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			475 lines
		
	
	
		
			15 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
			
		
		
	
	
			475 lines
		
	
	
		
			15 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
| // Copyright (C) 2019 Edouard Griffiths, F4EXB                                   //
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| //                                                                               //
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| // This program is free software; you can redistribute it and/or modify          //
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| // it under the terms of the GNU General Public License as published by          //
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| // the Free Software Foundation as version 3 of the License, or                  //
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| // (at your option) any later version.                                           //
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| //                                                                               //
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| // This program is distributed in the hope that it will be useful,               //
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| // but WITHOUT ANY WARRANTY; without even the implied warranty of                //
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| // MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the                  //
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| // GNU General Public License V3 for more details.                               //
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| //                                                                               //
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| // You should have received a copy of the GNU General Public License             //
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| // along with this program. If not, see <http://www.gnu.org/licenses/>.          //
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| ///////////////////////////////////////////////////////////////////////////////////
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| 
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| #include <QDebug>
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| 
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| #include "dsp/datafifo.h"
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| #include "util/messagequeue.h"
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| #include "maincore.h"
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| 
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| #include "wfmmodsource.h"
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| 
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| const int WFMModSource::m_rfFilterFFTLength = 1024;
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| const int WFMModSource::m_levelNbSamples = 480; // every 10ms
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| 
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| WFMModSource::WFMModSource() :
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|     m_channelSampleRate(384000),
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|     m_channelFrequencyOffset(0),
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|     m_modPhasor(0.0f),
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|     m_audioSampleRate(48000),
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|     m_audioFifo(12000),
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|     m_feedbackAudioSampleRate(48000),
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|     m_feedbackAudioFifo(48000),
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| 	m_levelCalcCount(0),
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| 	m_peakLevel(0.0f),
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| 	m_levelSum(0.0f),
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|     m_ifstream(nullptr),
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|     m_mutex(QMutex::Recursive)
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| {
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|     m_rfFilter = new fftfilt(-62500.0 / 384000.0, 62500.0 / 384000.0, m_rfFilterFFTLength);
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|     m_rfFilterBuffer = new Complex[m_rfFilterFFTLength];
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|     std::fill(m_rfFilterBuffer, m_rfFilterBuffer+m_rfFilterFFTLength, Complex{0,0});
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|     m_rfFilterBufferIndex = 0;
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| 	m_audioBuffer.resize(24000);
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| 	m_audioBufferFill = 0;
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| 	m_audioReadBuffer.resize(24000);
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| 	m_audioReadBufferFill = 0;
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| 	m_magsq = 0.0;
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| 	m_feedbackAudioBuffer.resize(1<<14);
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| 	m_feedbackAudioBufferFill = 0;
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|     m_demodBuffer.resize(1<<12);
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|     m_demodBufferFill = 0;
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| 
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|     applySettings(m_settings, true);
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|     applyChannelSettings(m_channelSampleRate, m_channelFrequencyOffset, true);
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| }
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| 
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| WFMModSource::~WFMModSource()
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| {
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|     delete m_rfFilter;
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|     delete[] m_rfFilterBuffer;
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| }
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| 
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| void WFMModSource::pull(SampleVector::iterator begin, unsigned int nbSamples)
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| {
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|     std::for_each(
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|         begin,
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|         begin + nbSamples,
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|         [this](Sample& s) {
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|             pullOne(s);
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|         }
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|     );
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| }
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| 
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| void WFMModSource::pullOne(Sample& sample)
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| {
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| 	if (m_settings.m_channelMute)
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| 	{
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| 		sample.m_real = 0.0f;
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| 		sample.m_imag = 0.0f;
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| 		return;
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| 	}
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| 
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| 	Complex ci, ri;
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|     fftfilt::cmplx *rf;
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|     int rf_out;
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|   	Real t;
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| 
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| 	if ((m_settings.m_modAFInput == WFMModSettings::WFMModInputFile)
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| 	 || (m_settings.m_modAFInput == WFMModSettings::WFMModInputAudio)
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|      || (m_settings.m_modAFInput == WFMModSettings::WFMModInputCWTone))
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| 	{
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|         if (m_interpolatorDistance > 1.0f) // decimate
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|         {
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|             modulateAudio();
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| 
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|             while (!m_interpolator.decimate(&m_interpolatorDistanceRemain, m_modSample, &ri)) {
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|                 modulateAudio();
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|             }
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|         }
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|         else // interpolate
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|         {
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|             if (m_interpolator.interpolate(&m_interpolatorDistanceRemain, m_modSample, &ri)) {
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|                 modulateAudio();
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|             }
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|         }
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| 
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|         t = ri.real();
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| 	    m_interpolatorDistanceRemain += m_interpolatorDistance;
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| 	}
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| 	else
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| 	{
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| 	    pullAF(t);
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|         calculateLevel(t);
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| 	}
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| 
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|     m_modPhasor += (m_settings.m_fmDeviation / (float) m_channelSampleRate) * t * M_PI * 2.0f;
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| 
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|     // limit phasor range to ]-pi,pi]
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|     if (m_modPhasor > M_PI) {
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|         m_modPhasor -= (2.0f * M_PI);
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|     }
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| 
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|     ci.real(cos(m_modPhasor) * 0.891235351562f * SDR_TX_SCALEF); // -1 dB
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|     ci.imag(sin(m_modPhasor) * 0.891235351562f * SDR_TX_SCALEF);
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| 
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|     // RF filtering
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|     rf_out = m_rfFilter->runFilt(ci, &rf);
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| 
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|     if (rf_out > 0)
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|     {
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|         memcpy((void *) m_rfFilterBuffer, (const void *) rf, rf_out*sizeof(Complex));
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|         m_rfFilterBufferIndex = 0;
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| 
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|     }
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| 
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|     ci = m_rfFilterBuffer[m_rfFilterBufferIndex] * m_carrierNco.nextIQ(); // shift to carrier frequency
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|     m_rfFilterBufferIndex++;
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| 
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|     double magsq = ci.real() * ci.real() + ci.imag() * ci.imag();
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| 	magsq /= (SDR_TX_SCALED*SDR_TX_SCALED);
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| 	m_movingAverage(magsq);
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| 	m_magsq = m_movingAverage.asDouble();
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| 
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| 	sample.m_real = (FixReal) ci.real();
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| 	sample.m_imag = (FixReal) ci.imag();
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| 
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|     m_demodBuffer[m_demodBufferFill] = t * std::numeric_limits<int16_t>::max();
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|     ++m_demodBufferFill;
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| 
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|     if (m_demodBufferFill >= m_demodBuffer.size())
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|     {
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|         QList<DataFifo*> *dataFifos = MainCore::instance()->getDataPipes().getFifos(m_channel, "demod");
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| 
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|         if (dataFifos)
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|         {
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|             QList<DataFifo*>::iterator it = dataFifos->begin();
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| 
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|             for (; it != dataFifos->end(); ++it) {
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|                 (*it)->write((quint8*) &m_demodBuffer[0], m_demodBuffer.size() * sizeof(qint16), DataFifo::DataTypeI16);
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|             }
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|         }
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| 
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|         m_demodBufferFill = 0;
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|     }
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| }
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| 
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| void WFMModSource::modulateAudio()
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| {
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|     Real t;
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|     pullAF(t);
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|     calculateLevel(t);
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|     m_modSample.real(t);
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|     m_modSample.imag(0.0f);
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| 
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|     if (m_settings.m_feedbackAudioEnable) {
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|         pushFeedback(t * m_settings.m_feedbackVolumeFactor * 16384.0f);
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|     }
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| }
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| 
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| void WFMModSource::prefetch(unsigned int nbSamples)
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| {
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|     unsigned int nbSamplesAudio = nbSamples * ((Real) m_audioSampleRate / (Real) m_channelSampleRate);
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|     pullAudio(nbSamplesAudio);
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| }
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| 
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| void WFMModSource::pullAudio(unsigned int nbSamplesAudio)
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| {
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|     QMutexLocker mlock(&m_mutex);
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| 
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|     if (nbSamplesAudio > m_audioBuffer.size()) {
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|         m_audioBuffer.resize(nbSamplesAudio);
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|     }
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| 
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|     std::copy(&m_audioReadBuffer[0], &m_audioReadBuffer[nbSamplesAudio], &m_audioBuffer[0]);
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|     m_audioBufferFill = 0;
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| 
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|     if (m_audioReadBufferFill > nbSamplesAudio) // copy back remaining samples at the start of the read buffer
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|     {
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|         std::copy(&m_audioReadBuffer[nbSamplesAudio], &m_audioReadBuffer[m_audioReadBufferFill], &m_audioReadBuffer[0]);
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|         m_audioReadBufferFill = m_audioReadBufferFill - nbSamplesAudio; // adjust current read buffer fill pointer
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|     }
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| }
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| 
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| void WFMModSource::pullAF(Real& sample)
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| {
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|     switch (m_settings.m_modAFInput)
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|     {
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|     case WFMModSettings::WFMModInputTone:
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|         sample = m_toneNco.next() * m_settings.m_volumeFactor;
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|         break;
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|     case WFMModSettings::WFMModInputFile:
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|         // sox f4exb_call.wav --encoding float --endian little f4exb_call.raw
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|         // ffplay -f f32le -ar 48k -ac 1 f4exb_call.raw
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|         if (m_ifstream && m_ifstream->is_open())
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|         {
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|             if (m_ifstream->eof())
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|             {
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|             	if (m_settings.m_playLoop)
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|             	{
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|                     m_ifstream->clear();
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|                     m_ifstream->seekg(0, std::ios::beg);
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|             	}
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|             }
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| 
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|             if (m_ifstream->eof())
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|             {
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|             	sample = 0.0f;
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|             }
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|             else
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|             {
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|                 Real s;
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|             	m_ifstream->read(reinterpret_cast<char*>(&s), sizeof(Real));
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|             	sample = s * m_settings.m_volumeFactor;
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|             }
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|         }
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|         else
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|         {
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|             sample = 0.0f;
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|         }
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|         break;
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|     case WFMModSettings::WFMModInputAudio:
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|         {
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|             if (m_audioBufferFill < m_audioBuffer.size())
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|             {
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|                 sample = ((m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) / 65536.0f) * m_settings.m_volumeFactor;
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|                 m_audioBufferFill++;
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|             }
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|             else
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|             {
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|                 unsigned int size = m_audioBuffer.size();
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|                 qDebug("WFMModSource::pullAF: starve audio samples: size: %u", size);
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|                 sample = ((m_audioBuffer[size-1].l + m_audioBuffer[size-1].r) / 65536.0f) * m_settings.m_volumeFactor;
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|             }
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|         }
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|         break;
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|     case WFMModSettings::WFMModInputCWTone:
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|         Real fadeFactor;
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| 
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|         if (m_cwKeyer.getSample())
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|         {
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|             m_cwKeyer.getCWSmoother().getFadeSample(true, fadeFactor);
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|             sample = m_cwToneNco.next() * m_settings.m_volumeFactor * fadeFactor * 0.99f;
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|         }
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|         else
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|         {
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|             if (m_cwKeyer.getCWSmoother().getFadeSample(false, fadeFactor))
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|             {
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|                 sample = m_cwToneNco.next() * m_settings.m_volumeFactor * fadeFactor * 0.99f;
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|             }
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|             else
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|             {
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|                 sample = 0.0f;
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|                 m_cwToneNco.setPhase(0);
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|             }
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|         }
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|         break;
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|     case WFMModSettings::WFMModInputNone:
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|     default:
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|         sample = 0.0f;
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|         break;
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|     }
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| }
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| 
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| void WFMModSource::pushFeedback(Complex c)
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| {
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|     Complex ci;
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| 
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|     if (m_feedbackInterpolatorDistance < 1.0f) // interpolate
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|     {
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|         while (!m_feedbackInterpolator.interpolate(&m_feedbackInterpolatorDistanceRemain, c, &ci))
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|         {
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|             processOneSample(ci);
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|             m_feedbackInterpolatorDistanceRemain += m_feedbackInterpolatorDistance;
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|         }
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|     }
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|     else // decimate
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|     {
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|         if (m_feedbackInterpolator.decimate(&m_feedbackInterpolatorDistanceRemain, c, &ci))
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|         {
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|             processOneSample(ci);
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|             m_feedbackInterpolatorDistanceRemain += m_feedbackInterpolatorDistance;
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|         }
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|     }
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| }
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| 
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| void WFMModSource::processOneSample(Complex& ci)
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| {
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|     m_feedbackAudioBuffer[m_feedbackAudioBufferFill].l = ci.real();
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|     m_feedbackAudioBuffer[m_feedbackAudioBufferFill].r = ci.imag();
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|     ++m_feedbackAudioBufferFill;
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| 
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|     if (m_feedbackAudioBufferFill >= m_feedbackAudioBuffer.size())
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|     {
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|         unsigned int res = m_feedbackAudioFifo.write((const quint8*)&m_feedbackAudioBuffer[0], m_feedbackAudioBufferFill);
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| 
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|         if (res != m_feedbackAudioBufferFill)
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|         {
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|             qDebug("WFMModSource::processOneSample: %u/%u audio samples written m_feedbackInterpolatorDistance: %f",
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|                 res, m_feedbackAudioBufferFill, m_feedbackInterpolatorDistance);
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|             m_feedbackAudioFifo.clear();
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|         }
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| 
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|         m_feedbackAudioBufferFill = 0;
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|     }
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| }
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| 
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| void WFMModSource::calculateLevel(const Real& sample)
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| {
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|     if (m_levelCalcCount < m_levelNbSamples)
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|     {
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|         m_peakLevel = std::max(std::fabs(m_peakLevel), sample);
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|         m_levelSum += sample * sample;
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|         m_levelCalcCount++;
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|     }
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|     else
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|     {
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|         m_rmsLevel = sqrt(m_levelSum / m_levelNbSamples);
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|         m_peakLevelOut = m_peakLevel;
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|         m_peakLevel = 0.0f;
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|         m_levelSum = 0.0f;
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|         m_levelCalcCount = 0;
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|     }
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| }
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| 
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| void WFMModSource::applyAudioSampleRate(int sampleRate)
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| {
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|     if (sampleRate < 0)
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|     {
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|         qWarning("WFMModSource::applyAudioSampleRate: %d", sampleRate);
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|         return;
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|     }
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| 
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|     qDebug("WFMModSource::applyAudioSampleRate: %d", sampleRate);
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| 
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|     m_interpolatorDistanceRemain = 0;
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|     m_interpolatorConsumed = false;
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|     m_interpolatorDistance = (Real) sampleRate / (Real) m_channelSampleRate;
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|     m_interpolator.create(48, sampleRate, m_settings.m_rfBandwidth / 2.2, 3.0);
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|     m_cwToneNco.setFreq(m_settings.m_toneFrequency, sampleRate);
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|     m_cwKeyer.setSampleRate(sampleRate);
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|     m_cwKeyer.reset();
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|     m_audioSampleRate = sampleRate;
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|     applyFeedbackAudioSampleRate(m_feedbackAudioSampleRate);
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| 
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|     QList<MessageQueue*> *messageQueues = MainCore::instance()->getMessagePipes().getMessageQueues(m_channel, "reportdemod");
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| 
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|     if (messageQueues)
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|     {
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|         QList<MessageQueue*>::iterator it = messageQueues->begin();
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| 
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|         for (; it != messageQueues->end(); ++it)
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|         {
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|             MainCore::MsgChannelDemodReport *msg = MainCore::MsgChannelDemodReport::create(m_channel, sampleRate);
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|             (*it)->push(msg);
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|         }
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|     }
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| }
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| 
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| void WFMModSource::applyFeedbackAudioSampleRate(int sampleRate)
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| {
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|     if (sampleRate < 0)
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|     {
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|         qWarning("WFMModSource::applyFeedbackAudioSampleRate: invalid sample rate %d", sampleRate);
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|         return;
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|     }
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| 
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|     qDebug("WFMModSource::applyFeedbackAudioSampleRate: %d", sampleRate);
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| 
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|     m_feedbackInterpolatorDistanceRemain = 0;
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|     m_feedbackInterpolatorConsumed = false;
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|     m_feedbackInterpolatorDistance = (Real) sampleRate / (Real) m_audioSampleRate;
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|     Real cutoff = std::min(sampleRate, m_audioSampleRate) / 2.2f;
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|     m_feedbackInterpolator.create(48, sampleRate, cutoff, 3.0);
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| 
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|     m_feedbackAudioSampleRate = sampleRate;
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| }
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| 
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| void WFMModSource::applySettings(const WFMModSettings& settings, bool force)
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| {
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|     if ((settings.m_afBandwidth != m_settings.m_afBandwidth) || force)
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|     {
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|         m_interpolatorDistanceRemain = 0;
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|         m_interpolatorConsumed = false;
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|         m_interpolatorDistance = (Real) m_audioSampleRate / (Real) m_channelSampleRate;
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|         m_interpolator.create(48, m_audioSampleRate, settings.m_afBandwidth / 2.2, 3.0);
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|     }
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| 
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|     if ((settings.m_rfBandwidth != m_settings.m_rfBandwidth) || force)
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|     {
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|         Real lowCut = -(settings.m_rfBandwidth / 2.2) / m_channelSampleRate;
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|         Real hiCut  = (settings.m_rfBandwidth / 2.2) / m_channelSampleRate;
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|         m_rfFilter->create_filter(lowCut, hiCut);
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|     }
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| 
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|     if ((settings.m_toneFrequency != m_settings.m_toneFrequency) || force)
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|     {
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|         m_toneNco.setFreq(settings.m_toneFrequency, m_channelSampleRate);
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|         m_cwToneNco.setFreq(settings.m_toneFrequency, m_audioSampleRate);
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|     }
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| 
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|     if ((settings.m_modAFInput != m_settings.m_modAFInput) || force)
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|     {
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|         if (settings.m_modAFInput == WFMModSettings::WFMModInputAudio) {
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|             connect(&m_audioFifo, SIGNAL(dataReady()), this, SLOT(handleAudio()));
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|         } else {
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|             disconnect(&m_audioFifo, SIGNAL(dataReady()), this, SLOT(handleAudio()));
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|         }
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|     }
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| 
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|     m_settings = settings;
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| }
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| 
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| void WFMModSource::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
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| {
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|     qDebug() << "WFMModSource::applyChannelSettings:"
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|             << " channelSampleRate: " << channelSampleRate
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|             << " channelFrequencyOffset: " << channelFrequencyOffset;
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| 
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|     if ((channelFrequencyOffset != m_channelFrequencyOffset)
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|      || (channelSampleRate != m_channelSampleRate) || force) {
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|         m_carrierNco.setFreq(channelFrequencyOffset, channelSampleRate);
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|     }
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| 
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|     if ((channelSampleRate != m_channelSampleRate) || force)
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|     {
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|         m_interpolatorDistanceRemain = 0;
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|         m_interpolatorConsumed = false;
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|         m_interpolatorDistance = (Real) m_audioSampleRate / (Real) channelSampleRate;
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|         m_interpolator.create(48, m_audioSampleRate, m_settings.m_afBandwidth / 2.2, 3.0);
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|         Real lowCut = -(m_settings.m_rfBandwidth / 2.0) / channelSampleRate;
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|         Real hiCut  = (m_settings.m_rfBandwidth / 2.0) / channelSampleRate;
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|         m_rfFilter->create_filter(lowCut, hiCut);
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|         m_toneNco.setFreq(m_settings.m_toneFrequency, channelSampleRate);
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|     }
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| 
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|     m_channelSampleRate = channelSampleRate;
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|     m_channelFrequencyOffset = channelFrequencyOffset;
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| }
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| 
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| void WFMModSource::handleAudio()
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| {
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|     QMutexLocker mlock(&m_mutex);
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|     unsigned int nbRead;
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| 
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|     while ((nbRead = m_audioFifo.read(reinterpret_cast<quint8*>(&m_audioReadBuffer[m_audioReadBufferFill]), 4096)) != 0)
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|     {
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|         if (m_audioReadBufferFill + nbRead + 4096 < m_audioReadBuffer.size()) {
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|             m_audioReadBufferFill += nbRead;
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|         }
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|     }
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| }
 |