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134 lines
4.3 KiB
C++
134 lines
4.3 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2019 Edouard Griffiths, F4EXB //
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// //
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// This program is free software; you can redistribute it and/or modify //
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// it under the terms of the GNU General Public License as published by //
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// the Free Software Foundation as version 3 of the License, or //
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// (at your option) any later version. //
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// //
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// This program is distributed in the hope that it will be useful, //
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// but WITHOUT ANY WARRANTY; without even the implied warranty of //
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
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// GNU General Public License V3 for more details. //
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// //
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// You should have received a copy of the GNU General Public License //
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// along with this program. If not, see <http://www.gnu.org/licenses/>. //
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///////////////////////////////////////////////////////////////////////////////////
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#ifndef INCLUDE_AMDEMODSINK_H
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#define INCLUDE_AMDEMODSINK_H
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#include <QVector>
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#include "dsp/channelsamplesink.h"
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#include "dsp/nco.h"
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#include "dsp/interpolator.h"
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#include "dsp/agc.h"
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#include "dsp/firfilter.h"
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#include "dsp/phaselockcomplex.h"
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#include "audio/audiofifo.h"
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#include "util/movingaverage.h"
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#include "util/doublebufferfifo.h"
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#include "amdemodsettings.h"
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class fftfilt;
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class ChannelAPI;
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class AMDemodSink : public ChannelSampleSink {
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public:
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AMDemodSink();
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~AMDemodSink();
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virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end);
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void applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force = false);
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void applySettings(const AMDemodSettings& settings, bool force = false);
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void applyAudioSampleRate(int sampleRate);
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int getAudioSampleRate() const { return m_audioSampleRate; }
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double getMagSq() const { return m_magsq; }
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bool getSquelchOpen() const { return m_squelchOpen; }
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bool getPllLocked() const { return m_settings.m_pll && m_pll.locked(); }
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Real getPllFrequency() const { return m_pll.getFreq(); }
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AudioFifo *getAudioFifo() { return &m_audioFifo; }
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void setChannel(ChannelAPI *channel) { m_channel = channel; }
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void getMagSqLevels(double& avg, double& peak, int& nbSamples)
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{
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if (m_magsqCount > 0)
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{
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m_magsq = m_magsqSum / m_magsqCount;
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m_magSqLevelStore.m_magsq = m_magsq;
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m_magSqLevelStore.m_magsqPeak = m_magsqPeak;
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}
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avg = m_magSqLevelStore.m_magsq;
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peak = m_magSqLevelStore.m_magsqPeak;
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nbSamples = m_magsqCount == 0 ? 1 : m_magsqCount;
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m_magsqSum = 0.0f;
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m_magsqPeak = 0.0f;
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m_magsqCount = 0;
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}
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private:
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struct MagSqLevelsStore
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{
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MagSqLevelsStore() :
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m_magsq(1e-12),
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m_magsqPeak(1e-12)
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{}
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double m_magsq;
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double m_magsqPeak;
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};
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enum RateState {
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RSInitialFill,
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RSRunning
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};
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int m_channelSampleRate;
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int m_channelFrequencyOffset;
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AMDemodSettings m_settings;
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ChannelAPI *m_channel;
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int m_audioSampleRate;
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NCO m_nco;
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Interpolator m_interpolator;
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Real m_interpolatorDistance;
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Real m_interpolatorDistanceRemain;
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Real m_squelchLevel;
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int m_squelchCount;
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bool m_squelchOpen;
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DoubleBufferFIFO<Real> m_squelchDelayLine;
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double m_magsq;
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double m_magsqSum;
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double m_magsqPeak;
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int m_magsqCount;
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MagSqLevelsStore m_magSqLevelStore;
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MovingAverageUtil<Real, double, 16> m_movingAverage;
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SimpleAGC<4800> m_volumeAGC;
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Bandpass<Real> m_bandpass;
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Lowpass<Real> m_lowpass;
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Lowpass<std::complex<float> > m_pllFilt;
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PhaseLockComplex m_pll;
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fftfilt* DSBFilter;
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fftfilt* SSBFilter;
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Real m_syncAMBuff[2*1024];
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uint32_t m_syncAMBuffIndex;
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MagAGC m_syncAMAGC;
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AudioVector m_audioBuffer;
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AudioFifo m_audioFifo;
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uint32_t m_audioBufferFill;
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QVector<qint16> m_demodBuffer;
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int m_demodBufferFill;
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void processOneSample(Complex &ci);
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};
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#endif // INCLUDE_AMDEMODSINK_H
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