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

This radio can be a receiver or a transmitter or both simultaniously.

9 files changed, 1327 insertions, 0 deletions

diff --git a/.gitignore b/.gitignore
index 8102018..574eb1f 100644
--- a/.gitignore
+++ b/.gitignore
@@ -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
diff --git a/configure.ac b/configure.ac
index 62e67a3..aa34386 100644
--- a/configure.ac
+++ b/configure.ac
@@ -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
diff --git a/src/Makefile.am b/src/Makefile.am
index 594dcf9..aa068e1 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -47,5 +47,6 @@ SUBDIRS += \
 	r2000 \
 	jolly \
 	tv \
+	radio \
 	test
 
diff --git a/src/libdebug/debug.c b/src/libdebug/debug.c
index e4dcfcc..fd891ec 100644
--- a/src/libdebug/debug.c
+++ b/src/libdebug/debug.c
@@ -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 }
 };
 
diff --git a/src/libdebug/debug.h b/src/libdebug/debug.h
index 782b56c..8a76d45 100644
--- a/src/libdebug/debug.h
+++ b/src/libdebug/debug.h
@@ -25,6 +25,7 @@
 #define DUHD		18
 #define DSOAPY		19
 #define DWAVE		20
+#define DRADIO		21
 
 void get_win_size(int *w, int *h);
 
diff --git a/src/radio/Makefile.am b/src/radio/Makefile.am
new file mode 100644
index 0000000..24c65da
--- /dev/null
+++ b/src/radio/Makefile.am
@@ -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
+
diff --git a/src/radio/main.c b/src/radio/main.c
new file mode 100644
index 0000000..d576ffe
--- /dev/null
+++ b/src/radio/main.c
@@ -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;
+}
+
diff --git a/src/radio/radio.c b/src/radio/radio.c
new file mode 100644
index 0000000..48c8daf
--- /dev/null
+++ b/src/radio/radio.c
@@ -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;
+}
+
diff --git a/src/radio/radio.h b/src/radio/radio.h
new file mode 100644
index 0000000..5bb5239
--- /dev/null
+++ b/src/radio/radio.h
@@ -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 */
+	void		*tx_sound;		/* sound card process */
+	void		*rx_sound;		/* sound card process */
+	jitter_t	tx_dejitter[2];		/* jitter buffer when reading from sound card */
+	jitter_t	rx_dejitter[2];		/* jitter buffer when writing to sound card */
+	sample_t	*testtone[2];		/* test tone sample */
+	int		testtone_length;
+	int		testtone_pos;
+	/* signal stage */
+	double		signal_samplerate;
+	double		signal_bandwidth;
+	samplerate_t	tx_resampler[2];	/* resampling from audio rate to signal rate (two channels) */
+	samplerate_t	rx_resampler[2];	/* resampling from signal rate to audi rate (two channels) */
+	emphasis_t	fm_emphasis[2];		/* FM pre emphasis */
+	double		fm_deviation;		/* deviation of fm signal */
+	fm_mod_t	fm_mod;			/* FM modulation */
+	fm_demod_t	fm_demod;		/* FM modulation */
+	double		pilot_phasestep;	/* phase change of pilot tone for each sample */
+	double		tx_pilot_phase;		/* current phase of tx sine */
+	double		rx_pilot_phase;		/* current phase of rx mixer */
+	iir_filter_t	tx_dc_removal[2];	/* AM/FM DC level removal */
+	iir_filter_t	tx_am_bw_limit;		/* AM bandwidth limiter */
+	iir_filter_t	rx_lp_pilot_I;		/* low pass filter for pilot tone extraction */
+	iir_filter_t	rx_lp_pilot_Q;		/* low pass filter for pilot tone extraction */
+	iir_filter_t	rx_lp_sum;		/* filter sum signal of stereo */
+	iir_filter_t	rx_lp_diff;		/* filter differential signal of stereo */
+	am_mod_t	am_mod;			/* AM modulation */
+	am_demod_t	am_demod;		/* AM modulation */
+	/* buffers */
+	sample_t	*audio_buffer;
+	int		audio_buffer_size;
+	sample_t	*signal_buffer;
+	uint8_t		*signal_power_buffer;
+	int		signal_buffer_size;
+	sample_t	*I_buffer;
+	sample_t	*Q_buffer;
+	sample_t	*carrier_buffer;
+} radio_t;
+
+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, int stereo, int rds, int rds2);
+void radio_exit(radio_t *radio);
+int radio_start(radio_t *radio);
+int radio_tx(radio_t *radio, float *baseband, int num);
+int radio_rx(radio_t *radio, float *baseband, int num);
+