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path: root/src/common/sdr.c
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/* SDR processing
 *
 * (C) 2017 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 <stdlib.h>
#include <stdint.h>
#include <string.h>
#include <math.h>
#include "filter.h"
#include "sender.h"
#ifdef HAVE_UHD
#include "uhd.h"
#endif
#include "debug.h"

//#define FAST_SINE

typedef struct sdr_chan {
	double tx_frequency;	/* frequency used */
	double rx_frequency;	/* frequency used */
	double offset;		/* offset to calculated center frequency */
	double tx_phase;	/* current phase of FM (used to shift and modulate ) */
	double rx_rot;		/* rotation step per sample to shift rx frequency (used to shift) */
	double rx_phase;	/* current rotation phase (used to shift) */
	double rx_last_phase;	/* last phase of FM (used to demodulate) */
	filter_lowpass_t rx_lp[2]; /* filters received IQ signal */
} sdr_chan_t;

typedef struct sdr {
	sdr_chan_t *chan;	/* settings for all channels */
	int paging_channel;	/* if set, points to paging channel */
	sdr_chan_t paging_chan;	/* settings for extra paging channel */
	double spl_deviation;	/* how to convert a sample step into deviation (Hz) */
	int channels;		/* number of frequencies */
	double samplerate;	/* IQ rate */
	double amplitude;	/* amplitude of each carrier */
	wave_rec_t wave_rx_rec;
	wave_rec_t wave_tx_rec;
	wave_play_t wave_rx_play;
} sdr_t;

static const char *sdr_device_args;
static double sdr_rx_gain, sdr_tx_gain;
const char *sdr_write_iq_rx_wave, *sdr_write_iq_tx_wave, *sdr_read_iq_rx_wave;

#ifdef FAST_SINE
static float sdr_sine[256];
#endif

int sdr_init(const char *device_args, double rx_gain, double tx_gain, const char *write_iq_rx_wave, const char *write_iq_tx_wave, const char *read_iq_rx_wave)
{
#ifdef FAST_SINE
	int i;

	for (i = 0; i < 256; i++) {
		sdr_sine[i] = sin(2.0*M_PI*i/256);
	}
#endif

	sdr_device_args = strdup(device_args);
	sdr_rx_gain = rx_gain;
	sdr_tx_gain = tx_gain;
	sdr_write_iq_rx_wave = write_iq_rx_wave;
	sdr_write_iq_tx_wave = write_iq_tx_wave;
	sdr_read_iq_rx_wave = read_iq_rx_wave;

	return 0;
}

void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_frequency, double *rx_frequency, int channels, double paging_frequency, int samplerate, double bandwidth, double sample_deviation)
{
	sdr_t *sdr;
	double tx_center_frequency, rx_center_frequency;
	int rc;
	int c;

	display_iq_init(samplerate);

	if (channels < 1) {
		PDEBUG(DSDR, DEBUG_ERROR, "No channel given, please fix!\n");
		abort();
	}

	sdr = calloc(sizeof(*sdr), 1);
	if (!sdr) {
		PDEBUG(DSDR, DEBUG_ERROR, "NO MEM!\n");
		goto error;
	}
	sdr->channels = channels;
	sdr->samplerate = samplerate;
	sdr->spl_deviation = sample_deviation;
	sdr->amplitude = 0.4 / (double)channels; // FIXME: actual amplitude 0.1?

	/* special case where we use a paging frequency */
	if (paging_frequency) {
		/* add extra paging channel */
		sdr->paging_channel = channels;
	}

	/* create list of channel states */
	sdr->chan = calloc(sizeof(*sdr->chan), channels + (sdr->paging_channel != 0));
	if (!sdr->chan) {
		PDEBUG(DSDR, DEBUG_ERROR, "NO MEM!\n");
		goto error;
	}
	for (c = 0; c < channels; c++) {
		PDEBUG(DSDR, DEBUG_INFO, "Frequency #%d: TX = %.6f MHz, RX = %.6f MHz\n", c, tx_frequency[c] / 1e6, rx_frequency[c] / 1e6);
		sdr->chan[c].tx_frequency = tx_frequency[c];
		sdr->chan[c].rx_frequency = rx_frequency[c];
		filter_lowpass_init(&sdr->chan[c].rx_lp[0], bandwidth, samplerate);
		filter_lowpass_init(&sdr->chan[c].rx_lp[1], bandwidth, samplerate);
	}
	if (sdr->paging_channel) {
		PDEBUG(DSDR, DEBUG_INFO, "Paging Frequency: TX = %.6f MHz\n", paging_frequency / 1e6);
		sdr->chan[sdr->paging_channel].tx_frequency = paging_frequency;
	}

	/* calculate required bandwidth (IQ rate) */
	double tx_low_frequency = sdr->chan[0].tx_frequency, tx_high_frequency = sdr->chan[0].tx_frequency;
	double rx_low_frequency = sdr->chan[0].rx_frequency, rx_high_frequency = sdr->chan[0].rx_frequency;
	double range;
	for (c = 1; c < channels; c++) {
		if (sdr->chan[c].tx_frequency < tx_low_frequency)
			tx_low_frequency = sdr->chan[c].tx_frequency;
		if (sdr->chan[c].tx_frequency > tx_high_frequency)
			tx_high_frequency = sdr->chan[c].tx_frequency;
		if (sdr->chan[c].rx_frequency < rx_low_frequency)
			rx_low_frequency = sdr->chan[c].rx_frequency;
		if (sdr->chan[c].rx_frequency > rx_high_frequency)
			rx_high_frequency = sdr->chan[c].rx_frequency;
	}
	if (sdr->paging_channel) {
		if (sdr->chan[sdr->paging_channel].tx_frequency < tx_low_frequency)
			tx_low_frequency = sdr->chan[sdr->paging_channel].tx_frequency;
		if (sdr->chan[sdr->paging_channel].tx_frequency > tx_high_frequency)
			tx_high_frequency = sdr->chan[sdr->paging_channel].tx_frequency;
	}
	/* range of TX */
	range = tx_high_frequency - tx_low_frequency;
	if (range)
		PDEBUG(DSDR, DEBUG_DEBUG, "Range between all TX Frequencies: %.6f MHz\n", range / 1e6);
	if (range * 2 > sdr->samplerate) {
		// why that? actually i don't know. i just want to be safe....
		PDEBUG(DSDR, DEBUG_NOTICE, "The sample rate must be at least twice the range between frequencies.\n");
		PDEBUG(DSDR, DEBUG_NOTICE, "The given rate is %.6f MHz, but required rate must be >= %.6f MHz\n", sdr->samplerate / 1e6, range * 2.0 / 1e6);
		PDEBUG(DSDR, DEBUG_NOTICE, "Please increase samplerate!\n");
		goto error;
	}
	tx_center_frequency = (tx_high_frequency + tx_low_frequency) / 2.0;
	/* range of RX */
	range = rx_high_frequency - rx_low_frequency;
	if (range)
		PDEBUG(DSDR, DEBUG_DEBUG, "Range between all RX Frequencies: %.6f MHz\n", range / 1e6);
	if (range * 2.0 > sdr->samplerate) {
		// why that? actually i don't know. i just want to be safe....
		PDEBUG(DSDR, DEBUG_NOTICE, "The sample rate must be at least twice the range between frequencies. Please increment samplerate!\n");
		goto error;
	}
	rx_center_frequency = (rx_high_frequency + rx_low_frequency) / 2.0;
	PDEBUG(DSDR, DEBUG_INFO, "Using center frequency: TX %.6f MHz, RX %.6f\n", tx_center_frequency / 1e6, rx_center_frequency / 1e6);
	/* set offsets to center frequency */
	for (c = 0; c < channels; c++) {
		double rx_offset;
		sdr->chan[c].offset = sdr->chan[c].tx_frequency - tx_center_frequency;
		rx_offset = sdr->chan[c].rx_frequency - rx_center_frequency;
		sdr->chan[c].rx_rot = 2 * M_PI * -rx_offset / sdr->samplerate;
		PDEBUG(DSDR, DEBUG_DEBUG, "Frequency #%d: TX offset: %.6f MHz, RX offset: %.6f MHz\n", c, sdr->chan[c].offset / 1e6, rx_offset / 1e6);
	}
	if (sdr->paging_channel) {
		sdr->chan[sdr->paging_channel].offset = sdr->chan[sdr->paging_channel].tx_frequency - tx_center_frequency;
		PDEBUG(DSDR, DEBUG_DEBUG, "Paging Frequency: TX offset: %.6f MHz\n", sdr->chan[sdr->paging_channel].offset / 1e6);
	}
	PDEBUG(DSDR, DEBUG_INFO, "Using gain: TX %.1f dB, RX %.1f dB\n", sdr_tx_gain, sdr_rx_gain);

	if (sdr_write_iq_rx_wave) {
		rc = wave_create_record(&sdr->wave_rx_rec, sdr_write_iq_rx_wave, sdr->samplerate, 2);
		if (rc < 0) {
			PDEBUG(DSENDER, DEBUG_ERROR, "Failed to create WAVE recoding instance!\n");
			goto error;
		}
	}
	if (sdr_write_iq_tx_wave) {
		rc = wave_create_record(&sdr->wave_tx_rec, sdr_write_iq_tx_wave, sdr->samplerate, 2);
		if (rc < 0) {
			PDEBUG(DSENDER, DEBUG_ERROR, "Failed to create WAVE recoding instance!\n");
			goto error;
		}
	}
	if (sdr_read_iq_rx_wave) {
		rc = wave_create_playback(&sdr->wave_rx_play, sdr_read_iq_rx_wave, sdr->samplerate, 2);
		if (rc < 0) {
			PDEBUG(DSENDER, DEBUG_ERROR, "Failed to create WAVE playback instance!\n");
			goto error;
		}
	}

#ifdef HAVE_UHD
	rc = uhd_open(sdr_device_args, tx_center_frequency, rx_center_frequency, sdr->samplerate, sdr_rx_gain, sdr_tx_gain);
	if (rc)
		goto error;
#endif

	return sdr;

error:
	sdr_close(sdr);
	return NULL;
}

void sdr_close(void *inst)
{
	sdr_t *sdr = (sdr_t *)inst;

#ifdef HAVE_UHD
	uhd_close();
#endif

	if (sdr) {
		wave_destroy_record(&sdr->wave_rx_rec);
		wave_destroy_record(&sdr->wave_tx_rec);
		wave_destroy_playback(&sdr->wave_rx_play);
		free(sdr->chan);
		free(sdr);
		sdr = NULL;
	}
}

int sdr_write(void *inst, int16_t **samples, int num, enum paging_signal __attribute__((unused)) *paging_signal, int *on, int channels)
{
	sdr_t *sdr = (sdr_t *)inst;
	float buff[num * 2];
	int c, s, ss;
	double rate, offset, phase, amplitude, dev;
	int sent;

	if (channels != sdr->channels) {
		PDEBUG(DSDR, DEBUG_ERROR, "Invalid number of channels, please fix!\n");
		abort();
	}

	/* process all channels */
	rate = sdr->samplerate;
	amplitude = sdr->amplitude;
	memset(buff, 0, sizeof(buff));
	for (c = 0; c < channels; c++) {
		phase = sdr->chan[c].tx_phase;
		/* switch offset to paging channel, if requested */
		if (on[c] && sdr->paging_channel)
			offset = sdr->chan[sdr->paging_channel].offset;
		else
			offset = sdr->chan[c].offset;
		/* modulate */
		for (s = 0, ss = 0; s < num; s++) {
			/* deviation is defined by the sample value and the offset */
			dev = offset + (double)samples[c][s] * sdr->spl_deviation;
#ifdef FAST_SINE
			phase += 256.0 * dev / rate;
			if (phase < 0.0)
				phase += 256.0;
			if (phase >= 256.0)
				phase -= 256.0;
			buff[ss++] += sdr_sine[((int)phase + 64) & 0xff] * amplitude;
			buff[ss++] += sdr_sine[(int)phase & 0xff] * amplitude;
#else
			phase += 2.0 * M_PI * dev / rate;
			if (phase < 0.0)
				phase += 2.0 * M_PI;
			if (phase >= 2.0 * M_PI)
				phase -= 2.0 * M_PI;
			buff[ss++] += cos(phase) * amplitude;
			buff[ss++] += sin(phase) * amplitude;
#endif
		}
		sdr->chan[c].tx_phase = phase;
	}

	if (sdr->wave_tx_rec.fp) {
		int16_t spl[2][num], *spl_list[2] = { spl[0], spl[1] };
		for (s = 0, ss = 0; s < num; s++) {
			if (buff[ss] >= 1.0)
				spl[0][s] = 32767;
			else if (buff[ss] <= -1.0)
				spl[0][s] = -32767;
			else
				spl[0][s] = 32767.0 * buff[ss];
			ss++;
			if (buff[ss] >= 1.0)
				spl[1][s] = 32767;
			else if (buff[ss] <= -1.0)
				spl[1][s] = -32767;
			else
				spl[1][s] = 32767.0 * buff[ss];
			ss++;
		}
		wave_write(&sdr->wave_tx_rec, spl_list, num);
	}

#ifdef HAVE_UHD
	sent = uhd_send(buff, num);
#endif
	if (sent < 0)
		return sent;
	
	return sent;
}

int sdr_read(void *inst, int16_t **samples, int num, int channels)
{
	sdr_t *sdr = (sdr_t *)inst;
	float buff[num * 2];
	double I[num], Q[num], i, q;
	int count;
	int c, s, ss;
	double phase, rot, last_phase, spl, dev, rate;

	rate = sdr->samplerate;

#ifdef HAVE_UHD
	count = uhd_receive(buff, num);
#endif
	if (count <= 0)
		return count;

	if (sdr->wave_rx_rec.fp) {
		int16_t spl[2][count], *spl_list[2] = { spl[0], spl[1] };
		for (s = 0, ss = 0; s < count; s++) {
			if (buff[ss] >= 1.0)
				spl[0][s] = 32767;
			else if (buff[ss] <= -1.0)
				spl[0][s] = -32767;
			else
				spl[0][s] = 32767.0 * buff[ss];
			ss++;
			if (buff[ss] >= 1.0)
				spl[1][s] = 32767;
			else if (buff[ss] <= -1.0)
				spl[1][s] = -32767;
			else
				spl[1][s] = 32767.0 * buff[ss];
			ss++;
		}
		wave_write(&sdr->wave_rx_rec, spl_list, count);
	}
	if (sdr->wave_rx_play.fp) {
		int16_t spl[2][count], *spl_list[2] = { spl[0], spl[1] };
		wave_read(&sdr->wave_rx_play, spl_list, count);
		for (s = 0, ss = 0; s < count; s++) {
			buff[ss++] = (double)spl[0][s] / 32767.0;
			buff[ss++] = (double)spl[1][s] / 32767.0;
		}
	}
	display_iq(buff, count);

	for (c = 0; c < channels; c++) {
		rot = sdr->chan[c].rx_rot;
		phase = sdr->chan[c].rx_phase;
		for (s = 0, ss = 0; s < count; s++) {
			phase += rot;
			i = buff[ss++];
			q = buff[ss++];
			I[s] = i * cos(phase) - q * sin(phase);
			Q[s] = i * sin(phase) + q * cos(phase);
		}
		sdr->chan[c].rx_phase = phase;
		filter_lowpass_process(&sdr->chan[c].rx_lp[0], I, count, 1);
		filter_lowpass_process(&sdr->chan[c].rx_lp[1], Q, count, 1);
		last_phase = sdr->chan[c].rx_last_phase;
		for (s = 0; s < count; s++) {
			phase = atan2(Q[s], I[s]);
			dev = (phase - last_phase) / 2 / M_PI;
			last_phase = phase;
			if (dev < -0.49)
				dev += 1.0;
			else if (dev > 0.49)
				dev -= 1.0;
			dev *= rate;
			spl = dev / sdr->spl_deviation;
			if (spl > 32766.0)
				spl = 32766.0;
			else if (spl < -32766.0)
				spl = -32766.0;
			samples[c][s] = spl;
		}
		sdr->chan[c].rx_last_phase = last_phase;
	}

	return count;
}

/* how many delay (in audio sample duration) do we have in the buffer */
int sdr_get_inbuffer(void __attribute__((__unused__)) *inst)
{
//	sdr_t *sdr = (sdr_t *)inst;
	int count;

#ifdef HAVE_UHD
	count = uhd_get_inbuffer();
#endif
	if (count < 0)
		return count;

	return count;
}