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NFMDemod: Add RF filter for freq deviation

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- Run CTCSS filter with reduced rate to much detection
- Convert tabs to spaces to be consistent in the file
- Fix AF squelch threshold setting after changing SR
This commit is contained in:
Kacper Michajłow 2020-11-05 18:55:39 +01:00
parent eea8828943
commit 786640ee1f
6 changed files with 124 additions and 103 deletions
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122
plugins/channelrx/demodnfm/nfmdemodsink.cpp
View File

@ -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),
@ -57,55 +61,57 @@ 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);
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();
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(ci);
m_interpolatorDistanceRemain += m_interpolatorDistance;
processOneSample(rf[i]);
}
}
else // decimate
{
if (m_interpolator.decimate(&m_interpolatorDistanceRemain, c, &ci))
else if (m_interpolatorDistance < 1.0f) // interpolate
{
processOneSample(ci);
m_interpolatorDistanceRemain += m_interpolatorDistance;
while (!m_interpolator.interpolate(&m_interpolatorDistanceRemain, rf[i], &ci))
{
processOneSample(ci);
m_interpolatorDistanceRemain += m_interpolatorDistance;
}
}
else // decimate
{
if (m_interpolator.decimate(&m_interpolatorDistanceRemain, rf[i], &ci))
{
processOneSample(ci);
m_interpolatorDistanceRemain += m_interpolatorDistance;
}
}
}
}
if (m_audioBufferFill > 0)
{
uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill);
if (m_audioBufferFill > 0)
{
uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill);
if (res != m_audioBufferFill) {
qDebug("NFMDemodSink::feed: %u/%u tail samples written", res, m_audioBufferFill);
}
if (res != m_audioBufferFill) {
qDebug("NFMDemodSink::feed: %u/%u tail samples written", res, m_audioBufferFill);
}
m_audioBufferFill = 0;
}
m_audioBufferFill = 0;
}
}
void NFMDemodSink::processOneSample(Complex &ci)
@ -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
@ -164,12 +170,15 @@ void NFMDemodSink::processOneSample(Complex &ci)
{
if (m_settings.m_ctcssOn)
{
Real ctcssSample = m_ctcssLowpass.filter(demod);
int factor = (m_audioSampleRate / 6000) - 1; // decimate -> 6k
if ((m_sampleCount & factor) == factor && m_ctcssDetector.analyze(&ctcssSample))
int factor = (m_audioSampleRate / CTCSS_DETECTOR_RATE) - 1; // decimate -> 6k
if ((m_sampleCount & factor) == factor)
{
int maxToneIndex;
ctcssIndex = m_ctcssDetector.getDetectedTone(maxToneIndex) ? maxToneIndex + 1 : 0;
Real ctcssSample = m_ctcssLowpass.filter(demod);
if (m_ctcssDetector.analyze(&ctcssSample))
{
int maxToneIndex;
ctcssIndex = m_ctcssDetector.getDetectedTone(maxToneIndex) ? maxToneIndex + 1 : 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_interpolatorDistanceRemain = 0;
m_interpolatorDistance = (Real) channelSampleRate / (Real) m_audioSampleRate;
m_interpolator.create(16, channelSampleRate, m_settings.m_fmDeviation);
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_interpolatorDistanceRemain = 0;
m_interpolatorDistance = (Real) m_channelSampleRate / (Real) m_audioSampleRate;
m_interpolator.create(16, m_channelSampleRate, settings.m_fmDeviation);
m_interpolatorDistance = Real(m_channelSampleRate) / Real(m_audioSampleRate);
m_interpolatorDistanceRemain = m_interpolatorDistance;
}
if ((settings.m_fmDeviation != m_settings.m_fmDeviation) || force)
{
m_phaseDiscri.setFMScaling((0.5f *m_audioSampleRate) / static_cast<float>(settings.m_fmDeviation)); // integrate 4x factor
if ((settings.m_fmDeviation != m_settings.m_fmDeviation) || force) {
Real lowCut = -Real(settings.m_fmDeviation) / m_channelSampleRate;
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_ctcssLowpass.create(m_filterTaps, sampleRate, 250.0);
m_filterTaps = (sampleRate / 48) | 1;
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;
}

77
plugins/channelrx/demodnfm/nfmdemodsink.h
View File

@ -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"
@ -37,20 +38,19 @@
class NFMDemodSink : public ChannelSampleSink {
public:
NFMDemodSink();
~NFMDemodSink();
virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end);
virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end);
const Real *getCtcssToneSet(int& nbTones) const {
nbTones = m_ctcssDetector.getNTones();
return m_ctcssDetector.getToneSet();
}
const Real *getCtcssToneSet(int& nbTones) const {
nbTones = m_ctcssDetector.getNTones();
return m_ctcssDetector.getToneSet();
}
void setSelectedCtcssIndex(int selectedCtcssIndex) {
m_ctcssIndexSelected = selectedCtcssIndex;
}
void setSelectedCtcssIndex(int selectedCtcssIndex) {
m_ctcssIndexSelected = selectedCtcssIndex;
}
bool getSquelchOpen() const { return m_squelchOpen; }
bool getSquelchOpen() const { return m_squelchOpen; }
void getMagSqLevels(double& avg, double& peak, int& nbSamples)
{
@ -89,53 +89,56 @@ private:
double m_magsqPeak;
};
enum RateState {
RSInitialFill,
RSRunning
};
enum RateState {
RSInitialFill,
RSRunning
};
int m_channelSampleRate;
int m_channelFrequencyOffset;
NFMDemodSettings m_settings;
NFMDemodSettings m_settings;
int m_audioSampleRate;
AudioVector m_audioBuffer;
uint m_audioBufferFill;
AudioFifo m_audioFifo;
NCO m_nco;
Interpolator m_interpolator;
Real m_interpolatorDistance;
Real m_interpolatorDistanceRemain;
Lowpass<Real> m_ctcssLowpass;
Bandpass<Real> m_bandpass;
NCO m_nco;
Interpolator m_interpolator;
fftfilt m_rfFilter;
Real m_interpolatorDistance;
Real m_interpolatorDistanceRemain;
Lowpass<Real> m_ctcssLowpass;
Bandpass<Real> m_bandpass;
Lowpass<Real> m_lowpass;
CTCSSDetector m_ctcssDetector;
int m_ctcssIndex; // 0 for nothing detected
int m_ctcssIndexSelected;
int m_sampleCount;
int m_squelchCount;
int m_squelchGate;
int m_filterTaps;
CTCSSDetector m_ctcssDetector;
int m_ctcssIndex; // 0 for nothing detected
int m_ctcssIndexSelected;
int m_sampleCount;
int m_squelchCount;
int m_squelchGate;
int m_filterTaps;
Real m_squelchLevel;
bool m_squelchOpen;
double m_magsq; //!< displayed averaged value
double m_magsqSum;
double m_magsqPeak;
Real m_squelchLevel;
bool m_squelchOpen;
bool m_afSquelchOpen;
double m_magsq; //!< displayed averaged value
double m_magsqSum;
double m_magsqPeak;
int m_magsqCount;
MagSqLevelsStore m_magSqLevelStore;
MovingAverageUtil<Real, double, 32> m_movingAverage;
AFSquelch m_afSquelch;
Real m_agcLevel; // AGC will aim to this level
DoubleBufferFIFO<Real> m_squelchDelayLine;
MovingAverageUtil<Real, double, 32> m_movingAverage;
AFSquelch m_afSquelch;
DoubleBufferFIFO<Real> m_squelchDelayLine;
PhaseDiscriminators m_phaseDiscri;
MessageQueue *m_messageQueueToGUI;
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; }

2
sdrbase/dsp/afsquelch.cpp
View File

@ -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++;

8
sdrbase/dsp/fftfilt.cpp
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@ -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;

1
sdrbase/dsp/fftfilt.h
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@ -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();

17
sdrbase/dsp/firfilter.h
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@ -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 = 0;
}
if (b == -1) {
b = n_samples - 1;
}
a = (a == n_samples - 1) ? 0 : a + 1;
b = (b == 0) ? n_samples - 1 : b - 1;
}
acc += m_samples[a] * m_taps[n_taps];
if (++m_ptr == n_samples) {
m_ptr = 0;
}
m_ptr = (m_ptr == n_samples - 1) ? 0 : m_ptr + 1;
return acc;
}