294 lines
8.1 KiB
C
294 lines
8.1 KiB
C
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/* SDR processing
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*
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* (C) 2017 by Andreas Eversberg <jolly@eversberg.eu>
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* All Rights Reserved
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*
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* This program is free software: you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation, either version 3 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program. If not, see <http://www.gnu.org/licenses/>.
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*/
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#include <stdlib.h>
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#include <stdint.h>
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#include <string.h>
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#include <math.h>
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#include "filter.h"
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#include "sdr.h"
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#ifdef HAVE_UHD
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#include "uhd.h"
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#endif
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#include "debug.h"
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//#define FAST_SINE
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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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double offset; /* offset to calculated center frequency */
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double tx_phase; /* current phase of FM (used to shift and modulate ) */
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double rx_rot; /* rotation step per sample to shift rx frequency (used to shift) */
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double rx_phase; /* current rotation phase (used to shift) */
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double rx_last_phase; /* last phase of FM (used to demodulate) */
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filter_lowpass_t rx_lp[2]; /* filters received IQ signal */
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} sdr_chan_t;
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typedef struct sdr {
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sdr_chan_t *chan;
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double spl_deviation; /* how to convert a sample step into deviation (Hz) */
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int channels; /* number of frequencies */
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double samplerate; /* IQ rate */
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double amplitude; /* amplitude of each carrier */
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} sdr_t;
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static const char *sdr_device_args;
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static double sdr_rx_gain, sdr_tx_gain;
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#ifdef FAST_SINE
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static float sdr_sine[256];
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#endif
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int sdr_init(const char *device_args, double rx_gain, double tx_gain)
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{
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#ifdef FAST_SINE
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int i;
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for (i = 0; i < 256; i++) {
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sdr_sine[i] = sin(2.0*M_PI*i/256);
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}
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#endif
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sdr_device_args = strdup(device_args);
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sdr_rx_gain = rx_gain;
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sdr_tx_gain = tx_gain;
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return 0;
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}
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void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_frequency, double *rx_frequency, int channels, int samplerate, double bandwidth, double sample_deviation)
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{
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sdr_t *sdr;
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double center_frequency;
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int rc;
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int c;
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if (channels < 1) {
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PDEBUG(DSDR, DEBUG_ERROR, "No channel given, please fix!\n");
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abort();
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}
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sdr = calloc(sizeof(*sdr), 1);
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if (!sdr) {
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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->channels = channels;
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sdr->samplerate = samplerate;
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sdr->spl_deviation = sample_deviation;
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sdr->amplitude = 0.4 / (double)channels; // FIXME: actual amplitude 0.1?
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/* create list of channel states */
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sdr->chan = calloc(sizeof(*sdr->chan), channels);
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if (!sdr->chan) {
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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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for (c = 0; c < channels; c++) {
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PDEBUG(DSDR, DEBUG_INFO, "Frequency #%d: TX = %.6f MHz, RX = %.6f MHz\n", c, tx_frequency[c] / 1e6, rx_frequency[c] / 1e6);
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sdr->chan[c].tx_frequency = tx_frequency[c];
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sdr->chan[c].rx_frequency = rx_frequency[c];
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#warning check rx frequency is in range
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filter_lowpass_init(&sdr->chan[c].rx_lp[0], bandwidth, samplerate);
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filter_lowpass_init(&sdr->chan[c].rx_lp[1], bandwidth, samplerate);
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}
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/* calculate required bandwidth (IQ rate) */
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if (channels == 1) {
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PDEBUG(DSDR, DEBUG_INFO, "Single frequency, so we use sample rate as IQ bandwidth: %.6f MHz\n", sdr->samplerate / 1e6);
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center_frequency = sdr->chan[0].tx_frequency;
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} else {
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double low_frequency = sdr->chan[0].tx_frequency, high_frequency = sdr->chan[0].tx_frequency, range;
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for (c = 1; c < channels; c++) {
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if (sdr->chan[c].tx_frequency < low_frequency)
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low_frequency = sdr->chan[c].tx_frequency;
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if (sdr->chan[c].tx_frequency > high_frequency)
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high_frequency = sdr->chan[c].tx_frequency;
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}
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range = high_frequency - low_frequency;
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PDEBUG(DSDR, DEBUG_INFO, "Range between frequencies: %.6f MHz\n", range / 1e6);
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if (range * 2 > sdr->samplerate) {
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// why that? actually i don't know. i just want to be safe....
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PDEBUG(DSDR, DEBUG_NOTICE, "The sample rate must be at least twice the range between frequencies. Please increment samplerate!\n");
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goto error;
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}
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center_frequency = (high_frequency + low_frequency) / 2.0;
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}
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PDEBUG(DSDR, DEBUG_INFO, "Using center frequency: %.6f MHz\n", center_frequency / 1e6);
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for (c = 0; c < channels; c++) {
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sdr->chan[c].offset = sdr->chan[c].tx_frequency - center_frequency;
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sdr->chan[c].rx_rot = 2 * M_PI * -sdr->chan[c].offset / sdr->samplerate;
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PDEBUG(DSDR, DEBUG_INFO, "Frequency #%d offset: %.6f MHz\n", c, sdr->chan[c].offset / 1e6);
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}
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PDEBUG(DSDR, DEBUG_INFO, "Using gain: TX %.1f dB, RX %.1f dB\n", sdr_tx_gain, sdr_rx_gain);
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#ifdef HAVE_UHD
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#warning hack
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rc = uhd_open(sdr_device_args, center_frequency, center_frequency - sdr->chan[0].tx_frequency + sdr->chan[0].rx_frequency, sdr->samplerate, sdr_rx_gain, sdr_tx_gain);
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if (rc)
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goto error;
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#endif
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return sdr;
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error:
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sdr_close(sdr);
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return NULL;
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}
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void sdr_close(void *inst)
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{
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sdr_t *sdr = (sdr_t *)inst;
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#ifdef HAVE_UHD
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uhd_close();
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#endif
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if (sdr) {
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free(sdr->chan);
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free(sdr);
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sdr = NULL;
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}
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}
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int sdr_write(void *inst, int16_t **samples, int num, int channels)
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{
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sdr_t *sdr = (sdr_t *)inst;
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float buff[num * 2];
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int c, s, ss;
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double rate, phase, amplitude, dev;
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int sent;
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if (channels != sdr->channels) {
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PDEBUG(DSDR, DEBUG_ERROR, "Invalid number of channels, please fix!\n");
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abort();
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}
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/* process all channels */
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rate = sdr->samplerate;
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amplitude = sdr->amplitude;
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memset(buff, 0, sizeof(buff));
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for (c = 0; c < channels; c++) {
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/* modulate */
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phase = sdr->chan[c].tx_phase;
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for (s = 0, ss = 0; s < num; s++) {
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/* deviation is defined by the sample value and the offset */
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dev = sdr->chan[c].offset + (double)samples[c][s] * sdr->spl_deviation;
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#ifdef FAST_SINE
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phase += 256.0 * dev / rate;
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if (phase < 0.0)
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phase += 256.0;
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if (phase >= 256.0)
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phase -= 256.0;
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buff[ss++] += sdr_sine[((int)phase + 64) & 0xff] * amplitude;
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buff[ss++] += sdr_sine[(int)phase & 0xff] * amplitude;
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#else
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phase += 2.0 * M_PI * dev / rate;
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if (phase < 0.0)
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phase += 2.0 * M_PI;
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if (phase >= 2.0 * M_PI)
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phase -= 2.0 * M_PI;
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buff[ss++] += cos(phase) * amplitude;
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buff[ss++] += sin(phase) * amplitude;
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#endif
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}
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sdr->chan[c].tx_phase = phase;
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}
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#ifdef HAVE_UHD
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sent = uhd_send(buff, num);
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#endif
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if (sent < 0)
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return sent;
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return sent;
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}
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int sdr_read(void *inst, int16_t **samples, int num, int channels)
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{
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sdr_t *sdr = (sdr_t *)inst;
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float buff[num * 2];
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double I[num], Q[num], i, q;
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int count;
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int c, s, ss;
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double phase, rot, last_phase, spl, dev, rate;
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rate = sdr->samplerate;
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#ifdef HAVE_UHD
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count = uhd_receive(buff, num);
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#endif
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if (count <= 0)
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return count;
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for (c = 0; c < channels; c++) {
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rot = sdr->chan[c].rx_rot;
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phase = sdr->chan[c].rx_phase;
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for (s = 0, ss = 0; s < count; s++) {
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phase += rot;
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i = buff[ss++];
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q = buff[ss++];
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I[s] = i * cos(phase) - q * sin(phase);
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Q[s] = i * sin(phase) + q * cos(phase);
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}
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sdr->chan[c].rx_phase = phase;
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#warning eine interation von 2 f<>hrt zu m<>ll (2. kanal gespiegeltes audio), muss man genauer mal analysieren
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filter_lowpass_process(&sdr->chan[c].rx_lp[0], I, count, 1);
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filter_lowpass_process(&sdr->chan[c].rx_lp[1], Q, count, 1);
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last_phase = sdr->chan[c].rx_last_phase;
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for (s = 0; s < count; s++) {
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phase = atan2(I[s], Q[s]);
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dev = (phase - last_phase) / 2 / M_PI;
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last_phase = phase;
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if (dev < -0.49)
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dev += 1.0;
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else if (dev > 0.49)
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dev -= 1.0;
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dev *= rate;
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spl = dev / sdr->spl_deviation;
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if (spl > 32766.0)
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spl = 32766.0;
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else if (spl < -32766.0)
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spl = -32766.0;
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samples[c][s] = spl;
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}
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sdr->chan[c].rx_last_phase = last_phase;
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}
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return count;
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}
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/* how many delay (in audio sample duration) do we have in the buffer */
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int sdr_get_inbuffer(void __attribute__((__unused__)) *inst)
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{
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// sdr_t *sdr = (sdr_t *)inst;
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int count;
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#ifdef HAVE_UHD
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count = uhd_get_inbuffer();
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#endif
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if (count < 0)
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return count;
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return count;
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}
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