mirror of https://github.com/f4exb/sdrangel.git
154 lines
4.6 KiB
C++
154 lines
4.6 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2019-2020, 2022-2023 Edouard Griffiths, F4EXB <f4exb06@gmail.com> //
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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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#ifndef INCLUDE_FT8DEMODSINK_H
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#define INCLUDE_FT8DEMODSINK_H
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#include <QVector>
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#include "dsp/channelsamplesink.h"
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#include "dsp/ncof.h"
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#include "dsp/interpolator.h"
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#include "dsp/fftfilt.h"
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#include "dsp/agc.h"
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#include "ft8demodsettings.h"
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class SpectrumVis;
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class ChannelAPI;
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class FT8Buffer;
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class FT8DemodSink : public ChannelSampleSink {
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public:
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FT8DemodSink();
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~FT8DemodSink();
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virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end);
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void setSpectrumSink(SpectrumVis* spectrumSink) { m_spectrumSink = spectrumSink; }
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void setFT8Buffer(FT8Buffer *buffer) { m_ft8Buffer = buffer; }
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void applyChannelSettings(int inputSampleRate, int inputFrequencyOffset, bool force = false);
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void applySettings(const FT8DemodSettings& settings, bool force = false);
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void applyFT8SampleRate();
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double getMagSq() const { return m_magsq; }
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bool getAudioActive() const { return m_audioActive; }
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void setChannel(ChannelAPI *channel) { m_channel = channel; }
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void getMagSqLevels(double& avg, double& peak, int& nbSamples)
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{
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if (m_magsqCount > 0)
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{
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m_magsq = m_magsqSum / m_magsqCount;
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m_magSqLevelStore.m_magsq = m_magsq;
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m_magSqLevelStore.m_magsqPeak = m_magsqPeak;
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}
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avg = m_magSqLevelStore.m_magsq;
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peak = m_magSqLevelStore.m_magsqPeak;
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nbSamples = m_magsqCount == 0 ? 1 : m_magsqCount;
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m_magsqSum = 0.0f;
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m_magsqPeak = 0.0f;
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m_magsqCount = 0;
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}
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/**
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* Level changed
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* \param rmsLevel RMS level in range 0.0 - 1.0
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* \param peakLevel Peak level in range 0.0 - 1.0
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* \param numSamples Number of samples analyzed
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*/
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void getLevels(qreal& rmsLevel, qreal& peakLevel, int& numSamples)
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{
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rmsLevel = m_rmsLevel;
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peakLevel = m_peakLevel;
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numSamples = m_levelInNbSamples;
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}
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private:
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struct MagSqLevelsStore
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{
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MagSqLevelsStore() :
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m_magsq(1e-12),
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m_magsqPeak(1e-12)
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{}
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double m_magsq;
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double m_magsqPeak;
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};
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struct LevelRMS
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{
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LevelRMS();
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void accumulate(float fsample);
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double m_sum;
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float m_peak;
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int m_n;
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bool m_reset;
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};
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FT8DemodSettings m_settings;
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ChannelAPI *m_channel;
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Real m_Bandwidth;
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Real m_LowCutoff;
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Real m_volume;
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int m_spanLog2;
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fftfilt::cmplx m_sum;
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int m_undersampleCount;
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int m_channelSampleRate;
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int m_channelFrequencyOffset;
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bool m_usb;
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double m_magsq;
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double m_magsqSum;
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double m_magsqPeak;
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int m_magsqCount;
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MagSqLevelsStore m_magSqLevelStore;
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MagAGC m_agc;
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bool m_agcActive;
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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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Interpolator m_interpolator;
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Real m_interpolatorDistance;
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Real m_interpolatorDistanceRemain;
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fftfilt* SSBFilter;
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SpectrumVis* m_spectrumSink;
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SampleVector m_sampleBuffer;
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FT8Buffer *m_ft8Buffer;
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QVector<qint16> m_demodBuffer;
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int m_demodBufferFill;
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LevelRMS m_levelIn;
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int m_levelInNbSamples;
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Real m_rmsLevel;
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Real m_peakLevel;
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static const int m_ssbFftLen;
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static const int m_agcTarget;
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void processOneSample(Complex &ci);
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void calculateLevel(int16_t& sample);
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};
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#endif // INCLUDE_FT8DEMODSINK_H
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