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https://github.com/f4exb/sdrangel.git
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289 lines
9.2 KiB
C++
289 lines
9.2 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2012 maintech GmbH, Otto-Hahn-Str. 15, 97204 Hoechberg, Germany //
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// written by Christian Daniel //
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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 INCLUDE_SSBDEMOD_H
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#define INCLUDE_SSBDEMOD_H
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#include <QMutex>
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#include <vector>
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#include "dsp/basebandsamplesink.h"
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#include "channel/channelsinkapi.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/message.h"
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#include "util/doublebufferfifo.h"
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#include "ssbdemodsettings.h"
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#define ssbFftLen 1024
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#define agcTarget 3276.8 // -10 dB amplitude => -20 dB power: center of normal signal
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class DeviceSourceAPI;
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class ThreadedBasebandSampleSink;
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class DownChannelizer;
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class SSBDemod : public BasebandSampleSink, public ChannelSinkAPI {
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public:
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class MsgConfigureSSBDemod : public Message {
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MESSAGE_CLASS_DECLARATION
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public:
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const SSBDemodSettings& getSettings() const { return m_settings; }
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bool getForce() const { return m_force; }
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static MsgConfigureSSBDemod* create(const SSBDemodSettings& settings, bool force)
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{
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return new MsgConfigureSSBDemod(settings, force);
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}
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private:
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SSBDemodSettings m_settings;
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bool m_force;
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MsgConfigureSSBDemod(const SSBDemodSettings& settings, bool force) :
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Message(),
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m_settings(settings),
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m_force(force)
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{ }
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};
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class MsgConfigureChannelizer : 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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int getCenterFrequency() const { return m_centerFrequency; }
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static MsgConfigureChannelizer* create(int sampleRate, int centerFrequency)
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{
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return new MsgConfigureChannelizer(sampleRate, centerFrequency);
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}
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private:
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int m_sampleRate;
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int m_centerFrequency;
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MsgConfigureChannelizer(int sampleRate, int centerFrequency) :
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Message(),
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m_sampleRate(sampleRate),
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m_centerFrequency(centerFrequency)
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{ }
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};
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SSBDemod(DeviceSourceAPI *deviceAPI);
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virtual ~SSBDemod();
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virtual void destroy() { delete this; }
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void setSampleSink(BasebandSampleSink* sampleSink) { m_sampleSink = sampleSink; }
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void configure(MessageQueue* messageQueue,
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Real Bandwidth,
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Real LowCutoff,
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Real volume,
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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 agc,
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bool agcClamping,
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int agcTimeLog2,
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int agcPowerThreshold,
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int agcThresholdGate);
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virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end, bool positiveOnly);
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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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virtual void getIdentifier(QString& id) { id = objectName(); }
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virtual void getTitle(QString& title) { title = m_settings.m_title; }
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virtual qint64 getCenterFrequency() const { return m_settings.m_inputFrequencyOffset; }
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virtual QByteArray serialize() const;
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virtual bool deserialize(const QByteArray& data);
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uint32_t getAudioSampleRate() const { return m_audioSampleRate; }
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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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avg = m_magsqCount == 0 ? 1e-10 : m_magsqSum / m_magsqCount;
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m_magsq = avg;
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peak = m_magsqPeak == 0.0 ? 1e-10 : 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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static const QString m_channelIdURI;
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static const QString m_channelId;
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private:
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class MsgConfigureSSBDemodPrivate : public Message {
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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 getLoCutoff() const { return m_LowCutoff; }
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Real getVolume() const { return m_volume; }
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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 getAGC() const { return m_agc; }
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bool getAGCClamping() const { return m_agcClamping; }
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int getAGCTimeLog2() const { return m_agcTimeLog2; }
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int getAGCPowerThershold() const { return m_agcPowerThreshold; }
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int getAGCThersholdGate() const { return m_agcThresholdGate; }
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static MsgConfigureSSBDemodPrivate* create(Real Bandwidth,
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Real LowCutoff,
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Real volume,
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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 agc,
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bool agcClamping,
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int agcTimeLog2,
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int agcPowerThreshold,
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int agcThresholdGate)
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{
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return new MsgConfigureSSBDemodPrivate(
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Bandwidth,
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LowCutoff,
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volume,
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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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agc,
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agcClamping,
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agcTimeLog2,
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agcPowerThreshold,
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agcThresholdGate);
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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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Real m_volume;
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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_agc;
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bool m_agcClamping;
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int m_agcTimeLog2;
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int m_agcPowerThreshold;
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int m_agcThresholdGate;
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MsgConfigureSSBDemodPrivate(Real Bandwidth,
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Real LowCutoff,
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Real volume,
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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 agc,
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bool agcClamping,
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int agcTimeLog2,
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int agcPowerThreshold,
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int agcThresholdGate) :
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Message(),
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m_Bandwidth(Bandwidth),
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m_LowCutoff(LowCutoff),
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m_volume(volume),
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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_agc(agc),
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m_agcClamping(agcClamping),
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m_agcTimeLog2(agcTimeLog2),
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m_agcPowerThreshold(agcPowerThreshold),
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m_agcThresholdGate(agcThresholdGate)
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{ }
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};
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DeviceSourceAPI *m_deviceAPI;
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ThreadedBasebandSampleSink* m_threadedChannelizer;
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DownChannelizer* m_channelizer;
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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_inputSampleRate;
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int m_inputFrequencyOffset;
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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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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_sampleSink;
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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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QMutex m_settingsMutex;
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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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};
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#endif // INCLUDE_SSBDEMOD_H
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