Rebased code from new eried repo commits. Changed to to reflect strijar implementation. Fixed previous issue with old ssb-am-tx ui_mictx code.

firmware/baseband/dsp_modulate.cpp

@@ -0,0 +1,137 @@
+/*
+ * Copyright (C) 2020 Belousov Oleg
+ *
+ * This file is part of PortaPack.
+ *
+ * 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 2, 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; see the file COPYING.  If not, write to
+ * the Free Software Foundation, Inc., 51 Franklin Street,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include "dsp_modulate.hpp"
+#include "sine_table_int8.hpp"
+
+namespace dsp {
+namespace modulate {
+
+Modulator::~Modulator() {
+}
+
+Mode Modulator::get_mode() {
+	return mode;
+}
+
+void Modulator::set_mode(Mode new_mode) {
+	mode = new_mode;
+}
+
+void Modulator::set_over(uint32_t new_over) {
+    over = new_over;
+}
+
+///
+
+SSB::SSB() : hilbert() {
+	mode = Mode::LSB;
+}
+
+void SSB::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer) {
+	int32_t		sample = 0;
+	int8_t		re = 0, im = 0;
+	
+	for (size_t counter = 0; counter < buffer.count; counter++) {
+		if (counter % 128 == 0) {
+			float	i = 0.0, q = 0.0;
+
+		    sample = audio.p[counter / over] >> 2;
+			//switch (mode) {
+				//case Mode::LSB:	
+			hilbert.execute(sample / 32768.0f, i, q);
+				//case Mode::USB:	hilbert.execute(sample / 32768.0f, q, i);
+				//default:	break;
+			//}
+
+			i *= 64.0f;
+			q *= 64.0f;
+			switch (mode) {
+				case Mode::LSB:	re = q;	im = i;	break;
+				case Mode::USB:	re = i;	im = q;	break;
+				default:	re = 0; im = 0;	break;
+			}
+			//re = q;
+			//im = i;
+			//break;
+		}
+		
+		buffer.p[counter] = { re, im };
+	}
+}
+
+///
+
+FM::FM() {
+	mode = Mode::FM;
+}
+
+void FM::set_fm_delta(uint32_t new_delta) {
+	fm_delta = new_delta;
+}
+
+void FM::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer) {
+	int32_t		sample = 0;
+	int8_t		re, im;
+
+	for (size_t counter = 0; counter < buffer.count; counter++) {
+		if (counter % over == 0) {
+		    sample = audio.p[counter / over] >> 8;
+		    delta = sample * fm_delta;
+		}
+
+		phase += delta;
+		sphase = phase >> 24;
+
+		re = (sine_table_i8[(sphase + 64) & 255]);
+		im = (sine_table_i8[sphase]);
+		
+		buffer.p[counter] = { re, im };
+	}
+}
+
+AM::AM() {
+	mode = Mode::AM;
+}
+
+void AM::execute(const buffer_s16_t& audio, const buffer_c8_t& buffer) {
+	int32_t         sample = 0;
+	int8_t          re, im;
+	float		q = 0.0;
+	
+	for (size_t counter = 0; counter < buffer.count; counter++) {
+		if (counter % 128 == 0) {
+			sample = audio.p[counter / over] >> 2;
+		}
+
+		q = sample / 32768.0f;
+		q *= 64.0f;
+		switch (mode) {
+			case Mode::AM:	re = q + 20; im = q + 20;
+			case Mode::DSB:	re = q; im = q;
+			default:	break;
+		}
+		buffer.p[counter] = { re, im };
+	}
+}
+
+}
+}