_frequency, int channels, double paging_frequency, int samplerate, int latspl, double max_deviation, double max_modulation);
void *sound_open(const char *audiodev, double *tx_frequency, double *rx_frequency, int *am, int channels, double paging_frequency, int samplerate, int latspl, double max_deviation, double max_modulation, double modulation_index);
int sound_start(void *inst);
void sound_close(void *inst);
int sound_write(void *inst, sample_t **samples, uint8_t **power, int num, enum paging_signal *paging_signal, int *on, int channels);

@ -131,7 +131,7 @@ static int sound_prepare(sound_t *sound)
return 0;
}
void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_frequency, double __attribute__((unused)) *rx_frequency, int channels, double __attribute__((unused)) paging_frequency, int samplerate, int __attribute((unused)) latspl, double max_deviation, double __attribute__((unused)) max_modulation)
void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_frequency, double __attribute__((unused)) *rx_frequency, int __attribute__((unused)) *am, int channels, double __attribute__((unused)) paging_frequency, int samplerate, int __attribute((unused)) latspl, double max_deviation, double __attribute__((unused)) max_modulation, double __attribute__((unused)) modulation_index)
{
sound_t *sound;
int rc, rc_rec, rc_play;

@ -51,6 +51,7 @@ endif
if HAVE_SDR
nmt_LDADD += \
$(top_builddir)/src/libsdr/libsdr.a \
$(top_builddir)/src/libam/libam.a \
$(top_builddir)/src/libfft/libfft.a \
$(UHD_LIBS) \
$(SOAPY_LIBS)

@ -39,6 +39,7 @@ endif
if HAVE_SDR
radiocom2000_LDADD += \
$(top_builddir)/src/libsdr/libsdr.a \
$(top_builddir)/src/libam/libam.a \
$(top_builddir)/src/libfft/libfft.a \
$(UHD_LIBS) \
$(SOAPY_LIBS)

@ -383,9 +383,11 @@ int main(int argc, char *argv[])
}
double tx_frequencies[1], rx_frequencies[1];
int am[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);
am[0] = 0;
sdr = sdr_open(NULL, tx_frequencies, rx_frequencies, am, 1, 0.0, samplerate, latspl, 0.0, 0.0, 0.0);
if (!sdr)
goto error;
sdr_start(sdr);

@ -98,7 +98,7 @@ int radio_init(radio_t *radio, int latspl, int samplerate, const char *tx_wave_f
/* 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);
radio->tx_sound = sound_open(tx_audiodev, NULL, NULL, NULL, radio->tx_audio_channels, 0.0, radio->tx_audio_samplerate, radio->latspl, 1.0, 0.0, 2.0);
if (!radio->tx_sound) {
rc = -EIO;
PDEBUG(DRADIO, DEBUG_ERROR, "Failed to open sound device!\n");
@ -164,7 +164,7 @@ int radio_init(radio_t *radio, int latspl, int samplerate, const char *tx_wave_f
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);
radio->rx_sound = sound_open(rx_audiodev, NULL, NULL, NULL, radio->rx_audio_channels, 0.0, radio->rx_audio_samplerate, radio->latspl, 1.0, 0.0, 2.0);
if (!radio->rx_sound) {
rc = -EIO;
PDEBUG(DRADIO, DEBUG_ERROR, "Failed to open sound device!\n");

@ -39,6 +39,7 @@ endif
if HAVE_SDR
tacs_LDADD += \
$(top_builddir)/src/libsdr/libsdr.a \
$(top_builddir)/src/libam/libam.a \
$(top_builddir)/src/libfft/libfft.a \
$(UHD_LIBS) \
$(SOAPY_LIBS)

@ -81,6 +81,7 @@ test_dms_LDADD += \
$(top_builddir)/src/libsdr/libsdr.a \
$(top_builddir)/src/libfft/libfft.a \
$(top_builddir)/src/libfm/libfm.a \
$(top_builddir)/src/libam/libam.a \
$(UHD_LIBS) \
$(SOAPY_LIBS)
endif
@ -115,6 +116,7 @@ test_sms_LDADD += \
$(top_builddir)/src/libsdr/libsdr.a \
$(top_builddir)/src/libfft/libfft.a \
$(top_builddir)/src/libfm/libfm.a \
$(top_builddir)/src/libam/libam.a \
$(UHD_LIBS) \
$(SOAPY_LIBS)
endif

@ -31,7 +31,8 @@ osmotv_LDADD = \
if HAVE_SDR
osmotv_LDADD += \
$(top_builddir)/src/libsdr/libsdr.a
$(top_builddir)/src/libsdr/libsdr.a \
$(top_builddir)/src/libam/libam.a
endif
osmotv_LDADD += \

@ -321,9 +321,11 @@ static void tx_bas(sample_t *sample_bas, __attribute__((__unused__)) sample_t *s
}
double tx_frequencies[1], rx_frequencies[1];
int am[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);
am[0] = 0;
sdr = sdr_open(NULL, tx_frequencies, rx_frequencies, am, 1, 0.0, samplerate, latspl, 0.0, 0.0, 0.0);
if (!sdr)
goto error;
sdr_start(sdr);

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