mirror of
https://github.com/f4exb/sdrangel.git
synced 2024-12-24 10:50:29 -05:00
143 lines
4.7 KiB
C++
143 lines
4.7 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_SSBDEMODSINK_H
|
|
#define INCLUDE_SSBDEMODSINK_H
|
|
|
|
#include <QVector>
|
|
|
|
#include "dsp/channelsamplesink.h"
|
|
#include "dsp/ncof.h"
|
|
#include "dsp/interpolator.h"
|
|
#include "dsp/fftfilt.h"
|
|
#include "dsp/agc.h"
|
|
#include "dsp/firfilter.h"
|
|
#include "audio/audiofifo.h"
|
|
#include "util/doublebufferfifo.h"
|
|
|
|
#include "ssbdemodsettings.h"
|
|
|
|
class SpectrumVis;
|
|
class ChannelAPI;
|
|
|
|
class SSBDemodSink : public ChannelSampleSink {
|
|
public:
|
|
SSBDemodSink();
|
|
~SSBDemodSink();
|
|
|
|
virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end);
|
|
|
|
void setSpectrumSink(SpectrumVis* spectrumSink) { m_spectrumSink = spectrumSink; }
|
|
void applyChannelSettings(int inputSampleRate, int inputFrequencyOffset, bool force = false);
|
|
void applySettings(const SSBDemodSettings& settings, bool force = false);
|
|
void applyAudioSampleRate(int sampleRate);
|
|
|
|
AudioFifo *getAudioFifo() { return &m_audioFifo; }
|
|
double getMagSq() const { return m_magsq; }
|
|
bool getAudioActive() const { return m_audioActive; }
|
|
void setChannel(ChannelAPI *channel) { m_channel = channel; }
|
|
void setAudioFifoLabel(const QString& label) { m_audioFifo.setLabel(label); }
|
|
void setDNR(bool dnr);
|
|
|
|
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;
|
|
};
|
|
|
|
SSBDemodSettings 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_audioBinaual;
|
|
bool m_audioFlipChannels;
|
|
bool m_usb;
|
|
bool m_dsb;
|
|
bool m_audioMute;
|
|
double m_magsq;
|
|
double m_magsqSum;
|
|
double m_magsqPeak;
|
|
int m_magsqCount;
|
|
MagSqLevelsStore m_magSqLevelStore;
|
|
MagAGC m_agc;
|
|
bool m_agcActive;
|
|
bool m_agcClamping;
|
|
int m_agcNbSamples; //!< number of audio (48 kHz) samples for AGC averaging
|
|
double m_agcPowerThreshold; //!< AGC power threshold (linear)
|
|
int m_agcThresholdGate; //!< Gate length in number of samples before threshold triggers
|
|
DoubleBufferFIFO<fftfilt::cmplx> m_squelchDelayLine;
|
|
bool m_audioActive; //!< True if an audio signal is produced (no AGC or AGC and above threshold)
|
|
Lowpass<Real> m_lowpassI;
|
|
Lowpass<Real> m_lowpassQ;
|
|
|
|
|
|
NCOF m_nco;
|
|
Interpolator m_interpolator;
|
|
Real m_interpolatorDistance;
|
|
Real m_interpolatorDistanceRemain;
|
|
fftfilt* SSBFilter;
|
|
fftfilt* DSBFilter;
|
|
|
|
SpectrumVis* m_spectrumSink;
|
|
SampleVector m_sampleBuffer;
|
|
|
|
AudioVector m_audioBuffer;
|
|
std::size_t m_audioBufferFill;
|
|
AudioFifo m_audioFifo;
|
|
quint32 m_audioSampleRate;
|
|
|
|
QVector<qint16> m_demodBuffer;
|
|
int m_demodBufferFill;
|
|
|
|
static const int m_ssbFftLen;
|
|
static const int m_agcTarget;
|
|
|
|
void processOneSample(Complex &ci);
|
|
};
|
|
|
|
#endif // INCLUDE_SSBDEMODSINK_H
|