Add libam, a library to do AM modulation and demodulation
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@ -44,6 +44,7 @@ src/libmncc/libmncc.a
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src/libsound/libsound.a
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src/libsdr/libsdr.a
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src/libsample/libsample.a
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src/libam/libam.a
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src/libclipper/libclipper.a
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src/anetz/libgermanton.a
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src/anetz/anetz
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@ -65,6 +65,7 @@ AC_OUTPUT(
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src/libscrambler/Makefile
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src/libemphasis/Makefile
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src/libfsk/Makefile
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src/libam/Makefile
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src/libfm/Makefile
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src/libfilter/Makefile
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src/libwave/Makefile
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@ -18,6 +18,7 @@ SUBDIRS = \
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libscrambler \
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libemphasis \
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libfsk \
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libam \
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libfm \
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libfilter \
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libwave \
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@ -0,0 +1,6 @@
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AM_CPPFLAGS = -Wall -Wextra -g $(all_includes)
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noinst_LIBRARIES = libam.a
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libam_a_SOURCES = \
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am.c
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@ -0,0 +1,133 @@
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/* AM modulation and de-modulation
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*
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* (C) 2018 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 <stdint.h>
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#include <string.h>
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#include <math.h>
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#include "../libsample/sample.h"
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#include "am.h"
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#define CARRIER_FILTER 30.0
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/* Amplitude modulation in SDR:
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* Just use the base band (audio signal) as real value, and 0.0 as imaginary
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* value. The you have two side bands. Be sure to have a DC level, so you
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* have a carrier.
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*/
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int am_mod_init(am_mod_t *mod, double samplerate, double offset, double gain, double bias)
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{
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memset(mod, 0, sizeof(*mod));
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mod->gain = gain;
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mod->bias = bias;
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mod->phasestep = 2.0 * M_PI * offset / samplerate;
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return 0;
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}
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void am_mod_exit(am_mod_t __attribute__((unused)) *mod)
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{
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}
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void am_modulate_complex(am_mod_t *mod, sample_t *amplitude, int num, float *baseband)
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{
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int s;
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double vector;
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double phasestep = mod->phasestep;
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double phase = mod->phase;
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double gain = mod->gain;
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double bias = mod->bias;
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for (s = 0; s < num; s++) {
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vector = *amplitude++ * gain + bias;
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*baseband++ = cos(phase) * vector;
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*baseband++ = sin(phase) * vector;
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phase += phasestep;
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if (phase < 0.0)
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phase += 2.0 * M_PI;
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else if (phase >= 2.0 * M_PI)
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phase -= 2.0 * M_PI;
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}
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mod->phase = phase;
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}
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/* init AM demodulator */
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int am_demod_init(am_demod_t *demod, double samplerate, double offset, double bandwidth, double gain)
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{
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memset(demod, 0, sizeof(*demod));
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demod->gain = gain;
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demod->phasestep = 2 * M_PI * -offset / samplerate;
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/* use fourth order (2 iter) filter, since it is as fast as second order (1 iter) filter */
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iir_lowpass_init(&demod->lp[0], bandwidth, samplerate, 2);
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iir_lowpass_init(&demod->lp[1], bandwidth, samplerate, 2);
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/* filter carrier */
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iir_lowpass_init(&demod->lp[2], CARRIER_FILTER, samplerate, 1);
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return 0;
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}
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void am_demod_exit(am_demod_t __attribute__((unused)) *demod)
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{
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}
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/* do amplitude demodulation of baseband and write them to samples */
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void am_demodulate_complex(am_demod_t *demod, sample_t *amplitude, int length, float *baseband, sample_t *I, sample_t *Q, sample_t *carrier)
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{
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int s, ss;
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double phasestep = demod->phasestep;
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double phase = demod->phase;
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double gain = demod->gain;
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double i, q;
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double _sin, _cos;
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/* rotate spectrum */
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for (s = 0, ss = 0; s < length; s++) {
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i = baseband[ss++];
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q = baseband[ss++];
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_sin = sin(phase);
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_cos = cos(phase);
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phase += phasestep;
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if (phase < 0.0)
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phase += 2.0 * M_PI;
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else if (phase >= 2.0 * M_PI)
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phase -= 2.0 * M_PI;
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I[s] = i * _cos - q * _sin;
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Q[s] = i * _sin + q * _cos;
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}
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demod->phase = phase;
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/* filter bandwidth */
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iir_process(&demod->lp[0], I, length);
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iir_process(&demod->lp[1], Q, length);
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/* demod */
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for (s = 0; s < length; s++)
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amplitude[s] = carrier[s] = sqrt(I[s] * I[s] + Q[s] * Q[s]);
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/* filter carrier */
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iir_process(&demod->lp[2], carrier, length);
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/* normalize */
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for (s = 0; s < length; s++)
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amplitude[s] = (amplitude[s] - carrier[s]) / carrier[s] * gain;
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}
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@ -0,0 +1,26 @@
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#include "../libfilter/iir_filter.h"
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typedef struct am_mod {
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double phasestep; /* angle to rotate vector per sample */
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double phase; /* current phase */
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double gain; /* gain to be multiplied to amplitude */
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double bias; /* DC offset to add (carrier amplitude) */
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} am_mod_t;
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int am_mod_init(am_mod_t *mod, double samplerate, double offset, double gain, double bias);
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void am_mod_exit(am_mod_t *mod);
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void am_modulate_complex(am_mod_t *mod, sample_t *amplitude, int num, float *baseband);
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typedef struct am_demod {
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double phasestep; /* angle to rotate vector per sample */
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double phase; /* current rotation phase (used to shift) */
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double last_phase; /* last phase of FM (used to demodulate) */
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iir_filter_t lp[3]; /* filters received IQ signal/carrier */
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double gain; /* gain to be expected from amplitude */
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double bias; /* DC offset to be expected (carrier amplitude) */
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} am_demod_t;
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int am_demod_init(am_demod_t *demod, double samplerate, double offset, double gain, double bias);
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void am_demod_exit(am_demod_t *demod);
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void am_demodulate_complex(am_demod_t *demod, sample_t *amplitude, int length, float *baseband, sample_t *I, sample_t *Q, sample_t *carrier);
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