mirror of
https://github.com/f4exb/sdrangel.git
synced 2024-11-29 19:28:47 -05:00
172 lines
5.5 KiB
C++
172 lines
5.5 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
|
|
// Copyright (C) 2019-2020, 2022-2023 Edouard Griffiths, F4EXB <f4exb06@gmail.com> //
|
|
// Copyright (C) 2022 Jon Beniston, M7RCE <jon@beniston.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_BFMDEMODSINK_H
|
|
#define INCLUDE_BFMDEMODSINK_H
|
|
|
|
#include <vector>
|
|
|
|
#include <QVector>
|
|
|
|
#include "dsp/channelsamplesink.h"
|
|
#include "dsp/nco.h"
|
|
#include "dsp/interpolator.h"
|
|
#include "dsp/firfilter.h"
|
|
#include "dsp/fftfilt.h"
|
|
#include "dsp/phaselock.h"
|
|
#include "dsp/filterrc.h"
|
|
#include "dsp/phasediscri.h"
|
|
#include "audio/audiofifo.h"
|
|
|
|
#include "rdsparser.h"
|
|
#include "rdsdecoder.h"
|
|
#include "rdsdemod.h"
|
|
#include "bfmdemodsettings.h"
|
|
|
|
class ChannelAPI;
|
|
class BasebandSampleSink;
|
|
|
|
class BFMDemodSink : public ChannelSampleSink {
|
|
public:
|
|
BFMDemodSink();
|
|
~BFMDemodSink();
|
|
|
|
virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end);
|
|
|
|
void setSpectrumSink(BasebandSampleSink* spectrumSink) { m_spectrumSink = spectrumSink; }
|
|
void setChannel(ChannelAPI *channel) { m_channel = channel; }
|
|
|
|
double getMagSq() const { return m_magsq; }
|
|
|
|
bool getPilotLock() const { return m_pilotPLL.locked(); }
|
|
Real getPilotLevel() const { return m_pilotPLL.get_pilot_level(); }
|
|
|
|
Real getDecoderQua() const { return m_rdsDecoder.m_qua; }
|
|
bool getDecoderSynced() const { return m_rdsDecoder.synced(); }
|
|
Real getDemodAcc() const { return m_rdsDemod.m_report.acc; }
|
|
Real getDemodQua() const { return m_rdsDemod.m_report.qua; }
|
|
Real getDemodFclk() const { return m_rdsDemod.m_report.fclk; }
|
|
int getSquelchState() const { return m_squelchState; }
|
|
|
|
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;
|
|
}
|
|
|
|
RDSParser& getRDSParser() { return m_rdsParser; }
|
|
|
|
void applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force = false);
|
|
void applySettings(const BFMDemodSettings& settings, bool force = false);
|
|
|
|
AudioFifo *getAudioFifo() { return &m_audioFifo; }
|
|
void setAudioFifoLabel(const QString& label) { m_audioFifo.setLabel(label); }
|
|
void applyAudioSampleRate(int sampleRate);
|
|
int getAudioSampleRate() const { return m_audioSampleRate; }
|
|
|
|
private:
|
|
struct MagSqLevelsStore
|
|
{
|
|
MagSqLevelsStore() :
|
|
m_magsq(1e-12),
|
|
m_magsqPeak(1e-12)
|
|
{}
|
|
double m_magsq;
|
|
double m_magsqPeak;
|
|
};
|
|
|
|
enum RateState {
|
|
RSInitialFill,
|
|
RSRunning
|
|
};
|
|
|
|
ChannelAPI *m_channel;
|
|
int m_channelSampleRate;
|
|
int m_channelFrequencyOffset;
|
|
BFMDemodSettings m_settings;
|
|
|
|
int m_audioSampleRate;
|
|
AudioVector m_audioBuffer;
|
|
std::size_t m_audioBufferFill;
|
|
AudioFifo m_audioFifo;
|
|
SampleVector m_sampleBuffer;
|
|
|
|
NCO m_nco;
|
|
Interpolator m_interpolator; //!< Interpolator between fixed demod bandwidth and audio bandwidth (rational)
|
|
Real m_interpolatorDistance;
|
|
Real m_interpolatorDistanceRemain;
|
|
|
|
Interpolator m_interpolatorStereo; //!< Twin Interpolator for stereo subcarrier
|
|
Real m_interpolatorStereoDistance;
|
|
Real m_interpolatorStereoDistanceRemain;
|
|
|
|
Interpolator m_interpolatorRDS; //!< Twin Interpolator for stereo subcarrier
|
|
Real m_interpolatorRDSDistance;
|
|
Real m_interpolatorRDSDistanceRemain;
|
|
|
|
Lowpass<Real> m_lowpass;
|
|
fftfilt* m_rfFilter;
|
|
static const int filtFftLen = 1024;
|
|
|
|
Real m_squelchLevel;
|
|
int m_squelchState;
|
|
|
|
Real m_m1Arg; //!> x^-1 real sample
|
|
|
|
double m_magsq;
|
|
double m_magsqSum;
|
|
double m_magsqPeak;
|
|
int m_magsqCount;
|
|
MagSqLevelsStore m_magSqLevelStore;
|
|
|
|
RDSPhaseLock m_pilotPLL;
|
|
Real m_pilotPLLSamples[4];
|
|
|
|
RDSDemod m_rdsDemod;
|
|
RDSDecoder m_rdsDecoder;
|
|
RDSParser m_rdsParser;
|
|
|
|
LowPassFilterRC m_deemphasisFilterX;
|
|
LowPassFilterRC m_deemphasisFilterY;
|
|
static const Real default_deemphasis;
|
|
|
|
Real m_fmExcursion;
|
|
static const int default_excursion;
|
|
|
|
PhaseDiscriminators m_phaseDiscri;
|
|
|
|
BasebandSampleSink *m_spectrumSink;
|
|
|
|
QVector<qint16> m_demodBuffer;
|
|
int m_demodBufferFill;
|
|
};
|
|
|
|
#endif // INCLUDE_BFMDEMODSINK_H
|