mirror of
https://github.com/f4exb/sdrangel.git
synced 2024-11-23 08:28:36 -05:00
460 lines
12 KiB
C++
460 lines
12 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2016 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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// //
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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 PLUGINS_CHANNELTX_MODSSB_SSBMOD_H_
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#define PLUGINS_CHANNELTX_MODSSB_SSBMOD_H_
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#include <QMutex>
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#include <vector>
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#include <iostream>
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#include <fstream>
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#include "dsp/basebandsamplesource.h"
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#include "dsp/basebandsamplesink.h"
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#include "dsp/ncof.h"
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#include "dsp/interpolator.h"
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#include "dsp/movingaverage.h"
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#include "dsp/agc.h"
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#include "dsp/fftfilt.h"
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#include "dsp/cwkeyer.h"
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#include "audio/audiofifo.h"
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#include "util/message.h"
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class SSBMod : public BasebandSampleSource {
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Q_OBJECT
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public:
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typedef enum
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{
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SSBModInputNone,
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SSBModInputTone,
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SSBModInputFile,
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SSBModInputAudio,
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SSBModInputCWTone
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} SSBModInputAF;
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class MsgConfigureFileSourceName : public Message
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{
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MESSAGE_CLASS_DECLARATION
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public:
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const QString& getFileName() const { return m_fileName; }
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static MsgConfigureFileSourceName* create(const QString& fileName)
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{
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return new MsgConfigureFileSourceName(fileName);
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}
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private:
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QString m_fileName;
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MsgConfigureFileSourceName(const QString& fileName) :
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Message(),
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m_fileName(fileName)
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{ }
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};
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class MsgConfigureFileSourceSeek : public Message
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{
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MESSAGE_CLASS_DECLARATION
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public:
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int getPercentage() const { return m_seekPercentage; }
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static MsgConfigureFileSourceSeek* create(int seekPercentage)
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{
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return new MsgConfigureFileSourceSeek(seekPercentage);
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}
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protected:
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int m_seekPercentage; //!< percentage of seek position from the beginning 0..100
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MsgConfigureFileSourceSeek(int seekPercentage) :
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Message(),
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m_seekPercentage(seekPercentage)
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{ }
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};
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class MsgConfigureFileSourceStreamTiming : public Message {
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MESSAGE_CLASS_DECLARATION
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public:
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static MsgConfigureFileSourceStreamTiming* create()
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{
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return new MsgConfigureFileSourceStreamTiming();
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}
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private:
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MsgConfigureFileSourceStreamTiming() :
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Message()
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{ }
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};
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class MsgConfigureAFInput : public Message
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{
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MESSAGE_CLASS_DECLARATION
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public:
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SSBModInputAF getAFInput() const { return m_afInput; }
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static MsgConfigureAFInput* create(SSBModInputAF afInput)
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{
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return new MsgConfigureAFInput(afInput);
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}
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private:
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SSBModInputAF m_afInput;
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MsgConfigureAFInput(SSBModInputAF afInput) :
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Message(),
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m_afInput(afInput)
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{ }
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};
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class MsgReportFileSourceStreamTiming : public Message
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{
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MESSAGE_CLASS_DECLARATION
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public:
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std::size_t getSamplesCount() const { return m_samplesCount; }
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static MsgReportFileSourceStreamTiming* create(std::size_t samplesCount)
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{
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return new MsgReportFileSourceStreamTiming(samplesCount);
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}
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protected:
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std::size_t m_samplesCount;
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MsgReportFileSourceStreamTiming(std::size_t samplesCount) :
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Message(),
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m_samplesCount(samplesCount)
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{ }
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};
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class MsgReportFileSourceStreamData : public Message {
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MESSAGE_CLASS_DECLARATION
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public:
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int getSampleRate() const { return m_sampleRate; }
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quint32 getRecordLength() const { return m_recordLength; }
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static MsgReportFileSourceStreamData* create(int sampleRate,
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quint32 recordLength)
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{
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return new MsgReportFileSourceStreamData(sampleRate, recordLength);
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}
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protected:
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int m_sampleRate;
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quint32 m_recordLength;
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MsgReportFileSourceStreamData(int sampleRate,
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quint32 recordLength) :
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Message(),
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m_sampleRate(sampleRate),
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m_recordLength(recordLength)
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{ }
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};
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//=================================================================
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SSBMod(BasebandSampleSink* sampleSink);
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~SSBMod();
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void configure(MessageQueue* messageQueue,
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Real bandwidth,
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Real lowCutoff,
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float toneFrequency,
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float volumeFactor,
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int spanLog2,
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bool audioBinaural,
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bool audioFlipChannels,
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bool dsb,
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bool audioMute,
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bool playLoop,
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bool agc,
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float agcOrder,
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int agcTime,
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int agcThreshold,
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int agcThresholdGate,
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int agcThresholdDelay);
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virtual void pull(Sample& sample);
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virtual void pullAudio(int nbSamples);
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virtual void start();
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virtual void stop();
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virtual bool handleMessage(const Message& cmd);
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double getMagSq() const { return m_magsq; }
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CWKeyer *getCWKeyer() { return &m_cwKeyer; }
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signals:
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/**
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* Level changed
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* \param rmsLevel RMS level in range 0.0 - 1.0
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* \param peakLevel Peak level in range 0.0 - 1.0
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* \param numSamples Number of audio samples analyzed
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*/
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void levelChanged(qreal rmsLevel, qreal peakLevel, int numSamples);
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private:
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class MsgConfigureSSBMod : public Message
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{
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MESSAGE_CLASS_DECLARATION
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public:
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Real getBandwidth() const { return m_bandwidth; }
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Real getLowCutoff() const { return m_lowCutoff; }
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float getToneFrequency() const { return m_toneFrequency; }
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float getVolumeFactor() const { return m_volumeFactor; }
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int getSpanLog2() const { return m_spanLog2; }
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bool getAudioBinaural() const { return m_audioBinaural; }
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bool getAudioFlipChannels() const { return m_audioFlipChannels; }
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bool getDSB() const { return m_dsb; }
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bool getAudioMute() const { return m_audioMute; }
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bool getPlayLoop() const { return m_playLoop; }
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bool getAGC() const { return m_agc; }
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float getAGCOrder() const { return m_agcOrder; }
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int getAGCTime() const { return m_agcTime; }
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int getAGCThreshold() const { return m_agcThreshold; }
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int getAGCThresholdGate() const { return m_agcThresholdGate; }
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int getAGCThresholdDelay() const { return m_agcThresholdDelay; }
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static MsgConfigureSSBMod* create(Real bandwidth,
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Real lowCutoff,
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float toneFrequency,
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float volumeFactor,
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int spanLog2,
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bool audioBinaural,
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bool audioFlipChannels,
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bool dsb,
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bool audioMute,
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bool playLoop,
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bool agc,
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float agcOrder,
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int agcTime,
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int agcThreshold,
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int agcThresholdGate,
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int agcThresholdDelay)
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{
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return new MsgConfigureSSBMod(bandwidth,
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lowCutoff,
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toneFrequency,
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volumeFactor,
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spanLog2,
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audioBinaural,
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audioFlipChannels,
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dsb,
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audioMute,
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playLoop,
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agc,
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agcOrder,
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agcTime,
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agcThreshold,
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agcThresholdGate,
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agcThresholdDelay);
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}
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private:
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Real m_bandwidth;
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Real m_lowCutoff;
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float m_toneFrequency;
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float m_volumeFactor;
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int m_spanLog2;
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bool m_audioBinaural;
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bool m_audioFlipChannels;
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bool m_dsb;
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bool m_audioMute;
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bool m_playLoop;
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bool m_agc;
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float m_agcOrder;
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int m_agcTime;
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int m_agcThreshold;
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int m_agcThresholdGate;
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int m_agcThresholdDelay;
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MsgConfigureSSBMod(Real bandwidth,
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Real lowCutoff,
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float toneFrequency,
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float volumeFactor,
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int spanLog2,
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bool audioBinaural,
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bool audioFlipChannels,
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bool dsb,
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bool audioMute,
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bool playLoop,
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bool agc,
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float agcOrder,
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int agcTime,
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int agcThreshold,
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int agcThresholdGate,
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int agcThresholdDelay) :
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Message(),
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m_bandwidth(bandwidth),
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m_lowCutoff(lowCutoff),
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m_toneFrequency(toneFrequency),
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m_volumeFactor(volumeFactor),
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m_spanLog2(spanLog2),
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m_audioBinaural(audioBinaural),
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m_audioFlipChannels(audioFlipChannels),
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m_dsb(dsb),
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m_audioMute(audioMute),
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m_playLoop(playLoop),
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m_agc(agc),
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m_agcOrder(agcOrder),
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m_agcTime(agcTime),
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m_agcThreshold(agcThreshold),
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m_agcThresholdGate(agcThresholdGate),
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m_agcThresholdDelay(agcThresholdDelay)
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{ }
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};
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//=================================================================
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struct AudioSample {
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qint16 l;
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qint16 r;
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};
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typedef std::vector<AudioSample> AudioVector;
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enum RateState {
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RSInitialFill,
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RSRunning
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};
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struct Config {
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int m_basebandSampleRate;
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int m_outputSampleRate;
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qint64 m_inputFrequencyOffset;
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Real m_bandwidth;
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Real m_lowCutoff;
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bool m_usb;
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float m_toneFrequency;
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float m_volumeFactor;
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quint32 m_audioSampleRate;
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int m_spanLog2;
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bool m_audioBinaural;
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bool m_audioFlipChannels;
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bool m_dsb;
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bool m_audioMute;
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bool m_playLoop;
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bool m_agc;
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float m_agcOrder;
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int m_agcTime;
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bool m_agcThresholdEnable;
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double m_agcThreshold;
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int m_agcThresholdGate;
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int m_agcThresholdDelay;
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Config() :
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m_basebandSampleRate(0),
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m_outputSampleRate(0),
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m_inputFrequencyOffset(0),
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m_bandwidth(3000.0f),
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m_lowCutoff(300.0f),
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m_usb(true),
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m_toneFrequency(1000.0f),
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m_volumeFactor(1.0f),
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m_audioSampleRate(0),
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m_spanLog2(3),
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m_audioBinaural(false),
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m_audioFlipChannels(false),
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m_dsb(false),
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m_audioMute(false),
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m_playLoop(false),
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m_agc(false),
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m_agcOrder(0.2),
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m_agcTime(9600),
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m_agcThresholdEnable(true),
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m_agcThreshold(1e-4),
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m_agcThresholdGate(192),
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m_agcThresholdDelay(2400)
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{ }
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};
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//=================================================================
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Config m_config;
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Config m_running;
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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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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_sampleSink;
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SampleVector m_sampleBuffer;
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// Real m_magsqSpectrum;
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// Real m_magsqSum;
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// Real m_magsqPeak;
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// int m_magsqCount;
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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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MovingAverage<double> m_movingAverage;
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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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QMutex m_settingsMutex;
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std::ifstream m_ifstream;
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QString m_fileName;
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quint64 m_fileSize; //!< raw file size (bytes)
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quint32 m_recordLength; //!< record length in seconds computed from file size
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int m_sampleRate;
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SSBModInputAF m_afInput;
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quint32 m_levelCalcCount;
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Real m_peakLevel;
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Real m_levelSum;
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CWKeyer m_cwKeyer;
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CWSmoother m_cwSmoother;
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MagAGC m_inAGC;
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static const int m_levelNbSamples;
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void apply();
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void pullAF(Complex& sample);
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void calculateLevel(Complex& sample);
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void modulateSample();
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void openFileStream();
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void seekFileStream(int seekPercentage);
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};
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#endif /* PLUGINS_CHANNELTX_MODSSB_SSBMOD_H_ */
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