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SSB demod: use delay line to squeeze squelch tail
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@ -51,6 +51,7 @@ SSBDemod::SSBDemod(DeviceSourceAPI *deviceAPI) :
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m_agcNbSamples(12000),
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m_agcPowerThreshold(1e-2),
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m_agcThresholdGate(0),
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m_squelchDelayLine(2*48000),
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m_audioActive(false),
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m_sampleSink(0),
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m_audioFifo(24000),
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@ -207,8 +208,10 @@ void SSBDemod::feed(const SampleVector::const_iterator& begin, const SampleVecto
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m_sum.imag(0.0);
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}
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double agcVal = m_agcActive ? m_agc.feedAndGetValue(sideband[i]) : 10.0; // 10.0 for 3276.8, 1.0 for 327.68
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m_audioActive = agcVal != 0.0;
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float agcVal = m_agcActive ? m_agc.feedAndGetValue(sideband[i]) : 10.0; // 10.0 for 3276.8, 1.0 for 327.68
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fftfilt::cmplx& delayedSample = m_squelchDelayLine.readBack(m_agc.getStepDownDelay());
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m_audioActive = delayedSample.real() != 0.0;
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m_squelchDelayLine.write(sideband[i]*agcVal);
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if (m_audioMute)
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{
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@ -217,23 +220,25 @@ void SSBDemod::feed(const SampleVector::const_iterator& begin, const SampleVecto
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}
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else
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{
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fftfilt::cmplx z = delayedSample * m_agc.getStepDownValue();
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if (m_audioBinaual)
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{
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if (m_audioFlipChannels)
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{
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m_audioBuffer[m_audioBufferFill].r = (qint16)(sideband[i].imag() * m_volume * agcVal);
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m_audioBuffer[m_audioBufferFill].l = (qint16)(sideband[i].real() * m_volume * agcVal);
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m_audioBuffer[m_audioBufferFill].r = (qint16)(z.imag() * m_volume);
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m_audioBuffer[m_audioBufferFill].l = (qint16)(z.real() * m_volume);
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}
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else
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{
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m_audioBuffer[m_audioBufferFill].r = (qint16)(sideband[i].real() * m_volume * agcVal);
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m_audioBuffer[m_audioBufferFill].l = (qint16)(sideband[i].imag() * m_volume * agcVal);
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m_audioBuffer[m_audioBufferFill].r = (qint16)(z.real() * m_volume);
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m_audioBuffer[m_audioBufferFill].l = (qint16)(z.imag() * m_volume);
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}
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}
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else
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{
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Real demod = (sideband[i].real() + sideband[i].imag()) * 0.7;
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qint16 sample = (qint16)(demod * m_volume * agcVal);
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Real demod = (z.real() + z.imag()) * 0.7;
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qint16 sample = (qint16)(demod * m_volume);
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m_audioBuffer[m_audioBufferFill].l = sample;
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m_audioBuffer[m_audioBufferFill].r = sample;
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}
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@ -29,6 +29,7 @@
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#include "dsp/agc.h"
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#include "audio/audiofifo.h"
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#include "util/message.h"
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#include "util/doublebufferfifo.h"
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#include "ssbdemodsettings.h"
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@ -259,6 +260,7 @@ private:
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int m_agcNbSamples; //!< number of audio (48 kHz) samples for AGC averaging
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double m_agcPowerThreshold; //!< AGC power threshold (linear)
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int m_agcThresholdGate; //!< Gate length in number of samples befor threshold triggers
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DoubleBufferFIFO<fftfilt::cmplx> m_squelchDelayLine;
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bool m_audioActive; //!< True if an audio signal is produced (no AGC or AGC and above threshold)
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NCOF m_nco;
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@ -8,7 +8,7 @@
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const PluginDescriptor SSBPlugin::m_pluginDescriptor = {
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QString("SSB Demodulator"),
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QString("3.14.0"),
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QString("3.14.4"),
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QString("(c) Edouard Griffiths, F4EXB"),
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QString("https://github.com/f4exb/sdrangel"),
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true,
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@ -176,3 +176,15 @@ double MagAGC::feedAndGetValue(const Complex& ci)
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return m_u0;
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}
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}
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float MagAGC::getStepDownValue() const
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{
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if (m_count < m_stepDownDelay)
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{
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return 1.0f;
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}
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else
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{
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return StepFunctions::smootherstep(m_stepDownCounter * m_stepDelta);
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}
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}
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@ -50,6 +50,9 @@ public:
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void setStepDownDelay(int stepDownDelay) { m_stepDownDelay = stepDownDelay; }
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void setClamping(bool clamping) { m_clamping = clamping; }
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void setClampMax(double clampMax) { m_clampMax = clampMax; }
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int getStepDownDelay() const { return m_stepDownDelay; }
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float getStepDownValue() const;
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private:
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bool m_squared; //!< use squared magnitude (power) to compute AGC value
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double m_magsq; //!< current squared magnitude (power)
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