mirror of
https://github.com/f4exb/sdrangel.git
synced 2024-12-23 10:05:46 -05:00
164 lines
5.3 KiB
C++
164 lines
5.3 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_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;
|
|
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
|