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sdrangel/plugins/channelrx/demodft8/ft8demodsink.h
2023-01-18 23:00:59 +01:00

155 lines
4.6 KiB
C++

///////////////////////////////////////////////////////////////////////////////////
// Copyright (C) 2019 Edouard Griffiths, F4EXB //
// //
// This program is free software; you can redistribute it and/or modify //
// it under the terms of the GNU General Public License as published by //
// the Free Software Foundation as version 3 of the License, or //
// (at your option) any later version. //
// //
// This program is distributed in the hope that it will be useful, //
// but WITHOUT ANY WARRANTY; without even the implied warranty of //
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
// GNU General Public License V3 for more details. //
// //
// You should have received a copy of the GNU General Public License //
// along with this program. If not, see <http://www.gnu.org/licenses/>. //
///////////////////////////////////////////////////////////////////////////////////
#ifndef INCLUDE_FT8DEMODSINK_H
#define INCLUDE_FT8DEMODSINK_H
#include <QVector>
#include "dsp/channelsamplesink.h"
#include "dsp/ncof.h"
#include "dsp/interpolator.h"
#include "dsp/fftfilt.h"
#include "dsp/agc.h"
#include "util/doublebufferfifo.h"
#include "ft8demodsettings.h"
class SpectrumVis;
class ChannelAPI;
class FT8Buffer;
class FT8DemodSink : public ChannelSampleSink {
public:
FT8DemodSink();
~FT8DemodSink();
virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end);
void setSpectrumSink(SpectrumVis* spectrumSink) { m_spectrumSink = spectrumSink; }
void setFT8Buffer(FT8Buffer *buffer) { m_ft8Buffer = buffer; }
void applyChannelSettings(int inputSampleRate, int inputFrequencyOffset, bool force = false);
void applySettings(const FT8DemodSettings& settings, bool force = false);
void applyFT8SampleRate();
double getMagSq() const { return m_magsq; }
bool getAudioActive() const { return m_audioActive; }
void setChannel(ChannelAPI *channel) { m_channel = channel; }
void getMagSqLevels(double& avg, double& peak, int& nbSamples)
{
if (m_magsqCount > 0)
{
m_magsq = m_magsqSum / m_magsqCount;
m_magSqLevelStore.m_magsq = m_magsq;
m_magSqLevelStore.m_magsqPeak = m_magsqPeak;
}
avg = m_magSqLevelStore.m_magsq;
peak = m_magSqLevelStore.m_magsqPeak;
nbSamples = m_magsqCount == 0 ? 1 : m_magsqCount;
m_magsqSum = 0.0f;
m_magsqPeak = 0.0f;
m_magsqCount = 0;
}
/**
* Level changed
* \param rmsLevel RMS level in range 0.0 - 1.0
* \param peakLevel Peak level in range 0.0 - 1.0
* \param numSamples Number of samples analyzed
*/
void getLevels(qreal& rmsLevel, qreal& peakLevel, int& numSamples)
{
rmsLevel = m_rmsLevel;
peakLevel = m_peakLevel;
numSamples = m_levelInNbSamples;
}
private:
struct MagSqLevelsStore
{
MagSqLevelsStore() :
m_magsq(1e-12),
m_magsqPeak(1e-12)
{}
double m_magsq;
double m_magsqPeak;
};
struct LevelRMS
{
LevelRMS();
void accumulate(float fsample);
double m_sum;
float m_peak;
int m_n;
bool m_reset;
};
FT8DemodSettings m_settings;
ChannelAPI *m_channel;
Real m_Bandwidth;
Real m_LowCutoff;
Real m_volume;
int m_spanLog2;
fftfilt::cmplx m_sum;
int m_undersampleCount;
int m_channelSampleRate;
int m_channelFrequencyOffset;
bool m_usb;
double m_magsq;
double m_magsqSum;
double m_magsqPeak;
int m_magsqCount;
MagSqLevelsStore m_magSqLevelStore;
MagAGC m_agc;
bool m_agcActive;
bool m_audioActive; //!< True if an audio signal is produced (no AGC or AGC and above threshold)
NCOF m_nco;
Interpolator m_interpolator;
Real m_interpolatorDistance;
Real m_interpolatorDistanceRemain;
fftfilt* SSBFilter;
SpectrumVis* m_spectrumSink;
SampleVector m_sampleBuffer;
FT8Buffer *m_ft8Buffer;
QVector<qint16> m_demodBuffer;
int m_demodBufferFill;
LevelRMS m_levelIn;
int m_levelInNbSamples;
Real m_rmsLevel;
Real m_peakLevel;
static const int m_ssbFftLen;
static const int m_agcTarget;
void processOneSample(Complex &ci);
void calculateLevel(int16_t& sample);
};
#endif // INCLUDE_FT8DEMODSINK_H