Add support for AM to libmobile and libsdr
This commit is contained in:
parent
c2f14834e5
commit
150a77b69d
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@ -55,6 +55,7 @@ endif
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if HAVE_SDR
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amps_LDADD += \
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$(top_builddir)/src/libsdr/libsdr.a \
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$(top_builddir)/src/libam/libam.a \
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$(top_builddir)/src/libfft/libfft.a \
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$(UHD_LIBS) \
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$(SOAPY_LIBS)
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@ -44,6 +44,7 @@ endif
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if HAVE_SDR
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anetz_LDADD += \
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$(top_builddir)/src/libsdr/libsdr.a \
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$(top_builddir)/src/libam/libam.a \
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$(top_builddir)/src/libfft/libfft.a \
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$(UHD_LIBS) \
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$(SOAPY_LIBS)
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@ -61,6 +61,7 @@ endif
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if HAVE_SDR
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bnetz_LDADD += \
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$(top_builddir)/src/libsdr/libsdr.a \
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$(top_builddir)/src/libam/libam.a \
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$(top_builddir)/src/libfft/libfft.a \
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$(UHD_LIBS) \
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$(SOAPY_LIBS)
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@ -346,7 +346,7 @@ int main(int argc, char *argv[])
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#ifdef HAVE_ALSA
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/* init sound */
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audio = sound_open(audiodev, NULL, NULL, 1, 0.0, samplerate, latspl, 1.0, 4000.0);
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audio = sound_open(audiodev, NULL, NULL, NULL, 1, 0.0, samplerate, latspl, 1.0, 4000.0, 2.0);
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if (!audio) {
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PDEBUG(DBNETZ, DEBUG_ERROR, "No sound device!\n");
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goto exit;
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@ -44,6 +44,7 @@ endif
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if HAVE_SDR
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cnetz_LDADD += \
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$(top_builddir)/src/libsdr/libsdr.a \
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$(top_builddir)/src/libam/libam.a \
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$(top_builddir)/src/libfft/libfft.a \
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$(UHD_LIBS) \
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$(SOAPY_LIBS)
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@ -1395,7 +1395,7 @@ int datenklo_open_audio(datenklo_t *datenklo, const char *audiodev, int latency,
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#ifdef HAVE_ALSA
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/* init sound */
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datenklo->audio = sound_open(audiodev, NULL, NULL, channels, 0.0, datenklo->samplerate, datenklo->latspl, 1.0, 4000.0);
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datenklo->audio = sound_open(audiodev, NULL, NULL, NULL, channels, 0.0, datenklo->samplerate, datenklo->latspl, 1.0, 4000.0, 2.0);
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if (!datenklo->audio) {
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PDEBUG(DDATENKLO, DEBUG_ERROR, "No sound device!\n");
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return -EIO;
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@ -52,6 +52,7 @@ endif
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if HAVE_SDR
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imts_LDADD += \
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$(top_builddir)/src/libsdr/libsdr.a \
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$(top_builddir)/src/libam/libam.a \
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$(top_builddir)/src/libfft/libfft.a \
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$(UHD_LIBS) \
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$(SOAPY_LIBS)
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@ -307,7 +307,7 @@ int main(int argc, char *argv[])
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#ifdef HAVE_ALSA
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/* init sound */
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audio = sound_open(audiodev, NULL, NULL, 1, 0.0, samplerate, latspl, 1.0, 4000.0);
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audio = sound_open(audiodev, NULL, NULL, NULL, 1, 0.0, samplerate, latspl, 1.0, 4000.0, 2.0);
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if (!audio) {
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PDEBUG(DBNETZ, DEBUG_ERROR, "No sound device!\n");
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goto exit;
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@ -38,6 +38,7 @@ endif
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if HAVE_SDR
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jollycom_LDADD += \
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$(top_builddir)/src/libsdr/libsdr.a \
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$(top_builddir)/src/libam/libam.a \
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$(top_builddir)/src/libfft/libfft.a \
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$(UHD_LIBS) \
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$(SOAPY_LIBS)
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@ -38,6 +38,7 @@ endif
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if HAVE_SDR
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jtacs_LDADD += \
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$(top_builddir)/src/libsdr/libsdr.a \
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$(top_builddir)/src/libam/libam.a \
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$(top_builddir)/src/libfft/libfft.a \
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$(UHD_LIBS) \
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$(SOAPY_LIBS)
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@ -111,8 +111,8 @@ void am_modulate_complex(am_mod_t *mod, sample_t *amplitude, int num, float *bas
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else if (phase >= 65536.0)
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phase -= 65536.0;
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} else {
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*baseband++ = cos(phase) * vector;
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*baseband++ = sin(phase) * vector;
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*baseband++ += cos(phase) * vector;
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*baseband++ += sin(phase) * vector;
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phase += rot;
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if (phase < 0.0)
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phase += 2.0 * M_PI;
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@ -299,7 +299,7 @@ int console_open_audio(int __attribute__((unused)) latspl)
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#ifdef HAVE_ALSA
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/* open sound device for call control */
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/* use factor 1.4 of speech level for complete range of sound card */
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console.sound = sound_open(console.audiodev, NULL, NULL, 1, 0.0, console.samplerate, latspl, 1.4, 4000.0);
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console.sound = sound_open(console.audiodev, NULL, NULL, NULL, 1, 0.0, console.samplerate, latspl, 1.4, 4000.0, 2.0);
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if (!console.sound) {
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PDEBUG(DSENDER, DEBUG_ERROR, "No sound device!\n");
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return -EIO;
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@ -183,36 +183,38 @@ int sender_open_audio(int latspl)
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channels++;
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}
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double tx_f[channels], rx_f[channels], paging_frequency = 0.0;
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int am[channels];
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for (i = 0, inst = master; inst; i++, inst = inst->slave) {
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tx_f[i] = inst->sendefrequenz;
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rx_f[i] = inst->empfangsfrequenz;
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am[i] = inst->am;
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if (inst->ruffrequenz)
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paging_frequency = inst->ruffrequenz;
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}
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if (master->write_rx_wave) {
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rc = wave_create_record(&master->wave_rx_rec, master->write_rx_wave, master->samplerate, channels, master->max_deviation);
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rc = wave_create_record(&master->wave_rx_rec, master->write_rx_wave, master->samplerate, channels, (master->max_deviation) ?: 1.0);
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if (rc < 0) {
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PDEBUG(DSENDER, DEBUG_ERROR, "Failed to create WAVE recoding instance!\n");
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return rc;
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}
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}
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if (master->write_tx_wave) {
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rc = wave_create_record(&master->wave_tx_rec, master->write_tx_wave, master->samplerate, channels, master->max_deviation);
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rc = wave_create_record(&master->wave_tx_rec, master->write_tx_wave, master->samplerate, channels, (master->max_deviation) ?: 1.0);
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if (rc < 0) {
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PDEBUG(DSENDER, DEBUG_ERROR, "Failed to create WAVE recoding instance!\n");
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return rc;
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}
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}
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if (master->read_rx_wave) {
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rc = wave_create_playback(&master->wave_rx_play, master->read_rx_wave, &master->samplerate, &channels, master->max_deviation);
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rc = wave_create_playback(&master->wave_rx_play, master->read_rx_wave, &master->samplerate, &channels, (master->max_deviation) ?: 1.0);
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if (rc < 0) {
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PDEBUG(DSENDER, DEBUG_ERROR, "Failed to create WAVE playback instance!\n");
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return rc;
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}
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}
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if (master->read_tx_wave) {
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rc = wave_create_playback(&master->wave_tx_play, master->read_tx_wave, &master->samplerate, &channels, master->max_deviation);
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rc = wave_create_playback(&master->wave_tx_play, master->read_tx_wave, &master->samplerate, &channels, (master->max_deviation) ?: 1.0);
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if (rc < 0) {
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PDEBUG(DSENDER, DEBUG_ERROR, "Failed to create WAVE playback instance!\n");
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return rc;
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@ -220,7 +222,7 @@ int sender_open_audio(int latspl)
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}
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/* open device */
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master->audio = master->audio_open(master->audiodev, tx_f, rx_f, channels, paging_frequency, master->samplerate, latspl, master->max_deviation, master->max_modulation);
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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);
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if (!master->audio) {
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PDEBUG(DSENDER, DEBUG_ERROR, "No device for transceiver!\n");
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return -EIO;
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@ -293,6 +295,19 @@ void sender_set_fm(sender_t *sender, double max_deviation, double max_modulation
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PDEBUG_CHAN(DSENDER, DEBUG_DEBUG, "Deviation at speech level: %.1f kHz\n", speech_deviation / 1000.0);
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}
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/* set amplitude modulation and parameters */
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void sender_set_am(sender_t *sender, double max_modulation, double speech_level, double max_display, double modulation_index)
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{
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sender->am = 1;
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sender->max_deviation = 0;
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sender->max_modulation = max_modulation;
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sender->speech_deviation = speech_level;
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sender->max_display = max_display;
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sender->modulation_index = modulation_index;
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PDEBUG_CHAN(DSENDER, DEBUG_DEBUG, "Modulation degree: %.0f %%, Maximum modulation: %.1f kHz\n", modulation_index / 100.0, max_modulation / 1000.0);
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}
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static void gain_samples(sample_t *samples, int length, double gain)
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{
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int i;
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@ -33,16 +33,18 @@ typedef struct sender {
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double empfangsfrequenz; /* receiver frequency */
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double ruffrequenz; /* special paging frequency used for B-Netz */
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/* fm levels */
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double max_deviation; /* max frequency deviation */
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/* FM/AM levels */
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int am; /* use AM instead of FM */
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double max_deviation; /* max frequency deviation / level */
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double max_modulation; /* max frequency modulated */
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double speech_deviation; /* deviation of 1000 Hz reference tone at speech level */
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double speech_deviation; /* deviation / level of 1000 Hz reference tone at speech level */
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double modulation_index; /* AM modulation index */
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double max_display; /* level of displaying wave form */
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/* audio */
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void *audio;
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char audiodev[64]; /* audio device name (alsa or sdr) */
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void *(*audio_open)(const char *, double *, double *, int, double, int, int, double, double);
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void *(*audio_open)(const char *, double *, double *, int *, int, double, int, int, double, double, double);
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int (*audio_start)(void *);
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void (*audio_close)(void *);
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int (*audio_write)(void *, sample_t **, uint8_t **, int, enum paging_signal *, int *, int);
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@ -93,6 +95,7 @@ extern int cant_recover;
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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);
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void sender_destroy(sender_t *sender);
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void sender_set_fm(sender_t *sender, double max_deviation, double max_modulation, double speech_deviation, double max_display);
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void sender_set_am(sender_t *sender, double max_modulation, double speech_deviation, double max_display, double modulation_index);
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int sender_open_audio(int latspl);
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int sender_start_audio(void);
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void process_sender_audio(sender_t *sender, int *quit, int latspl);
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@ -30,6 +30,7 @@ enum paging_signal;
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#include <unistd.h>
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#include "../libsample/sample.h"
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#include "../libfm/fm.h"
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#include "../libam/am.h"
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#include "../libtimer/timer.h"
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#include "../libmobile/sender.h"
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#include "sdr_config.h"
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@ -71,8 +72,11 @@ typedef struct sdr_thread {
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typedef struct sdr_chan {
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double tx_frequency; /* frequency used */
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double rx_frequency; /* frequency used */
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fm_mod_t mod; /* modulator instance */
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fm_demod_t demod; /* demodulator instance */
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int am; /* use AM instead of FM */
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fm_mod_t fm_mod; /* modulator instance */
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fm_demod_t fm_demod; /* demodulator instance */
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am_mod_t am_mod; /* modulator instance */
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am_demod_t am_demod; /* demodulator instance */
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dispmeasparam_t *dmp_rf_level;
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dispmeasparam_t *dmp_freq_offset;
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dispmeasparam_t *dmp_deviation;
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@ -94,9 +98,10 @@ typedef struct sdr {
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wave_rec_t wave_tx_rec;
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wave_play_t wave_rx_play;
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wave_play_t wave_tx_play;
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float *modbuff; /* buffer for FM transmodulation */
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float *modbuff; /* buffer for transmodulation */
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sample_t *modbuff_I;
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sample_t *modbuff_Q;
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sample_t *modbuff_carrier;
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sample_t *wavespl0; /* sample buffer for wave generation */
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sample_t *wavespl1;
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} sdr_t;
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@ -131,7 +136,7 @@ static void show_spectrum(const char *direction, double halfbandwidth, double ce
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PDEBUG(DSDR, DEBUG_INFO, "Frequency P = %.4f MHz (Paging Frequency)\n", paging_frequency / 1e6);
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}
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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)
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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)
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{
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sdr_t *sdr;
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int threads = 1, oversample = 1; /* always use threads */
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@ -231,6 +236,11 @@ void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_freq
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PDEBUG(DSDR, DEBUG_ERROR, "NO MEM!\n");
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goto error;
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}
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sdr->modbuff_carrier = calloc(sdr->latspl, sizeof(*sdr->modbuff_carrier));
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if (!sdr->modbuff_carrier) {
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PDEBUG(DSDR, DEBUG_ERROR, "NO MEM!\n");
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goto error;
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}
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sdr->wavespl0 = calloc(sdr->latspl, sizeof(*sdr->wavespl0));
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if (!sdr->wavespl0) {
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PDEBUG(DSDR, DEBUG_ERROR, "NO MEM!\n");
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@ -297,7 +307,14 @@ void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_freq
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double tx_offset;
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tx_offset = sdr->chan[c].tx_frequency - tx_center_frequency;
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PDEBUG(DSDR, DEBUG_DEBUG, "Frequency #%d: TX offset: %.6f MHz\n", c, tx_offset / 1e6);
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rc = fm_mod_init(&sdr->chan[c].mod, samplerate, tx_offset, sdr->amplitude);
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sdr->chan[c].am = am[c];
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if (am[c]) {
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double gain, bias;
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gain = modulation_index / 2.0;
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bias = 1.0 - gain;
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rc = am_mod_init(&sdr->chan[c].am_mod, samplerate, tx_offset, sdr->amplitude * gain, sdr->amplitude * bias);
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} else
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rc = fm_mod_init(&sdr->chan[c].fm_mod, samplerate, tx_offset, sdr->amplitude);
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if (rc < 0)
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goto error;
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}
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@ -305,7 +322,7 @@ void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_freq
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double tx_offset;
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tx_offset = sdr->chan[sdr->paging_channel].tx_frequency - tx_center_frequency;
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PDEBUG(DSDR, DEBUG_DEBUG, "Paging Frequency: TX offset: %.6f MHz\n", tx_offset / 1e6);
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rc = fm_mod_init(&sdr->chan[sdr->paging_channel].mod, samplerate, tx_offset, sdr->amplitude);
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rc = fm_mod_init(&sdr->chan[sdr->paging_channel].fm_mod, samplerate, tx_offset, sdr->amplitude);
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if (rc < 0)
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goto error;
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}
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@ -400,7 +417,11 @@ void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_freq
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double rx_offset;
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rx_offset = sdr->chan[c].rx_frequency - rx_center_frequency;
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PDEBUG(DSDR, DEBUG_DEBUG, "Frequency #%d: RX offset: %.6f MHz\n", c, rx_offset / 1e6);
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rc = fm_demod_init(&sdr->chan[c].demod, samplerate, rx_offset, bandwidth);
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sdr->chan[c].am = am[c];
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if (am[c])
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rc = am_demod_init(&sdr->chan[c].am_demod, samplerate, rx_offset, bandwidth, 1.0 / modulation_index);
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else
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rc = fm_demod_init(&sdr->chan[c].fm_demod, samplerate, rx_offset, bandwidth / 2.0);
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if (rc < 0)
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goto error;
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}
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@ -428,8 +449,10 @@ void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_freq
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if (!sender)
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continue;
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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);
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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);
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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);
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if (!am[c]) {
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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);
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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);
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}
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}
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}
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@ -700,6 +723,7 @@ void sdr_close(void *inst)
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free(sdr->modbuff);
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free(sdr->modbuff_I);
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free(sdr->modbuff_Q);
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free(sdr->modbuff_carrier);
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free(sdr->wavespl0);
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free(sdr->wavespl1);
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wave_destroy_record(&sdr->wave_rx_rec);
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@ -710,11 +734,13 @@ void sdr_close(void *inst)
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int c;
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|
||||
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);
|
||||
|
|
Loading…
Reference in New Issue