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sdrangel/plugins/channelrx/demodssb/ssbdemod.h

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///////////////////////////////////////////////////////////////////////////////////
// Copyright (C) 2012 maintech GmbH, Otto-Hahn-Str. 15, 97204 Hoechberg, Germany //
// written by Christian Daniel //
// //
// 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 //
// //
// 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_SSBDEMOD_H
#define INCLUDE_SSBDEMOD_H
#include <dsp/basebandsamplesink.h>
#include <QMutex>
#include <vector>
#include "dsp/ncof.h"
#include "dsp/interpolator.h"
#include "dsp/fftfilt.h"
#include "dsp/agc.h"
#include "audio/audiofifo.h"
#include "util/message.h"
#define ssbFftLen 1024
#define agcTarget 3276.8 // -10 dB amplitude => -20 dB power: center of normal signal
class SSBDemod : public BasebandSampleSink {
public:
SSBDemod(BasebandSampleSink* sampleSink);
virtual ~SSBDemod();
void configure(MessageQueue* messageQueue,
Real Bandwidth,
Real LowCutoff,
Real volume,
int spanLog2,
bool audioBinaural,
bool audioFlipChannels,
bool dsb,
bool audioMute,
bool agc,
bool agcClamping,
int agcTimeLog2,
int agcPowerThreshold,
int agcThresholdGate);
virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end, bool positiveOnly);
virtual void start();
virtual void stop();
virtual bool handleMessage(const Message& cmd);
double getMagSq() const { return m_magsq; }
bool getAudioActive() const { return m_audioActive; }
void getMagSqLevels(double& avg, double& peak, int& nbSamples)
{
avg = m_magsqCount == 0 ? 1e-10 : m_magsqSum / m_magsqCount;
m_magsq = avg;
peak = m_magsqPeak == 0.0 ? 1e-10 : m_magsqPeak;
nbSamples = m_magsqCount == 0 ? 1 : m_magsqCount;
m_magsqSum = 0.0f;
m_magsqPeak = 0.0f;
m_magsqCount = 0;
}
private:
class MsgConfigureSSBDemod : public Message {
MESSAGE_CLASS_DECLARATION
public:
Real getBandwidth() const { return m_Bandwidth; }
Real getLoCutoff() const { return m_LowCutoff; }
Real getVolume() const { return m_volume; }
int getSpanLog2() const { return m_spanLog2; }
bool getAudioBinaural() const { return m_audioBinaural; }
bool getAudioFlipChannels() const { return m_audioFlipChannels; }
bool getDSB() const { return m_dsb; }
bool getAudioMute() const { return m_audioMute; }
bool getAGC() const { return m_agc; }
bool getAGCClamping() const { return m_agcClamping; }
int getAGCTimeLog2() const { return m_agcTimeLog2; }
int getAGCPowerThershold() const { return m_agcPowerThreshold; }
int getAGCThersholdGate() const { return m_agcThresholdGate; }
static MsgConfigureSSBDemod* create(Real Bandwidth,
Real LowCutoff,
Real volume,
int spanLog2,
bool audioBinaural,
bool audioFlipChannels,
bool dsb,
bool audioMute,
bool agc,
bool agcClamping,
int agcTimeLog2,
int agcPowerThreshold,
int agcThresholdGate)
{
return new MsgConfigureSSBDemod(
Bandwidth,
LowCutoff,
volume,
spanLog2,
audioBinaural,
audioFlipChannels,
dsb,
audioMute,
agc,
agcClamping,
agcTimeLog2,
agcPowerThreshold,
agcThresholdGate);
}
private:
Real m_Bandwidth;
Real m_LowCutoff;
Real m_volume;
int m_spanLog2;
bool m_audioBinaural;
bool m_audioFlipChannels;
bool m_dsb;
bool m_audioMute;
bool m_agc;
bool m_agcClamping;
int m_agcTimeLog2;
int m_agcPowerThreshold;
int m_agcThresholdGate;
MsgConfigureSSBDemod(Real Bandwidth,
Real LowCutoff,
Real volume,
int spanLog2,
bool audioBinaural,
bool audioFlipChannels,
bool dsb,
bool audioMute,
bool agc,
bool agcClamping,
int agcTimeLog2,
int agcPowerThreshold,
int agcThresholdGate) :
Message(),
m_Bandwidth(Bandwidth),
m_LowCutoff(LowCutoff),
m_volume(volume),
m_spanLog2(spanLog2),
m_audioBinaural(audioBinaural),
m_audioFlipChannels(audioFlipChannels),
m_dsb(dsb),
m_audioMute(audioMute),
m_agc(agc),
m_agcClamping(agcClamping),
m_agcTimeLog2(agcTimeLog2),
m_agcPowerThreshold(agcPowerThreshold),
m_agcThresholdGate(agcThresholdGate)
{ }
};
struct AudioSample {
qint16 l;
qint16 r;
};
typedef std::vector<AudioSample> AudioVector;
Real m_Bandwidth;
Real m_LowCutoff;
Real m_volume;
int m_spanLog2;
fftfilt::cmplx m_sum;
int m_undersampleCount;
int m_sampleRate;
int m_frequency;
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;
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
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_sampleDistanceRemain;
fftfilt* SSBFilter;
fftfilt* DSBFilter;
BasebandSampleSink* m_sampleSink;
SampleVector m_sampleBuffer;
AudioVector m_audioBuffer;
uint m_audioBufferFill;
AudioFifo m_audioFifo;
quint32 m_audioSampleRate;
QMutex m_settingsMutex;
};
#endif // INCLUDE_SSBDEMOD_H
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