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sdrangel/plugins/channelrx/demodam/amdemodsink.h

135 lines
4.4 KiB
C++

///////////////////////////////////////////////////////////////////////////////////
// Copyright (C) 2019-2020, 2022-2023 Edouard Griffiths, F4EXB <f4exb06@gmail.com> //
// //
// 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_AMDEMODSINK_H
#define INCLUDE_AMDEMODSINK_H
#include <QVector>
#include "dsp/channelsamplesink.h"
#include "dsp/nco.h"
#include "dsp/interpolator.h"
#include "dsp/agc.h"
#include "dsp/firfilter.h"
#include "dsp/phaselockcomplex.h"
#include "audio/audiofifo.h"
#include "util/movingaverage.h"
#include "util/doublebufferfifo.h"
#include "amdemodsettings.h"
class fftfilt;
class ChannelAPI;
class AMDemodSink : public ChannelSampleSink {
public:
AMDemodSink();
~AMDemodSink();
virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end);
void applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force = false);
void applySettings(const AMDemodSettings& settings, bool force = false);
void applyAudioSampleRate(int sampleRate);
int getAudioSampleRate() const { return m_audioSampleRate; }
double getMagSq() const { return m_magsq; }
bool getSquelchOpen() const { return m_squelchOpen; }
bool getPllLocked() const { return m_settings.m_pll && m_pll.locked(); }
Real getPllFrequency() const { return m_pll.getFreq(); }
AudioFifo *getAudioFifo() { return &m_audioFifo; }
void setAudioFifoLabel(const QString& label) { m_audioFifo.setLabel(label); }
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;
}
private:
struct MagSqLevelsStore
{
MagSqLevelsStore() :
m_magsq(1e-12),
m_magsqPeak(1e-12)
{}
double m_magsq;
double m_magsqPeak;
};
enum RateState {
RSInitialFill,
RSRunning
};
int m_channelSampleRate;
int m_channelFrequencyOffset;
AMDemodSettings m_settings;
ChannelAPI *m_channel;
int m_audioSampleRate;
NCO m_nco;
Interpolator m_interpolator;
Real m_interpolatorDistance;
Real m_interpolatorDistanceRemain;
Real m_squelchLevel;
int m_squelchCount;
bool m_squelchOpen;
DoubleBufferFIFO<Real> m_squelchDelayLine;
double m_magsq;
double m_magsqSum;
double m_magsqPeak;
int m_magsqCount;
MagSqLevelsStore m_magSqLevelStore;
MovingAverageUtil<Real, double, 16> m_movingAverage;
SimpleAGC<4800> m_volumeAGC;
Bandpass<Real> m_bandpass;
Lowpass<Real> m_lowpass;
Lowpass<std::complex<float> > m_pllFilt;
PhaseLockComplex m_pll;
fftfilt* DSBFilter;
fftfilt* SSBFilter;
Real m_syncAMBuff[2*1024];
uint32_t m_syncAMBuffIndex;
MagAGC m_syncAMAGC;
AudioVector m_audioBuffer;
AudioFifo m_audioFifo;
std::size_t m_audioBufferFill;
QVector<qint16> m_demodBuffer;
int m_demodBufferFill;
void processOneSample(Complex &ci);
};
#endif // INCLUDE_AMDEMODSINK_H