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sdrangel/plugins/channelrx/demoddsd/dsddemodsink.h

165 lines
5.3 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_DSDDEMODSINK_H
#define INCLUDE_DSDDEMODSINK_H
#include <QVector>
#include "dsp/channelsamplesink.h"
#include "dsp/phasediscri.h"
#include "dsp/nco.h"
#include "dsp/interpolator.h"
#include "dsp/firfilter.h"
#include "dsp/afsquelch.h"
#include "dsp/afsquelch.h"
#include "audio/audiofifo.h"
#include "util/movingaverage.h"
#include "util/doublebufferfifo.h"
#include "dsddemodsettings.h"
#include "dsddecoder.h"
class BasebandSampleSink;
class ChannelAPI;
class Feature;
class DSDDemodSink : public ChannelSampleSink {
public:
DSDDemodSink();
~DSDDemodSink();
virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end);
void applyAudioSampleRate(int sampleRate);
void applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force = false);
void applySettings(const DSDDemodSettings& settings, bool force = false);
AudioFifo *getAudioFifo1() { return &m_audioFifo1; }
AudioFifo *getAudioFifo2() { return &m_audioFifo2; }
void setAudioFifoLabel(const QString& label) {
m_audioFifo1.setLabel("1:" + label);
m_audioFifo2.setLabel("2:" + label);
}
int getAudioSampleRate() const { return m_audioSampleRate; }
void setChannel(ChannelAPI *channel) { m_channel = channel; }
void setScopeXYSink(BasebandSampleSink* scopeSink) { m_scopeXY = scopeSink; }
void configureMyPosition(float myLatitude, float myLongitude);
double getMagSq() { return m_magsq; }
bool getSquelchOpen() const { return m_squelchOpen; }
const DSDDecoder& getDecoder() const { return m_dsdDecoder; }
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;
}
const char *updateAndGetStatusText();
void setAmbeFeature(Feature *feature) { m_ambeFeature = feature; }
private:
struct MagSqLevelsStore
{
MagSqLevelsStore() :
m_magsq(1e-12),
m_magsqPeak(1e-12)
{}
double m_magsq;
double m_magsqPeak;
};
typedef enum
{
signalFormatNone,
signalFormatDMR,
signalFormatDStar,
signalFormatDPMR,
signalFormatYSF,
signalFormatNXDN
} SignalFormat; //!< Used for status text formatting
enum RateState {
RSInitialFill,
RSRunning
};
int m_channelSampleRate;
int m_channelFrequencyOffset;
DSDDemodSettings m_settings;
ChannelAPI *m_channel;
Feature *m_ambeFeature;
int m_audioSampleRate;
QVector<qint16> m_demodBuffer;
int m_demodBufferFill;
NCO m_nco;
Interpolator m_interpolator;
Real m_interpolatorDistance;
Real m_interpolatorDistanceRemain;
int m_sampleCount;
int m_squelchCount;
int m_squelchGate;
double m_squelchLevel;
bool m_squelchOpen;
DoubleBufferFIFO<Real> m_squelchDelayLine;
MovingAverageUtil<Real, double, 16> m_movingAverage;
double m_magsq;
double m_magsqSum;
double m_magsqPeak;
int m_magsqCount;
MagSqLevelsStore m_magSqLevelStore;
SampleVector m_scopeSampleBuffer;
AudioVector m_audioBuffer;
uint m_audioBufferFill;
FixReal *m_sampleBuffer; //!< samples ring buffer
int m_sampleBufferIndex;
int m_scaleFromShort;
AudioFifo m_audioFifo1;
AudioFifo m_audioFifo2;
BasebandSampleSink* m_scopeXY;
bool m_scopeEnabled;
DSDDecoder m_dsdDecoder;
char m_formatStatusText[82+1]; //!< Fixed signal format dependent status text
SignalFormat m_signalFormat; //!< Used to keep formatting during successive calls for the same standard type
PhaseDiscriminators m_phaseDiscri;
void formatStatusText();
bool isNotYSFWide();
};
#endif // INCLUDE_DSDDEMODSINK_H