mirror of
https://github.com/f4exb/sdrangel.git
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130 lines
4.3 KiB
C
130 lines
4.3 KiB
C
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///////////////////////////////////////////////////////////////////////////////////
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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_SSBDEMODSINK_H
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#define INCLUDE_SSBDEMODSINK_H
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#include <vector>
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#include "dsp/channelsamplesink.h"
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#include "dsp/ncof.h"
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#include "dsp/interpolator.h"
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#include "dsp/fftfilt.h"
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#include "dsp/agc.h"
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#include "audio/audiofifo.h"
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#include "util/doublebufferfifo.h"
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#include "ssbdemodsettings.h"
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class BasebandSampleSink;
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class SSBDemodSink : public ChannelSampleSink {
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public:
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SSBDemodSink();
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~SSBDemodSink();
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virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end);
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void setSpectrumSink(BasebandSampleSink* spectrumSink) { m_spectrumSink = spectrumSink; }
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void applyChannelSettings(int inputSampleRate, int inputFrequencyOffset, bool force = false);
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void applySettings(const SSBDemodSettings& settings, bool force = false);
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void applyAudioSampleRate(int sampleRate);
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AudioFifo *getAudioFifo() { return &m_audioFifo; }
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double getMagSq() const { return m_magsq; }
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bool getAudioActive() const { return m_audioActive; }
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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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SSBDemodSettings m_settings;
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Real m_Bandwidth;
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Real m_LowCutoff;
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Real m_volume;
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int m_spanLog2;
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fftfilt::cmplx m_sum;
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int m_undersampleCount;
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int m_channelSampleRate;
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int m_channelFrequencyOffset;
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bool m_audioBinaual;
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bool m_audioFlipChannels;
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bool m_usb;
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bool m_dsb;
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bool m_audioMute;
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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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MagAGC m_agc;
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bool m_agcActive;
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bool m_agcClamping;
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int m_agcNbSamples; //!< number of audio (48 kHz) samples for AGC averaging
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double m_agcPowerThreshold; //!< AGC power threshold (linear)
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int m_agcThresholdGate; //!< Gate length in number of samples befor threshold triggers
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DoubleBufferFIFO<fftfilt::cmplx> m_squelchDelayLine;
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bool m_audioActive; //!< True if an audio signal is produced (no AGC or AGC and above threshold)
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NCOF 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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fftfilt* SSBFilter;
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fftfilt* DSBFilter;
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BasebandSampleSink* m_spectrumSink;
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SampleVector m_sampleBuffer;
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AudioVector m_audioBuffer;
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uint m_audioBufferFill;
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AudioFifo m_audioFifo;
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quint32 m_audioSampleRate;
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static const int m_ssbFftLen;
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static const int m_agcTarget;
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void processOneSample(Complex &ci);
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};
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#endif // INCLUDE_SSBDEMODSINK_H
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