/////////////////////////////////////////////////////////////////////////////////// // Copyright (C) 2019-2020, 2022-2023 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 . // /////////////////////////////////////////////////////////////////////////////////// #ifndef INCLUDE_SSBDEMODSINK_H #define INCLUDE_SSBDEMODSINK_H #include #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 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; double m_magsqCur; double m_magsqPrev; 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 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; SpectrumVis* m_spectrumSink; SampleVector m_sampleBuffer; AudioVector m_audioBuffer; std::size_t m_audioBufferFill; AudioFifo m_audioFifo; quint32 m_audioSampleRate; QVector m_demodBuffer; int m_demodBufferFill; static const int m_ssbFftLen; static const int m_agcTarget; void processOneSample(Complex &ci); }; #endif // INCLUDE_SSBDEMODSINK_H