48kHz.

plugins/channel/nfm/nfmdemod.cpp

@@ -36,9 +36,9 @@ NFMDemod::NFMDemod(AudioFifo* audioFifo, SampleSink* sampleSink) :
 
 	m_nco.setFreq(m_frequency, m_sampleRate);
 	m_interpolator.create(16, m_sampleRate, 12500);
-	m_sampleDistanceRemain = (Real)m_sampleRate / 44100.0;
+	m_sampleDistanceRemain = (Real)m_sampleRate / 48000.0;
 
-	m_lowpass.create(21, 44100, 3000);
+	m_lowpass.create(21, 48000, 3000);
 
 	m_audioBuffer.resize(256);
 	m_audioBufferFill = 0;
@@ -96,7 +96,7 @@ void NFMDemod::feed(SampleVector::const_iterator begin, SampleVector::const_iter
 				}
 			}
 
-			m_sampleDistanceRemain += (Real)m_sampleRate / 44100.0;
+			m_sampleDistanceRemain += (Real)m_sampleRate / 48000.0;
 		}
 	}
 	if(m_audioFifo->write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 0) != m_audioBufferFill)
@@ -125,7 +125,7 @@ bool NFMDemod::handleMessage(Message* cmd)
 		m_sampleRate = signal->getSampleRate();
 		m_nco.setFreq(-signal->getFrequencyOffset(), m_sampleRate);
 		m_interpolator.create(16, m_sampleRate, m_rfBandwidth / 2.1);
-		m_sampleDistanceRemain = m_sampleRate / 44100.0;
+		m_sampleDistanceRemain = m_sampleRate / 48000.0;
 		m_squelchState = 0;
 		cmd->completed();
 		return true;
@@ -133,7 +133,7 @@ bool NFMDemod::handleMessage(Message* cmd)
 		MsgConfigureNFMDemod* cfg = (MsgConfigureNFMDemod*)cmd;
 		m_rfBandwidth = cfg->getRFBandwidth();
 		m_interpolator.create(16, m_sampleRate, m_rfBandwidth / 2.1);
-		m_lowpass.create(21, 44100, cfg->getAFBandwidth());
+		m_lowpass.create(21, 48000, cfg->getAFBandwidth());
 		m_squelchLevel = pow(10.0, cfg->getSquelch() / 20.0);
 		m_squelchLevel *= m_squelchLevel;
 		m_volume = cfg->getVolume();

plugins/channel/nfm/nfmdemodgui.cpp

@@ -155,7 +155,7 @@ NFMDemodGUI::NFMDemodGUI(PluginAPI* pluginAPI, QWidget* parent) :
 	connect(this, SIGNAL(widgetRolled(QWidget*,bool)), this, SLOT(onWidgetRolled(QWidget*,bool)));
 	connect(this, SIGNAL(menuDoubleClickEvent()), this, SLOT(onMenuDoubleClicked()));
 
-	m_audioFifo = new AudioFifo(4, 44100 / 4);
+	m_audioFifo = new AudioFifo(4, 48000 / 4);
 	m_spectrumVis = new SpectrumVis(ui->glSpectrum);
 	m_nfmDemod = new NFMDemod(m_audioFifo, m_spectrumVis);
 	m_channelizer = new Channelizer(m_nfmDemod);
@@ -164,7 +164,7 @@ NFMDemodGUI::NFMDemodGUI(PluginAPI* pluginAPI, QWidget* parent) :
 	m_pluginAPI->addSampleSink(m_threadedSampleSink);
 
 	ui->glSpectrum->setCenterFrequency(0);
-	ui->glSpectrum->setSampleRate(44100);
+	ui->glSpectrum->setSampleRate(48000);
 	ui->glSpectrum->setDisplayWaterfall(true);
 	ui->glSpectrum->setDisplayMaxHold(true);
 	m_spectrumVis->configure(m_threadedSampleSink->getMessageQueue(), 64, 10, FFTWindow::BlackmanHarris);
@@ -200,7 +200,7 @@ void NFMDemodGUI::applySettings()
 {
 	setTitleColor(m_channelMarker->getColor());
 	m_channelizer->configure(m_threadedSampleSink->getMessageQueue(),
-		44100,
+		48000,
 		m_channelMarker->getCenterFrequency());
 	m_nfmDemod->configure(m_threadedSampleSink->getMessageQueue(),
 		m_rfBW[ui->rfBW->value()],

sdrbase/dsp/channelizer.cpp

@@ -173,8 +173,6 @@ void Channelizer::freeFilterChain()
 	m_filterStages.clear();
 }
 
-#if 0
-
 ///////////////////////////////////////////////////////////////////////////////////
 // Copyright (C) 2012 maintech GmbH, Otto-Hahn-Str. 15, 97204 Hoechberg, Germany //
 // written by Christian Daniel                                                   //
@@ -191,138 +189,3 @@ void Channelizer::freeFilterChain()
 // You should have received a copy of the GNU General Public License             //
 // along with this program. If not, see <http://www.gnu.org/licenses/>.          //
 ///////////////////////////////////////////////////////////////////////////////////
-
-#include <QTime>
-#include <stdio.h>
-#include "channelizer.h"
-#include "hardware/audiooutput.h"
-
-Channelizer::Channelizer()
-{
-#if 0
-	m_spectrum.configure(128 , 25, FFTWindow::Bartlett);
-	m_buffer.resize(2048);
-	m_bufferFill = 0;
-	m_nco.setFreq(-100000, 500000);
-	m_interpolator.create(51, 32, 32 * 500000, 150000 / 2);
-	m_distance = 500000.0 / 176400.0;
-
-	m_interpolator2.create(19, 8, 8 * 176400, 15000 / 2);
-	m_distance2 = 4;
-
-	m_audioFifo.setSize(4, 44100 / 2 * 4);
-	m_audioOutput = new AudioOutput;
-	m_audioOutput->start(0, 44100, &m_audioFifo);
-	m_resampler = 1.0;
-
-	m_resamplerCtrl.setup(0.00001, 0, -0.00001);
-#endif
-}
-
-Channelizer::~Channelizer()
-{
-#if 0
-	m_audioOutput->stop();
-	delete m_audioOutput;
-#endif
-}
-
-#if 0
-void Channelizer::setGLSpectrum(GLSpectrum* glSpectrum)
-{
-	m_spectrum.setGLSpectrum(glSpectrum);
-}
-#endif
-
-size_t Channelizer::workUnitSize()
-{
-#if 0
-	return m_buffer.size();
-#endif
-	return 0;
-}
-
-size_t Channelizer::work(SampleVector::const_iterator begin, SampleVector::const_iterator end)
-{
-#if 0
-	int buffered = m_audioOutput->bufferedSamples();
-
-	if(m_audioFifo.fill() < (m_audioFifo.size() / 6)) {
-		while(m_audioFifo.fill() < (m_audioFifo.size() / 2)) {
-			quint32 d = 0;
-			m_audioFifo.write((quint8*)&d, 4);
-		}
-		qDebug("underflow - fill %d (vs %d)", m_audioFifo.fill(), m_audioFifo.size() / 4 / 2);
-	}
-
-	buffered = m_audioOutput->bufferedSamples();
-	int fill = m_audioFifo.fill() / 4 + buffered;
-	float err = (float)fill / ((m_audioFifo.size() / 4) / 2);
-
-	float ctrl = m_resamplerCtrl.feed(err);
-	//float resamplerRate = (ctrl / 1.0);
-	float resamplerRate = err;
-
-	if(resamplerRate < 0.9999)
-		resamplerRate = 0.9999;
-	else if(resamplerRate > 1.0001)
-		resamplerRate = 1.0001;
-	m_resampler = m_resampler * 0.99 + resamplerRate * 0.01;
-	//m_resampler = resamplerRate;
-
-	if(m_resampler < 0.995)
-		m_resampler = 0.995;
-	else if(m_resampler > 1.005)
-		m_resampler = 1.005;
-
-	//qDebug("%lld %5d %f %f %f", QDateTime::currentMSecsSinceEpoch(), fill, ctrl, m_resampler, err);
-
-	struct AudioSample {
-		qint16 l;
-		qint16 r;
-	};
-
-	size_t count = end - begin;
-	Complex ci;
-	bool consumed;
-	bool consumed2;
-
-	for(SampleVector::const_iterator it = begin; it < end; it++) {
-		Complex c(it->real() / 32768.0, it->imag() / 32768.0);
-		c *= m_nco.nextIQ();
-
-		consumed = false;
-		if(m_interpolator.interpolate(&m_distance, c, &consumed, &ci)) {
-
-			Complex d = ci * conj(m_lastSample);
-			m_lastSample = ci;
-			//Complex demod(atan2(d.imag(), d.real()) * 0.5, 0);
-			Real demod = atan2(d.imag(), d.real()) / M_PI;
-
-			consumed2 = false;
-			c = Complex(demod, 0);
-			while(!consumed2) {
-				if(m_interpolator2.interpolate(&m_distance2, c, &consumed2, &ci)) {
-					m_buffer[m_bufferFill++] = Sample(ci.real() * 32767.0, 0.0);
-
-					AudioSample s;
-					s.l = ci.real() * 32767.0;
-					s.r = s.l;
-					m_audioFifo.write((quint8*)&s, 4, 1);
-
-					if(m_bufferFill >= m_buffer.size()) {
-						m_spectrum.feed(m_buffer.begin(), m_buffer.end());
-						m_bufferFill = 0;
-					}
-					m_distance2 += 4.0 * m_resampler;
-				}
-			}
-			m_distance += 500000 / 176400.0;
-		}
-	}
-
-	return count;
-#endif
-}
-
-#endif

sdrbase/dsp/dspengine.cpp

@@ -301,7 +301,7 @@ DSPEngine::State DSPEngine::gotoRunning()
 		return gotoError("Could not start sample source");
 	m_deviceDescription = m_sampleSource->getDeviceDescription();
 
-	m_audioOutput.start(0, 44100);
+	m_audioOutput.start(0, 48000);
 	for(SampleSinks::const_iterator it = m_sampleSinks.begin(); it != m_sampleSinks.end(); it++)
 		(*it)->start();
 	m_sampleRate = 0; // make sure, report is sent