Add libam, a library to do AM modulation and demodulation

.gitignore

@@ -44,6 +44,7 @@ src/libmncc/libmncc.a
 src/libsound/libsound.a
 src/libsdr/libsdr.a
 src/libsample/libsample.a
+src/libam/libam.a
 src/libclipper/libclipper.a
 src/anetz/libgermanton.a
 src/anetz/anetz

configure.ac

@@ -65,6 +65,7 @@ AC_OUTPUT(
     src/libscrambler/Makefile
     src/libemphasis/Makefile
     src/libfsk/Makefile
+    src/libam/Makefile
     src/libfm/Makefile
     src/libfilter/Makefile
     src/libwave/Makefile

src/Makefile.am

@@ -18,6 +18,7 @@ SUBDIRS = \
 	libscrambler \
 	libemphasis \
 	libfsk \
+	libam \
 	libfm \
 	libfilter \
 	libwave \

src/libam/Makefile.am

@@ -0,0 +1,6 @@
+AM_CPPFLAGS = -Wall -Wextra -g $(all_includes)
+
+noinst_LIBRARIES = libam.a
+
+libam_a_SOURCES = \
+	am.c

src/libam/am.c

@@ -0,0 +1,133 @@
+/* AM modulation and de-modulation
+ *
+ * (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 <stdint.h>
+#include <string.h>
+#include <math.h>
+#include "../libsample/sample.h"
+#include "am.h"
+
+#define CARRIER_FILTER 30.0
+
+/* Amplitude modulation in SDR:
+ * Just use the base band (audio signal) as real value, and 0.0 as imaginary
+ * value. The you have two side bands. Be sure to have a DC level, so you
+ * have a carrier.
+ */
+
+int am_mod_init(am_mod_t *mod, double samplerate, double offset, double gain, double bias)
+{
+	memset(mod, 0, sizeof(*mod));
+	mod->gain = gain;
+	mod->bias = bias;
+	mod->phasestep = 2.0 * M_PI * offset / samplerate;
+
+	return 0;
+}
+
+void am_mod_exit(am_mod_t __attribute__((unused)) *mod)
+{
+}
+
+void am_modulate_complex(am_mod_t *mod, sample_t *amplitude, int num, float *baseband)
+{
+	int s;
+	double vector;
+	double phasestep = mod->phasestep;
+	double phase = mod->phase;
+	double gain = mod->gain;
+	double bias = mod->bias;
+
+	for (s = 0; s < num; s++) {
+		vector = *amplitude++ * gain + bias;
+		*baseband++ = cos(phase) * vector;
+		*baseband++ = sin(phase) * vector;
+		phase += phasestep;
+		if (phase < 0.0)
+			phase += 2.0 * M_PI;
+		else if (phase >= 2.0 * M_PI)
+			phase -= 2.0 * M_PI;
+	}
+
+	mod->phase = phase;
+}
+
+/* init AM demodulator */
+int am_demod_init(am_demod_t *demod, double samplerate, double offset, double bandwidth, double gain)
+{
+	memset(demod, 0, sizeof(*demod));
+	demod->gain = gain;
+	demod->phasestep = 2 * M_PI * -offset / samplerate;
+
+	/* use fourth order (2 iter) filter, since it is as fast as second order (1 iter) filter */
+	iir_lowpass_init(&demod->lp[0], bandwidth, samplerate, 2);
+	iir_lowpass_init(&demod->lp[1], bandwidth, samplerate, 2);
+
+	/* filter carrier */
+	iir_lowpass_init(&demod->lp[2], CARRIER_FILTER, samplerate, 1);
+
+	return 0;
+}
+
+void am_demod_exit(am_demod_t __attribute__((unused)) *demod)
+{
+}
+
+/* do amplitude demodulation of baseband and write them to samples */
+void am_demodulate_complex(am_demod_t *demod, sample_t *amplitude, int length, float *baseband, sample_t *I, sample_t *Q, sample_t *carrier)
+{
+	int s, ss;
+	double phasestep = demod->phasestep;
+	double phase = demod->phase;
+	double gain = demod->gain;
+	double i, q;
+	double _sin, _cos;
+
+	/* rotate spectrum */
+	for (s = 0, ss = 0; s < length; s++) {
+		i = baseband[ss++];
+		q = baseband[ss++];
+		_sin = sin(phase);
+		_cos = cos(phase);
+		phase += phasestep;
+		if (phase < 0.0)
+			phase += 2.0 * M_PI;
+		else if (phase >= 2.0 * M_PI)
+			phase -= 2.0 * M_PI;
+		I[s] = i * _cos - q * _sin;
+		Q[s] = i * _sin + q * _cos;
+	}
+	demod->phase = phase;
+
+	/* filter bandwidth */
+	iir_process(&demod->lp[0], I, length);
+	iir_process(&demod->lp[1], Q, length);
+
+	/* demod */
+	for (s = 0; s < length; s++)
+		amplitude[s] = carrier[s] = sqrt(I[s] * I[s] + Q[s] * Q[s]);
+
+	/* filter carrier */
+	iir_process(&demod->lp[2], carrier, length);
+
+	/* normalize */
+	for (s = 0; s < length; s++)
+		amplitude[s] = (amplitude[s] - carrier[s]) / carrier[s] * gain;
+}
+

src/libam/am.h

@@ -0,0 +1,26 @@
+#include "../libfilter/iir_filter.h"
+
+typedef struct am_mod {
+	double	phasestep;	/* angle to rotate vector per sample */
+	double	phase;		/* current phase */
+	double	gain;		/* gain to be multiplied to amplitude */
+	double	bias;		/* DC offset to add (carrier amplitude) */
+} am_mod_t;
+
+int am_mod_init(am_mod_t *mod, double samplerate, double offset, double gain, double bias);
+void am_mod_exit(am_mod_t *mod);
+void am_modulate_complex(am_mod_t *mod, sample_t *amplitude, int num, float *baseband);
+
+typedef struct am_demod {
+	double	phasestep;	/* angle to rotate vector per sample */
+	double	phase;		/* current rotation phase (used to shift) */
+	double	last_phase;	/* last phase of FM (used to demodulate) */
+	iir_filter_t lp[3];	/* filters received IQ signal/carrier */
+	double	gain;		/* gain to be expected from amplitude */
+	double	bias;		/* DC offset to be expected (carrier amplitude) */
+} am_demod_t;
+
+int am_demod_init(am_demod_t *demod, double samplerate, double offset, double gain, double bias);
+void am_demod_exit(am_demod_t *demod);
+void am_demodulate_complex(am_demod_t *demod, sample_t *amplitude, int length, float *baseband, sample_t *I, sample_t *Q, sample_t *carrier);
+