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Single sideband.
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@ -48,7 +48,7 @@ public:
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void create_filter(float f1, float f2);
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void rtty_filter(float);
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int run(const cmplx& in, cmplx **out);
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int run(const cmplx& in, cmplx **out, bool usb);
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};
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#endif
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@ -36,15 +36,18 @@ SSBDemod::SSBDemod(AudioFifo* audioFifo, SampleSink* sampleSink) :
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m_interpolator.create(16, m_sampleRate, 5000);
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m_sampleDistanceRemain = (Real)m_sampleRate / 48000.0;
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m_lowpass.create(21, 48000, 5000);
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m_audioBuffer.resize(512);
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m_audioBufferFill = 0;
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m_undersampleCount = 0;
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m_usb = true;
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SSBFilter = new fftfilt(0.01, m_Bandwidth / 48000.0, ssbFftLen);
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// if (!USBFilter) segfault;
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}
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SSBDemod::~SSBDemod()
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{
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if (SSBFilter) delete SSBFilter;
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}
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void SSBDemod::configure(MessageQueue* messageQueue, Real Bandwidth, Real volume)
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@ -56,6 +59,8 @@ void SSBDemod::configure(MessageQueue* messageQueue, Real Bandwidth, Real volume
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void SSBDemod::feed(SampleVector::const_iterator begin, SampleVector::const_iterator end, bool positiveOnly)
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{
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Complex ci;
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cmplx *sideband;
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int n_out;
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bool consumed;
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for(SampleVector::const_iterator it = begin; it < end; ++it) {
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@ -63,10 +68,14 @@ void SSBDemod::feed(SampleVector::const_iterator begin, SampleVector::const_iter
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c *= m_nco.nextIQ();
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consumed = false;
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// could squelch audio if RTL_SDR samplerate is not being fully decimated.
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if(m_interpolator.interpolate(&m_sampleDistanceRemain, c, &consumed, &ci)) {
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Real demod = ci.imag() + ci.real();
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demod = 32768.0 * m_lowpass.filter(demod * 0.7);
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n_out = SSBFilter->run(ci, &sideband, m_usb);
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m_sampleDistanceRemain += (Real)m_sampleRate / 48000.0;
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} else
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n_out = 0;
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for (int i = 0; i < n_out; i++) {
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Real demod = (sideband[i].real() + sideband[i].imag()) * 0.7 * 32768.0;
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// Downsample by 4x for audio display
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if (!(m_undersampleCount++ & 3))
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@ -80,7 +89,6 @@ void SSBDemod::feed(SampleVector::const_iterator begin, SampleVector::const_iter
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uint res = m_audioFifo->write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 1);
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m_audioBufferFill = 0;
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}
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m_sampleDistanceRemain += (Real)m_sampleRate / 48000.0;
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}
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}
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if(m_audioFifo->write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 0) != m_audioBufferFill)
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@ -102,6 +110,8 @@ void SSBDemod::stop()
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bool SSBDemod::handleMessage(Message* cmd)
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{
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float band;
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if(DSPSignalNotification::match(cmd)) {
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DSPSignalNotification* signal = (DSPSignalNotification*)cmd;
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qDebug("%d samples/sec, %lld Hz offset", signal->getSampleRate(), signal->getFrequencyOffset());
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@ -113,9 +123,20 @@ bool SSBDemod::handleMessage(Message* cmd)
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return true;
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} else if(MsgConfigureSSBDemod::match(cmd)) {
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MsgConfigureSSBDemod* cfg = (MsgConfigureSSBDemod*)cmd;
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m_Bandwidth = cfg->getBandwidth();
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m_interpolator.create(16, m_sampleRate, m_Bandwidth);
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m_lowpass.create(21, 48000, cfg->getBandwidth());
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band = cfg->getBandwidth();
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if (band < 0) {
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band = -band;
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m_usb = false;
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} else
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m_usb = true;
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if (band < 500.0f)
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band = 500.0f;
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m_Bandwidth = band;
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m_interpolator.create(16, m_sampleRate, band * 2.0f);
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SSBFilter->create_filter(0.3f / 48.0f, m_Bandwidth / 48000.0f);
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m_volume = cfg->getVolume();
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m_volume *= m_volume * 0.1;
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cmd->completed();
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@ -22,10 +22,12 @@
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#include "dsp/samplesink.h"
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#include "dsp/nco.h"
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#include "dsp/interpolator.h"
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#include "dsp/lowpass.h"
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#include "dsp/fftfilt.h"
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#include "audio/audiofifo.h"
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#include "util/message.h"
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#define ssbFftLen 1024
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class AudioFifo;
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class SSBDemod : public SampleSink {
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@ -75,11 +77,12 @@ private:
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int m_undersampleCount;
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int m_sampleRate;
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int m_frequency;
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bool m_usb;
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NCO m_nco;
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Interpolator m_interpolator;
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Real m_sampleDistanceRemain;
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Lowpass<Real> m_lowpass;
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fftfilt* SSBFilter;
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AudioVector m_audioBuffer;
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uint m_audioBufferFill;
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@ -42,10 +42,10 @@
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<item row="0" column="1">
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<widget class="QSlider" name="BW">
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<property name="minimum">
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<number>1</number>
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<number>-8</number>
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</property>
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<property name="maximum">
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<number>20</number>
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<number>8</number>
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</property>
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<property name="pageStep">
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<number>1</number>
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@ -68,7 +68,7 @@ void USBDemod::feed(SampleVector::const_iterator begin, SampleVector::const_iter
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b += it->imag();
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c = Complex(a / 65536.0, b / 65536.0);
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n_out = USBFilter->run(c, &sideband);
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n_out = USBFilter->run(c, &sideband, true);
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for (int i = 0; i < n_out; i++) {
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Real demod = (sideband[i].real() + sideband[i].imag()) * 0.7 * 32768.0;
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@ -113,7 +113,7 @@ bool USBDemod::handleMessage(Message* cmd)
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} else if(MsgConfigureUSBDemod::match(cmd)) {
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MsgConfigureUSBDemod* cfg = (MsgConfigureUSBDemod*)cmd;
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m_Bandwidth = cfg->getBandwidth();
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USBFilter->create_filter(0.01, m_Bandwidth / 48000.0);
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USBFilter->create_filter(0.3 / 48.0, m_Bandwidth / 48000.0);
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m_volume = cfg->getVolume();
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m_volume *= m_volume * 0.1;
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cmd->completed();
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@ -22,7 +22,6 @@
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#include "dsp/samplesink.h"
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#include "dsp/nco.h"
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#include "dsp/interpolator.h"
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#include "dsp/lowpass.h"
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#include "dsp/fftfilt.h"
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#include "audio/audiofifo.h"
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#include "util/message.h"
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@ -144,7 +144,7 @@ void fftfilt::create_filter(float f1, float f2)
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}
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// Filter with fast convolution (overlap-add algorithm).
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int fftfilt::run(const cmplx & in, cmplx **out)
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int fftfilt::run(const cmplx & in, cmplx **out, bool usb)
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{
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timedata[inptr++] = in;
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@ -155,12 +155,19 @@ int fftfilt::run(const cmplx & in, cmplx **out)
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memcpy(freqdata, timedata, flen * sizeof(cmplx));
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fft->ComplexFFT(freqdata);
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// Discard negative frequencies for ssb
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for (int i = 0; i < flen2; i++) {
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freqdata[i] *= filter[i];
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freqdata[flen2 + i] = 0;
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}
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// Discard frequencies for ssb
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if ( usb )
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for (int i = 0; i < flen2; i++) {
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freqdata[i] *= filter[i];
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freqdata[flen2 + i] = 0;
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}
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else
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for (int i = 0; i < flen2; i++) {
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freqdata[i] = 0;
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freqdata[flen2 + i] *= filter[flen2 + i];
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}
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// in-place FFT: freqdata overwritten with filtered timedata
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fft->InverseComplexFFT(freqdata);
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// overlap and add
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