Changed NFM RF threshold squelch for after demod squelch

CMakeLists.txt

@@ -49,6 +49,7 @@ set(sdrbase_SOURCES
 	sdrbase/audio/audiofifo.cpp
 	sdrbase/audio/audiooutput.cpp
 
+	sdrbase/dsp/afsquelch.cpp
 	sdrbase/dsp/channelizer.cpp
 	sdrbase/dsp/channelmarker.cpp
 	sdrbase/dsp/ctcssdetector.cpp

include-gpl/dsp/afsquelch.h

@@ -0,0 +1,87 @@
+///////////////////////////////////////////////////////////////////////////////////
+// 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/>.          //
+///////////////////////////////////////////////////////////////////////////////////
+
+#ifndef INCLUDE_GPL_DSP_AFSQUELCH_H_
+#define INCLUDE_GPL_DSP_AFSQUELCH_H_
+
+#include "dsp/dsptypes.h"
+
+/** AFSquelch: AF squelch class based on the Modified Goertzel
+ * algorithm.
+ */
+class AFSquelch {
+public:
+    // Constructors and Destructor
+	AFSquelch();
+    // allows user defined tone pair
+	AFSquelch(unsigned int nbTones, const Real *tones);
+    virtual ~AFSquelch();
+
+    // setup the basic parameters and coefficients
+    void setCoefficients(
+    		int N,              // the algorithm "block"  size
+			int SampleRate,     // input signal sample rate
+			int _samplesAttack, // number of results before squelch opens
+			int _samplesDecay); // number of results keeping squelch open
+
+    // set the detection threshold
+    void setThreshold(double _threshold) {
+    	threshold = _threshold;
+    }
+
+    // analyze a sample set and optionally filter
+    // the tone frequencies.
+    bool analyze(Real *sample); // input signal sample
+
+    // get the tone set
+    const Real *getToneSet() const
+    {
+    	return toneSet;
+    }
+
+    bool open() const {
+    	return isOpen;
+    }
+
+    void reset();                       // reset the analysis algorithm
+
+protected:
+    void feedback(Real sample);
+    void feedForward();
+    void evaluate();
+
+private:
+    int N;
+    int sampleRate;
+    int samplesProcessed;
+    int maxPowerIndex;
+    int nTones;
+    int samplesAttack;
+    int attackCount;
+    int samplesDecay;
+    int decayCount;
+    bool isOpen;
+    double threshold;
+    double *k;
+    double *coef;
+    Real *toneSet;
+    double *u0;
+    double *u1;
+    double *power;
+};
+
+
+#endif /* INCLUDE_GPL_DSP_CTCSSDETECTOR_H_ */

plugins/channel/nfm/nfmdemod.cpp

@@ -26,11 +26,15 @@
 
 #include <iostream>
 
+static const Real afSqTones[2] = {2000.0, 8000.0};
+
 MESSAGE_CLASS_DEFINITION(NFMDemod::MsgConfigureNFMDemod, Message)
 
 NFMDemod::NFMDemod(AudioFifo* audioFifo, SampleSink* sampleSink) :
 	m_ctcssIndex(0),
 	m_sampleCount(0),
+	m_afSquelch(2, afSqTones),
+	m_squelchOpen(false),
 	m_sampleSink(sampleSink),
 	m_audioFifo(audioFifo)
 {
@@ -38,7 +42,7 @@ NFMDemod::NFMDemod(AudioFifo* audioFifo, SampleSink* sampleSink) :
 	m_config.m_inputFrequencyOffset = 0;
 	m_config.m_rfBandwidth = 12500;
 	m_config.m_afBandwidth = 3000;
-	m_config.m_squelch = -40.0;
+	m_config.m_squelch = -30.0;
 	m_config.m_volume = 2.0;
 	m_config.m_audioSampleRate = 48000;
 
@@ -52,6 +56,8 @@ NFMDemod::NFMDemod(AudioFifo* audioFifo, SampleSink* sampleSink) :
 	m_AGC.resize(4096, m_agcLevel, 0, 0.1*m_agcLevel);
 
 	m_ctcssDetector.setCoefficients(3000, 6000.0); // 0.5s / 2 Hz resolution
+	m_afSquelch.setCoefficients(24, 48000.0, 1, 1); // 4000 Hz span, 250us
+	m_afSquelch.setThreshold(0.001);
 }
 
 NFMDemod::~NFMDemod()
@@ -109,56 +115,52 @@ void NFMDemod::feed(SampleVector::const_iterator begin, SampleVector::const_iter
 			if(m_interpolator.interpolate(&m_interpolatorDistanceRemain, c, &ci)) {
 				m_sampleBuffer.push_back(Sample(ci.real() * 32767.0, ci.imag() * 32767.0));
 
-				m_movingAverage.feed(ci.real() * ci.real() + ci.imag() * ci.imag());
-				if(m_movingAverage.average() >= m_squelchLevel)
-					m_squelchState = m_running.m_audioSampleRate/ 20;
-
 				qint16 sample;
-				if(m_squelchState > 0)
+
+				m_AGC.feed(abs(ci));
+				ci *= (m_agcLevel / m_AGC.getValue());
+
+				// demod
+				/*
+				Real argument = arg(ci);
+				Real demod = argument - m_lastArgument;
+				m_lastArgument = argument;
+				*/
+
+				/*
+				// Original NFM
+				Complex d = conj(m_m1Sample) * ci;
+				Real demod = atan2(d.imag(), d.real());
+				demod /= M_PI;
+				*/
+
+				/*
+				Real argument1 = arg(ci);//atan2(ci.imag(), ci.real());
+				Real argument2 = m_lastSample.real();
+				Real demod = angleDist(argument2, argument1);
+				m_lastSample = Complex(argument1, 0);
+				*/
+
+				// Alternative without atan - needs AGC
+				// http://www.embedded.com/design/configurable-systems/4212086/DSP-Tricks--Frequency-demodulation-algorithms-
+				Real ip = ci.real() - m_m2Sample.real();
+				Real qp = ci.imag() - m_m2Sample.imag();
+				Real h1 = m_m1Sample.real() * qp;
+				Real h2 = m_m1Sample.imag() * ip;
+				Real demod = (h1 - h2) * 10000;
+
+				m_m2Sample = m_m1Sample;
+				m_m1Sample = ci;
+				m_sampleCount++;
+
+				// AF processing
+
+				if(m_afSquelch.analyze(&demod)) {
+					m_squelchOpen = m_afSquelch.open();
+				}
+
+				if (m_squelchOpen)
 				{
-					m_squelchState--;
-
-					m_AGC.feed(abs(ci));
-					ci *= (m_agcLevel / m_AGC.getValue());
-
-					// demod
-					/*
-					Real argument = arg(ci);
-					Real demod = argument - m_lastArgument;
-					m_lastArgument = argument;
-					*/
-
-					/*
-					// Original NFM
-					Complex d = conj(m_m1Sample) * ci;
-					Real demod = atan2(d.imag(), d.real());
-					demod /= M_PI;
-					*/
-
-					/*
-					Real argument1 = arg(ci);//atan2(ci.imag(), ci.real());
-					Real argument2 = m_lastSample.real();
-					Real demod = angleDist(argument2, argument1);
-					m_lastSample = Complex(argument1, 0);
-					*/
-
-					// Alternative without atan - needs AGC
-					// http://www.embedded.com/design/configurable-systems/4212086/DSP-Tricks--Frequency-demodulation-algorithms-
-					Real ip = ci.real() - m_m2Sample.real();
-					Real qp = ci.imag() - m_m2Sample.imag();
-					Real h1 = m_m1Sample.real() * qp;
-					Real h2 = m_m1Sample.imag() * ip;
-					Real demod = (h1 - h2) * 10000;
-
-					m_m2Sample = m_m1Sample;
-					m_m1Sample = ci;
-					m_sampleCount++;
-
-					// AF processing
-
-					//demod = m_lowpass.filter(demod);
-					//sample = demod * 32700;
-
 					Real ctcss_sample = m_lowpass.filter(demod);
 
 					if ((m_sampleCount & 7) == 7) // decimate 48k -> 6k
@@ -186,7 +188,7 @@ void NFMDemod::feed(SampleVector::const_iterator begin, SampleVector::const_iter
 
 					if (m_ctcssIndexSelected && (m_ctcssIndexSelected != m_ctcssIndex))
 					{
-						sample = 0.0;
+						sample = 0;
 					}
 					else
 					{
@@ -234,7 +236,6 @@ void NFMDemod::feed(SampleVector::const_iterator begin, SampleVector::const_iter
 
 void NFMDemod::start()
 {
-	m_squelchState = 0;
 	m_audioFifo->clear();
 	m_interpolatorRegulation = 0.9999;
 	m_interpolatorDistance = 1.0;
@@ -292,8 +293,10 @@ void NFMDemod::apply()
 	}
 
 	if(m_config.m_squelch != m_running.m_squelch) {
-		m_squelchLevel = pow(10.0, m_config.m_squelch / 20.0);
+		m_squelchLevel = pow(10.0, m_config.m_squelch / 10.0);
 		m_squelchLevel *= m_squelchLevel;
+		m_afSquelch.setThreshold(m_squelchLevel);
+		m_afSquelch.reset();
 	}
 
 	m_running.m_inputSampleRate = m_config.m_inputSampleRate;

plugins/channel/nfm/nfmdemod.h

@@ -27,6 +27,7 @@
 #include "dsp/movingaverage.h"
 #include "dsp/agc.h"
 #include "dsp/ctcssdetector.h"
+#include "dsp/afsquelch.h"
 #include "audio/audiofifo.h"
 #include "util/message.h"
 
@@ -136,8 +137,10 @@ private:
 	int m_ctcssIndexSelected;
 	int m_sampleCount;
 
-	Real m_squelchLevel;
+	double m_squelchLevel;
-	int m_squelchState;
+	//int m_squelchState;
+	AFSquelch m_afSquelch;
+	bool m_squelchOpen;
 
 	Real m_lastArgument;
 	Complex m_m1Sample;

plugins/channel/nfm/nfmdemodgui.ui

@@ -253,10 +253,17 @@
      </widget>
     </item>
     <item row="5" column="1">
-     <widget class="QComboBox" name="ctcss"/>
+     <widget class="QComboBox" name="ctcss">
+      <property name="toolTip">
+       <string>Set CTCSS</string>
+      </property>
+     </widget>
     </item>
     <item row="5" column="2">
      <widget class="QLabel" name="ctcssText">
+      <property name="toolTip">
+       <string>CTCSS detected</string>
+      </property>
       <property name="text">
        <string>--</string>
       </property>

sdrbase/dsp/afsquelch.cpp

@@ -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;
+		}
+	}
+}