Changed NFM RF threshold squelch for after demod squelch

2026-06-18 05:38:47 -04:00 · 2015-06-19 08:27:29 +02:00
parent 70cce80995
commit e66d9a417f
6 changed files with 370 additions and 55 deletions
@@ -0,0 +1,214 @@
+///////////////////////////////////////////////////////////////////////////////////
+// Copyright (C) 2015 Edouard Griffiths, F4EXB.                                  //
+//                                                                               //
+// 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 as version 3 of the License, or                  //
+//                                                                               //
+// 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 V3 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 <cmath>
+#include "dsp/afsquelch.h"
+
+AFSquelch::AFSquelch() :
+			N(0),
+			sampleRate(0),
+			samplesProcessed(0),
+			maxPowerIndex(0),
+			nTones(2),
+			samplesAttack(0),
+			attackCount(0),
+			samplesDecay(0),
+			decayCount(0),
+			isOpen(false),
+			threshold(0.0)
+{
+	k = new double[nTones];
+	coef = new double[nTones];
+	toneSet = new Real[nTones];
+	u0 = new double[nTones];
+	u1 = new double[nTones];
+	power = new double[nTones];
+
+	toneSet[0]  = 2000.0;
+	toneSet[1]  = 10000.0;
+}
+
+AFSquelch::AFSquelch(unsigned int nbTones, const Real *tones) :
+			N(0),
+			sampleRate(0),
+			samplesProcessed(0),
+			maxPowerIndex(0),
+			nTones(nbTones),
+			samplesAttack(0),
+			attackCount(0),
+			samplesDecay(0),
+			decayCount(0),
+			isOpen(false),
+			threshold(0.0)
+{
+	k = new double[nTones];
+	coef = new double[nTones];
+	toneSet = new Real[nTones];
+	u0 = new double[nTones];
+	u1 = new double[nTones];
+	power = new double[nTones];
+
+	for (int j = 0; j < nTones; ++j)
+	{
+		toneSet[j] = tones[j];
+	}
+}
+
+
+AFSquelch::~AFSquelch()
+{
+	delete[] k;
+	delete[] coef;
+	delete[] toneSet;
+	delete[] u0;
+	delete[] u1;
+	delete[] power;
+}
+
+
+void AFSquelch::setCoefficients(int _N, int _samplerate, int _samplesAttack, int _samplesDecay )
+{
+	N = _N;                   // save the basic parameters for use during analysis
+	sampleRate = _samplerate;
+	samplesAttack = _samplesAttack;
+	samplesDecay = _samplesDecay;
+
+	// for each of the frequencies (tones) of interest calculate
+	// k and the associated filter coefficient as per the Goertzel
+	// algorithm. Note: we are using a real value (as apposed to
+	// an integer as described in some references. k is retained
+	// for later display. The tone set is specified in the
+	// constructor. Notice that the resulting coefficients are
+	// independent of N.
+	for (int j = 0; j < nTones; ++j)
+	{
+		k[j] = ((double)N * toneSet[j]) / (double)sampleRate;
+		coef[j] = 2.0 * cos((2.0 * M_PI * toneSet[j])/(double)sampleRate);
+	}
+}
+
+
+// Analyze an input signal
+bool AFSquelch::analyze(Real *sample)
+{
+
+	feedback(*sample); // Goertzel feedback
+	samplesProcessed += 1;
+
+	if (samplesProcessed == N) // completed a block of N
+	{
+		feedForward(); // calculate the power at each tone
+		samplesProcessed = 0;
+		return true; // have a result
+	}
+	else
+	{
+		return false;
+	}
+}
+
+
+void AFSquelch::feedback(Real in)
+{
+	double t;
+
+	// feedback for each tone
+	for (int j = 0; j < nTones; ++j)
+	{
+		t = u0[j];
+		u0[j] = in + (coef[j] * u0[j]) - u1[j];
+		u1[j] = t;
+	}
+}
+
+
+void AFSquelch::feedForward()
+{
+	for (int j = 0; j < nTones; ++j)
+	{
+		power[j] = (u0[j] * u0[j]) + (u1[j] * u1[j]) - (coef[j] * u0[j] * u1[j]);
+		u0[j] = u1[j] = 0.0; // reset for next block.
+	}
+
+	evaluate();
+}
+
+
+void AFSquelch::reset()
+{
+	for (int j = 0; j < nTones; ++j)
+	{
+		power[j] = u0[j] = u1[j] = 0.0; // reset
+	}
+
+	samplesProcessed = 0;
+	maxPowerIndex = 0;
+	isOpen = false;
+}
+
+
+void AFSquelch::evaluate()
+{
+	double maxPower = 0.0;
+	double minPower;
+	int minIndex = 0, maxIndex = 0;
+
+	for (int j = 0; j < nTones; ++j)
+	{
+		if (power[j] > maxPower) {
+			maxPower = power[j];
+			maxIndex = j;
+		}
+	}
+
+	minPower = maxPower;
+
+	for (int j = 0; j < nTones; ++j)
+	{
+		if (power[j] < minPower) {
+			minPower = power[j];
+			minIndex = j;
+		}
+	}
+
+	// principle is to open if power is uneven because noise gives even power
+	bool open = ((maxPower - minPower) > threshold) && (minIndex > maxIndex);
+
+	if (open)
+	{
+		if (attackCount < samplesAttack)
+		{
+			attackCount++;
+		}
+		else
+		{
+			isOpen = true;
+			decayCount = 0;
+		}
+	}
+	else
+	{
+		if (decayCount < samplesDecay)
+		{
+			decayCount++;
+		}
+		else
+		{
+			isOpen = false;
+			attackCount = 0;
+		}
+	}
+}