Add support for AM to libmobile and libsdr

src/amps/Makefile.am

@@ -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)

src/anetz/Makefile.am

@@ -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)

src/bnetz/Makefile.am

@@ -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)

src/bnetz/dialer.c

@@ -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;

src/cnetz/Makefile.am

@@ -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)

src/datenklo/datenklo.c

@@ -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;

src/imts/Makefile.am

@@ -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)

src/imts/dialer.c

@@ -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;

src/jolly/Makefile.am

@@ -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)

src/jtacs/Makefile.am

@@ -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)

src/libam/am.c

@@ -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;

src/libmncc/mncc_console.c

@@ -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;

src/libmobile/sender.c

@@ -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;

src/libmobile/sender.h

@@ -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);

src/libsdr/sdr.c

@@ -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);
 		}
 	}
 

src/libsdr/sdr.h

@@ -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);

src/libsound/sound.h

@@ -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);

src/libsound/sound_alsa.c

@@ -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;

src/nmt/Makefile.am

@@ -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)

src/r2000/Makefile.am

@@ -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)

src/radio/main.c

@@ -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);

src/radio/radio.c

@@ -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");

src/tacs/Makefile.am

@@ -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)

src/test/Makefile.am

@@ -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

src/tv/Makefile.am

@@ -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 += \

src/tv/main.c

@@ -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);