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138 lines
4.7 KiB
C++
138 lines
4.7 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_SSBMODSOURCE_H
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#define INCLUDE_SSBMODSOURCE_H
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#include <QMutex>
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#include <iostream>
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#include <fstream>
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#include "dsp/channelsamplesource.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/cwkeyer.h"
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#include "util/movingaverage.h"
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#include "audio/audiocompressorsnd.h"
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#include "audio/audiofifo.h"
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#include "ssbmodsettings.h"
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class BasebandSampleSink;
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class SSBModSource : public ChannelSampleSource
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{
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public:
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SSBModSource();
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virtual ~SSBModSource();
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virtual void pull(SampleVector::iterator begin, unsigned int nbSamples);
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virtual void pullOne(Sample& sample);
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virtual void prefetch(unsigned int nbSamples);
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void setInputFileStream(std::ifstream *ifstream) { m_ifstream = ifstream; }
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AudioFifo *getAudioFifo() { return &m_audioFifo; }
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AudioFifo *getFeedbackAudioFifo() { return &m_feedbackAudioFifo; }
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void applyAudioSampleRate(unsigned int sampleRate);
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void applyFeedbackAudioSampleRate(unsigned int sampleRate);
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unsigned int getAudioSampleRate() const { return m_audioSampleRate; }
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unsigned int getFeedbackAudioSampleRate() const { return m_feedbackAudioSampleRate; }
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CWKeyer& getCWKeyer() { return m_cwKeyer; }
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double getMagSq() const { return m_magsq; }
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void getLevels(qreal& rmsLevel, qreal& peakLevel, int& numSamples) const
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{
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rmsLevel = m_rmsLevel;
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peakLevel = m_peakLevelOut;
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numSamples = m_levelNbSamples;
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}
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void applySettings(const SSBModSettings& settings, bool force = false);
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void applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force = 0);
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void setSpectrumSink(BasebandSampleSink *sampleSink) { m_spectrumSink = sampleSink; }
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private:
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int m_channelSampleRate;
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int m_channelFrequencyOffset;
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SSBModSettings m_settings;
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NCOF m_carrierNco;
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NCOF m_toneNco;
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Complex m_modSample;
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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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bool m_interpolatorConsumed;
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Interpolator m_feedbackInterpolator;
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Real m_feedbackInterpolatorDistance;
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Real m_feedbackInterpolatorDistanceRemain;
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bool m_feedbackInterpolatorConsumed;
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fftfilt* m_SSBFilter;
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fftfilt* m_DSBFilter;
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Complex* m_SSBFilterBuffer;
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Complex* m_DSBFilterBuffer;
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int m_SSBFilterBufferIndex;
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int m_DSBFilterBufferIndex;
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static const int m_ssbFftLen;
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BasebandSampleSink* m_spectrumSink;
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SampleVector m_sampleBuffer;
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fftfilt::cmplx m_sum;
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int m_undersampleCount;
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int m_sumCount;
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double m_magsq;
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MovingAverageUtil<double, double, 16> m_movingAverage;
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quint32 m_audioSampleRate;
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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_feedbackAudioSampleRate;
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AudioVector m_feedbackAudioBuffer;
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uint m_feedbackAudioBufferFill;
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AudioFifo m_feedbackAudioFifo;
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quint32 m_levelCalcCount;
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qreal m_rmsLevel;
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qreal m_peakLevelOut;
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Real m_peakLevel;
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Real m_levelSum;
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std::ifstream *m_ifstream;
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CWKeyer m_cwKeyer;
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AudioCompressorSnd m_audioCompressor;
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int m_agcStepLength;
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static const int m_levelNbSamples;
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void processOneSample(Complex& ci);
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void pullAF(Complex& sample);
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void pullAudio(unsigned int nbSamples);
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void pushFeedback(Complex sample);
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void calculateLevel(Complex& sample);
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void modulateSample();
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};
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#endif // INCLUDE_SSBMODSOURCE_H
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