1
0
mirror of https://github.com/f4exb/sdrangel.git synced 2024-11-05 00:11:16 -05:00
sdrangel/plugins/channelrx/demodssb/ssbdemodsink.h
2019-12-04 07:20:05 +01:00

130 lines
4.3 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_SSBDEMODSINK_H
#define INCLUDE_SSBDEMODSINK_H
#include <vector>
#include "dsp/channelsamplesink.h"
#include "dsp/ncof.h"
#include "dsp/interpolator.h"
#include "dsp/fftfilt.h"
#include "dsp/agc.h"
#include "audio/audiofifo.h"
#include "util/doublebufferfifo.h"
#include "ssbdemodsettings.h"
class BasebandSampleSink;
class SSBDemodSink : public ChannelSampleSink {
public:
SSBDemodSink();
~SSBDemodSink();
virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end);
void setSpectrumSink(BasebandSampleSink* 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 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;
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 befor threshold triggers
DoubleBufferFIFO<fftfilt::cmplx> m_squelchDelayLine;
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;
fftfilt* DSBFilter;
BasebandSampleSink* m_spectrumSink;
SampleVector m_sampleBuffer;
AudioVector m_audioBuffer;
uint m_audioBufferFill;
AudioFifo m_audioFifo;
quint32 m_audioSampleRate;
static const int m_ssbFftLen;
static const int m_agcTarget;
void processOneSample(Complex &ci);
};
#endif // INCLUDE_SSBDEMODSINK_H