Merge pull request #690 from kasper93/rf_filter

NFMDemod: Add RF filter for freq deviation

CMakeLists.txt

@@ -11,6 +11,8 @@ set(CMAKE_CXX_STANDARD 11)
 set(CMAKE_CXX_STANDARD_REQUIRED ON)
 set(CMAKE_CXX_EXTENSIONS OFF)
 
+set(CMAKE_EXPORT_COMPILE_COMMANDS ON)
+
 # configure version
 set(sdrangel_VERSION_MAJOR "4")
 set(sdrangel_VERSION_MINOR "21")
@@ -207,7 +209,7 @@ elseif (WIN32)
 
   # compile with full multicore
   if(MSVC)
-    add_compile_definitions(/MP)
+    add_compile_options(/MP)
   endif()
 
   # in alternative we can use ExternalProject

plugins/channelrx/demodnfm/nfmdemodsink.cpp

@@ -35,6 +35,8 @@
 
 const double NFMDemodSink::afSqTones[] = {1000.0, 6000.0}; // {1200.0, 8000.0};
 const double NFMDemodSink::afSqTones_lowrate[] = {1000.0, 3500.0};
+const unsigned NFMDemodSink::FFT_FILTER_LENGTH = 1024;
+const unsigned NFMDemodSink::CTCSS_DETECTOR_RATE = 6000;
 
 NFMDemodSink::NFMDemodSink() :
         m_channelSampleRate(48000),
@@ -42,13 +44,15 @@ NFMDemodSink::NFMDemodSink() :
         m_audioSampleRate(48000),
         m_audioBufferFill(0),
         m_audioFifo(48000),
+        m_rfFilter(FFT_FILTER_LENGTH),
         m_ctcssIndex(0),
         m_sampleCount(0),
         m_squelchCount(0),
         m_squelchGate(4800),
-        m_filterTaps(48000 / 48 + 1),
+        m_filterTaps((48000 / 48) | 1),
         m_squelchLevel(-990),
         m_squelchOpen(false),
+        m_afSquelchOpen(false),
         m_magsq(0.0f),
         m_magsqSum(0.0f),
         m_magsqPeak(0.0f),
@@ -57,30 +61,31 @@ NFMDemodSink::NFMDemodSink() :
         m_squelchDelayLine(24000),
         m_messageQueueToGUI(nullptr)
 {
-	m_agcLevel = 1.0;
     m_audioBuffer.resize(1<<16);
-    m_phaseDiscri.setFMScaling(0.5f);
 
     applySettings(m_settings, true);
     applyChannelSettings(m_channelSampleRate, m_channelFrequencyOffset, true);
 }
 
-NFMDemodSink::~NFMDemodSink()
-{
-}
-
 void NFMDemodSink::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end)
 {
-	Complex ci;
-
     for (SampleVector::const_iterator it = begin; it != end; ++it)
     {
         Complex c(it->real(), it->imag());
         c *= m_nco.nextIQ();
 
-        if (m_interpolatorDistance < 1.0f) // interpolate
+        Complex ci;
+        fftfilt::cmplx *rf;
+        int rf_out = m_rfFilter.runFilt(c, &rf); // filter RF before demod
+        for (int i = 0 ; i < rf_out; i++)
         {
-            while (!m_interpolator.interpolate(&m_interpolatorDistanceRemain, c, &ci))
+            if (m_interpolatorDistance == 1.0f)
+            {
+                processOneSample(rf[i]);
+            }
+            else if (m_interpolatorDistance < 1.0f) // interpolate
+            {
+                while (!m_interpolator.interpolate(&m_interpolatorDistanceRemain, rf[i], &ci))
                 {
                     processOneSample(ci);
                     m_interpolatorDistanceRemain += m_interpolatorDistance;
@@ -88,13 +93,14 @@ void NFMDemodSink::feed(const SampleVector::const_iterator& begin, const SampleV
             }
             else // decimate
             {
-            if (m_interpolator.decimate(&m_interpolatorDistanceRemain, c, &ci))
+                if (m_interpolator.decimate(&m_interpolatorDistanceRemain, rf[i], &ci))
                 {
                     processOneSample(ci);
                     m_interpolatorDistanceRemain += m_interpolatorDistance;
                 }
             }
         }
+    }
 
     if (m_audioBufferFill > 0)
     {
@@ -124,12 +130,12 @@ void NFMDemodSink::processOneSample(Complex &ci)
     m_magsqCount++;
     m_sampleCount++;
 
-    bool squelchOpen = false;
+    bool squelchOpen = m_afSquelchOpen && m_settings.m_deltaSquelch;
     if (m_settings.m_deltaSquelch)
     {
         if (m_afSquelch.analyze(demod))
         {
-            squelchOpen = m_afSquelch.evaluate();
+            m_afSquelchOpen = squelchOpen = m_afSquelch.evaluate();
 
             if (!squelchOpen) {
                 m_squelchDelayLine.zeroBack(m_audioSampleRate/10); // zero out evaluation period
@@ -163,15 +169,18 @@ void NFMDemodSink::processOneSample(Complex &ci)
     if (m_squelchOpen)
     {
         if (m_settings.m_ctcssOn)
+        {
+            int factor = (m_audioSampleRate / CTCSS_DETECTOR_RATE) - 1; // decimate -> 6k
+            if ((m_sampleCount & factor) == factor)
             {
                 Real ctcssSample = m_ctcssLowpass.filter(demod);
-            int factor = (m_audioSampleRate / 6000) - 1; // decimate -> 6k
+                if (m_ctcssDetector.analyze(&ctcssSample))
-            if ((m_sampleCount & factor) == factor && m_ctcssDetector.analyze(&ctcssSample))
                 {
                     int maxToneIndex;
                     ctcssIndex = m_ctcssDetector.getDetectedTone(maxToneIndex) ?  maxToneIndex + 1 : 0;
                 }
             }
+        }
 
         if (!m_settings.m_audioMute && (!m_settings.m_ctcssOn || m_ctcssIndexSelected == ctcssIndex || m_ctcssIndexSelected == 0))
         {
@@ -227,9 +236,13 @@ void NFMDemodSink::applyChannelSettings(int channelSampleRate, int channelFreque
 
     if ((channelSampleRate != m_channelSampleRate) || force)
     {
-        m_interpolator.create(16, channelSampleRate, m_settings.m_rfBandwidth / 2.2);
+        m_interpolator.create(16, channelSampleRate, m_settings.m_fmDeviation);
-        m_interpolatorDistanceRemain = 0;
+        m_interpolatorDistance = Real(channelSampleRate) / Real(m_audioSampleRate);
-        m_interpolatorDistance =  (Real) channelSampleRate / (Real) m_audioSampleRate;
+        m_interpolatorDistanceRemain = m_interpolatorDistance;
+
+        Real lowCut = -Real(m_settings.m_fmDeviation) / channelSampleRate;
+        Real hiCut  = Real(m_settings.m_fmDeviation) / channelSampleRate;
+        m_rfFilter.create_filter(lowCut, hiCut);
     }
 
     m_channelSampleRate = channelSampleRate;
@@ -256,14 +269,16 @@ void NFMDemodSink::applySettings(const NFMDemodSettings& settings, bool force)
 
     if ((settings.m_rfBandwidth != m_settings.m_rfBandwidth) || force)
     {
-        m_interpolator.create(16, m_channelSampleRate, settings.m_rfBandwidth / 2.2);
+        m_interpolator.create(16, m_channelSampleRate, settings.m_fmDeviation);
-        m_interpolatorDistanceRemain = 0;
+        m_interpolatorDistance = Real(m_channelSampleRate) / Real(m_audioSampleRate);
-        m_interpolatorDistance =  (Real) m_channelSampleRate / (Real) m_audioSampleRate;
+        m_interpolatorDistanceRemain = m_interpolatorDistance;
     }
 
-    if ((settings.m_fmDeviation != m_settings.m_fmDeviation) || force)
+    if ((settings.m_fmDeviation != m_settings.m_fmDeviation) || force) {
-    {
+        Real lowCut = -Real(settings.m_fmDeviation) / m_channelSampleRate;
-        m_phaseDiscri.setFMScaling((0.5f *m_audioSampleRate) / static_cast<float>(settings.m_fmDeviation)); // integrate 4x factor
+        Real hiCut  = Real(settings.m_fmDeviation) / m_channelSampleRate;
+        m_rfFilter.create_filter(lowCut, hiCut);
+        m_phaseDiscri.setFMScaling(Real(m_audioSampleRate) / (2.0f * settings.m_fmDeviation));
     }
 
     if ((settings.m_afBandwidth != m_settings.m_afBandwidth) || force)
@@ -313,24 +328,25 @@ void NFMDemodSink::applyAudioSampleRate(unsigned int sampleRate)
 
     qDebug("NFMDemodSink::applyAudioSampleRate: %u m_channelSampleRate: %d", sampleRate, m_channelSampleRate);
 
-    m_filterTaps = sampleRate / 48 + 1;
+    m_filterTaps = (sampleRate / 48) | 1;
-    m_ctcssLowpass.create(m_filterTaps, sampleRate, 250.0);
+    m_ctcssLowpass.create((CTCSS_DETECTOR_RATE / 48) | 1, CTCSS_DETECTOR_RATE, 250.0);
     m_bandpass.create(m_filterTaps, sampleRate, 300.0, m_settings.m_afBandwidth);
     m_lowpass.create(m_filterTaps, sampleRate, m_settings.m_afBandwidth);
     m_squelchGate = (sampleRate / 100) * m_settings.m_squelchGate; // gate is given in 10s of ms at 48000 Hz audio sample rate
     m_squelchCount = 0; // reset squelch open counter
-    m_ctcssDetector.setCoefficients(sampleRate/16, sampleRate/8.0f); // 0.5s / 2 Hz resolution
+    m_ctcssDetector.setCoefficients(sampleRate/16, CTCSS_DETECTOR_RATE); // 0.5s / 2 Hz resolution
 
     if (sampleRate < 16000) {
         m_afSquelch.setCoefficients(sampleRate/2000, 600, sampleRate, 200, 0, afSqTones_lowrate); // 0.5ms test period, 300ms average span, audio SR, 100ms attack, no decay
     } else {
         m_afSquelch.setCoefficients(sampleRate/2000, 600, sampleRate, 200, 0, afSqTones); // 0.5ms test period, 300ms average span, audio SR, 100ms attack, no decay
     }
+    m_afSquelch.setThreshold(m_squelchLevel);
 
-    m_phaseDiscri.setFMScaling((0.5f * sampleRate) / static_cast<float>(m_settings.m_fmDeviation));
+    m_phaseDiscri.setFMScaling(Real(sampleRate) / (2.0f * m_settings.m_fmDeviation));
     m_audioFifo.setSize(sampleRate);
     m_squelchDelayLine.resize(sampleRate/2);
-    m_interpolatorDistanceRemain = 0;
     m_interpolatorDistance = Real(m_channelSampleRate) / Real(sampleRate);
+    m_interpolatorDistanceRemain = m_interpolatorDistance;
     m_audioSampleRate = sampleRate;
 }

plugins/channelrx/demodnfm/nfmdemodsink.h

@@ -24,6 +24,7 @@
 #include "dsp/phasediscri.h"
 #include "dsp/nco.h"
 #include "dsp/interpolator.h"
+#include "dsp/fftfilt.h"
 #include "dsp/firfilter.h"
 #include "dsp/afsquelch.h"
 #include "dsp/agc.h"
@@ -37,7 +38,6 @@
 class NFMDemodSink : public ChannelSampleSink {
 public:
     NFMDemodSink();
-	~NFMDemodSink();
 
     virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end);
 
@@ -105,6 +105,7 @@ private:
 
     NCO m_nco;
     Interpolator m_interpolator;
+    fftfilt m_rfFilter;
     Real m_interpolatorDistance;
     Real m_interpolatorDistanceRemain;
     Lowpass<Real> m_ctcssLowpass;
@@ -120,6 +121,7 @@ private:
 
     Real m_squelchLevel;
     bool m_squelchOpen;
+    bool m_afSquelchOpen;
     double m_magsq; //!< displayed averaged value
     double m_magsqSum;
     double m_magsqPeak;
@@ -128,7 +130,6 @@ private:
 
     MovingAverageUtil<Real, double, 32> m_movingAverage;
     AFSquelch m_afSquelch;
-	Real m_agcLevel; // AGC will aim to  this level
     DoubleBufferFIFO<Real> m_squelchDelayLine;
 
     PhaseDiscriminators m_phaseDiscri;
@@ -136,6 +137,8 @@ private:
 
     static const double afSqTones[];
     static const double afSqTones_lowrate[];
+    static const unsigned FFT_FILTER_LENGTH;
+    static const unsigned CTCSS_DETECTOR_RATE;
 
     void processOneSample(Complex &ci);
     MessageQueue *getMessageQueueToGUI() { return m_messageQueueToGUI; }

sdrbase/dsp/afsquelch.cpp

@@ -214,9 +214,9 @@ bool AFSquelch::evaluate()
 	}
 
 //    m_isOpen = ((minPower/maxPower < m_threshold) && (minIndex > maxIndex));
-
     if ((minPower/maxPower < m_threshold) && (minIndex > maxIndex)) // open condition
     {
+
         if (m_squelchCount < m_samplesAttack + m_samplesDecay)
         {
             m_squelchCount++;

sdrbase/dsp/fftfilt.cpp

@@ -75,6 +75,14 @@ void fftfilt::init_filter()
 // f1 == 0 ==> low pass filter
 // f2 == 0 ==> high pass filter
 //------------------------------------------------------------------------------
+fftfilt::fftfilt(int len)
+{
+	flen	= len;
+	pass    = 0;
+	window  = 0;
+	init_filter();
+}
+
 fftfilt::fftfilt(float f1, float f2, int len)
 {
 	flen	= len;

sdrbase/dsp/fftfilt.h

@@ -19,6 +19,7 @@ enum {NONE, BLACKMAN, HAMMING, HANNING};
 public:
 	typedef std::complex<float> cmplx;
 
+	fftfilt(int len);
 	fftfilt(float f1, float f2, int len);
 	fftfilt(float f2, int len);
 	~fftfilt();

sdrbase/dsp/firfilter.h

@@ -42,24 +42,17 @@ public:
 
         m_samples[m_ptr] = sample;
 
-        for (size_t i = 0; i < n_taps; ++i)
+        for (int i = 0; i < n_taps; ++i)
         {
-            acc += (m_samples[a++] + m_samples[b--]) * m_taps[i];
+            acc += (m_samples[a] + m_samples[b]) * m_taps[i];
 
-            if (a == n_samples) {
+            a = (a == n_samples - 1) ? 0             : a + 1;
-                a = 0;
+            b = (b == 0)             ? n_samples - 1 : b - 1;
-            }
-
-            if (b == -1) {
-                b = n_samples - 1;
-            }
         }
 
         acc += m_samples[a] * m_taps[n_taps];
 
-        if (++m_ptr == n_samples) {
+        m_ptr = (m_ptr == n_samples - 1) ? 0 : m_ptr + 1;
-            m_ptr = 0;
-        }
 
         return acc;
     }