Single sideband.

include-gpl/dsp/fftfilt.h

@@ -48,7 +48,7 @@ public:
 	void create_filter(float f1, float f2);
 	void rtty_filter(float);
 
-	int run(const cmplx& in, cmplx **out);
+	int run(const cmplx& in, cmplx **out, bool usb);
 };
 
 #endif

plugins/channel/ssb/ssbdemod.cpp

@@ -36,15 +36,18 @@ SSBDemod::SSBDemod(AudioFifo* audioFifo, SampleSink* sampleSink) :
 	m_interpolator.create(16, m_sampleRate, 5000);
 	m_sampleDistanceRemain = (Real)m_sampleRate / 48000.0;
 
-	m_lowpass.create(21, 48000, 5000);
-
 	m_audioBuffer.resize(512);
 	m_audioBufferFill = 0;
 	m_undersampleCount = 0;
+
+	m_usb = true;
+	SSBFilter = new fftfilt(0.01, m_Bandwidth / 48000.0, ssbFftLen);
+	// if (!USBFilter) segfault;
 }
 
 SSBDemod::~SSBDemod()
 {
+	if (SSBFilter) delete SSBFilter;
 }
 
 void SSBDemod::configure(MessageQueue* messageQueue, Real Bandwidth, Real volume)
@@ -56,6 +59,8 @@ void SSBDemod::configure(MessageQueue* messageQueue, Real Bandwidth, Real volume
 void SSBDemod::feed(SampleVector::const_iterator begin, SampleVector::const_iterator end, bool positiveOnly)
 {
 	Complex ci;
+	cmplx *sideband;
+	int n_out;
 	bool consumed;
 
 	for(SampleVector::const_iterator it = begin; it < end; ++it) {
@@ -63,10 +68,14 @@ void SSBDemod::feed(SampleVector::const_iterator begin, SampleVector::const_iter
 		c *= m_nco.nextIQ();
 
 		consumed = false;
-		// could squelch audio if RTL_SDR samplerate is not being fully decimated.
 		if(m_interpolator.interpolate(&m_sampleDistanceRemain, c, &consumed, &ci)) {
-			Real demod = ci.imag() + ci.real();
-			demod = 32768.0 * m_lowpass.filter(demod * 0.7);
+			n_out = SSBFilter->run(ci, &sideband, m_usb);
+			m_sampleDistanceRemain += (Real)m_sampleRate / 48000.0;
+		} else
+			n_out = 0;
+
+		for (int i = 0; i < n_out; i++) {
+			Real demod = (sideband[i].real() + sideband[i].imag()) * 0.7 * 32768.0;
 
 			// Downsample by 4x for audio display
 			if (!(m_undersampleCount++ & 3))
@@ -80,7 +89,6 @@ void SSBDemod::feed(SampleVector::const_iterator begin, SampleVector::const_iter
 				uint res = m_audioFifo->write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 1);
 				m_audioBufferFill = 0;
 			}
-			m_sampleDistanceRemain += (Real)m_sampleRate / 48000.0;
 		}
 	}
 	if(m_audioFifo->write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 0) != m_audioBufferFill)
@@ -102,6 +110,8 @@ void SSBDemod::stop()
 
 bool SSBDemod::handleMessage(Message* cmd)
 {
+	float band;
+
 	if(DSPSignalNotification::match(cmd)) {
 		DSPSignalNotification* signal = (DSPSignalNotification*)cmd;
 		qDebug("%d samples/sec, %lld Hz offset", signal->getSampleRate(), signal->getFrequencyOffset());
@@ -113,9 +123,20 @@ bool SSBDemod::handleMessage(Message* cmd)
 		return true;
 	} else if(MsgConfigureSSBDemod::match(cmd)) {
 		MsgConfigureSSBDemod* cfg = (MsgConfigureSSBDemod*)cmd;
-		m_Bandwidth = cfg->getBandwidth();
-		m_interpolator.create(16, m_sampleRate, m_Bandwidth);
-		m_lowpass.create(21, 48000, cfg->getBandwidth());
+
+		band = cfg->getBandwidth();
+		if (band < 0) {
+			band = -band;
+			m_usb = false;
+		} else
+			m_usb = true;
+		if (band < 500.0f)
+			band = 500.0f;
+		m_Bandwidth = band;
+
+		m_interpolator.create(16, m_sampleRate, band * 2.0f);
+		SSBFilter->create_filter(0.3f / 48.0f, m_Bandwidth / 48000.0f);
+
 		m_volume = cfg->getVolume();
 		m_volume *= m_volume * 0.1;
 		cmd->completed();

plugins/channel/ssb/ssbdemod.h

@@ -22,10 +22,12 @@
 #include "dsp/samplesink.h"
 #include "dsp/nco.h"
 #include "dsp/interpolator.h"
-#include "dsp/lowpass.h"
+#include "dsp/fftfilt.h"
 #include "audio/audiofifo.h"
 #include "util/message.h"
 
+#define ssbFftLen 1024
+
 class AudioFifo;
 
 class SSBDemod : public SampleSink {
@@ -75,11 +77,12 @@ private:
 	int m_undersampleCount;
 	int m_sampleRate;
 	int m_frequency;
+	bool m_usb;
 
 	NCO m_nco;
 	Interpolator m_interpolator;
 	Real m_sampleDistanceRemain;
-	Lowpass<Real> m_lowpass;
+	fftfilt* SSBFilter;
 
 	AudioVector m_audioBuffer;
 	uint m_audioBufferFill;

plugins/channel/ssb/ssbdemodgui.ui

@@ -42,10 +42,10 @@
     <item row="0" column="1">
      <widget class="QSlider" name="BW">
       <property name="minimum">
-       <number>1</number>
+       <number>-8</number>
       </property>
       <property name="maximum">
-       <number>20</number>
+       <number>8</number>
       </property>
       <property name="pageStep">
        <number>1</number>

plugins/channel/usb/usbdemod.cpp

@@ -68,7 +68,7 @@ void USBDemod::feed(SampleVector::const_iterator begin, SampleVector::const_iter
 		b += it->imag();
 		c = Complex(a / 65536.0, b / 65536.0);
 
-		n_out = USBFilter->run(c, &sideband);
+		n_out = USBFilter->run(c, &sideband, true);
 		for (int i = 0; i < n_out; i++) {
 			Real demod = (sideband[i].real() + sideband[i].imag()) * 0.7 * 32768.0;
 
@@ -113,7 +113,7 @@ bool USBDemod::handleMessage(Message* cmd)
 	} else if(MsgConfigureUSBDemod::match(cmd)) {
 		MsgConfigureUSBDemod* cfg = (MsgConfigureUSBDemod*)cmd;
 		m_Bandwidth = cfg->getBandwidth();
-		USBFilter->create_filter(0.01, m_Bandwidth / 48000.0);
+		USBFilter->create_filter(0.3 / 48.0, m_Bandwidth / 48000.0);
 		m_volume = cfg->getVolume();
 		m_volume *= m_volume * 0.1;
 		cmd->completed();

plugins/channel/usb/usbdemod.h

@@ -22,7 +22,6 @@
 #include "dsp/samplesink.h"
 #include "dsp/nco.h"
 #include "dsp/interpolator.h"
-#include "dsp/lowpass.h"
 #include "dsp/fftfilt.h"
 #include "audio/audiofifo.h"
 #include "util/message.h"

sdrbase/dsp/fftfilt.cxx

@@ -144,7 +144,7 @@ void fftfilt::create_filter(float f1, float f2)
 }
 
 // Filter with fast convolution (overlap-add algorithm).
-int fftfilt::run(const cmplx & in, cmplx **out)
+int fftfilt::run(const cmplx & in, cmplx **out, bool usb)
 {
 	timedata[inptr++] = in;
 
@@ -155,12 +155,19 @@ int fftfilt::run(const cmplx & in, cmplx **out)
 	memcpy(freqdata, timedata, flen * sizeof(cmplx));
 	fft->ComplexFFT(freqdata);
 
-	// Discard negative frequencies for ssb
-	for (int i = 0; i < flen2; i++) {
-		freqdata[i] *= filter[i];
-		freqdata[flen2 + i] = 0;
-	}
+	// Discard frequencies for ssb
+	if ( usb )
+		for (int i = 0; i < flen2; i++) {
+			freqdata[i] *= filter[i];
+			freqdata[flen2 + i] = 0;
+		}
+	else
+		for (int i = 0; i < flen2; i++) {
+			freqdata[i] = 0;
+			freqdata[flen2 + i] *= filter[flen2 + i];
+		}
 
+	// in-place FFT: freqdata overwritten with filtered timedata
 	fft->InverseComplexFFT(freqdata);
 
 	// overlap and add