/////////////////////////////////////////////////////////////////////////////////// // Copyright (C) 2012 maintech GmbH, Otto-Hahn-Str. 15, 97204 Hoechberg, Germany // // written by Christian Daniel // // // // 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 // // // // 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_SSBDEMOD_H #define INCLUDE_SSBDEMOD_H #include #include #include "dsp/basebandsamplesink.h" #include "channel/channelsinkapi.h" #include "dsp/ncof.h" #include "dsp/interpolator.h" #include "dsp/fftfilt.h" #include "dsp/agc.h" #include "audio/audiofifo.h" #include "util/message.h" #include "ssbdemodsettings.h" #define ssbFftLen 1024 #define agcTarget 3276.8 // -10 dB amplitude => -20 dB power: center of normal signal class DeviceSourceAPI; class ThreadedBasebandSampleSink; class DownChannelizer; class AudioNetSink; class SSBDemod : public BasebandSampleSink, public ChannelSinkAPI { public: class MsgConfigureSSBDemod : public Message { MESSAGE_CLASS_DECLARATION public: const SSBDemodSettings& getSettings() const { return m_settings; } bool getForce() const { return m_force; } static MsgConfigureSSBDemod* create(const SSBDemodSettings& settings, bool force) { return new MsgConfigureSSBDemod(settings, force); } private: SSBDemodSettings m_settings; bool m_force; MsgConfigureSSBDemod(const SSBDemodSettings& settings, bool force) : Message(), m_settings(settings), m_force(force) { } }; class MsgConfigureChannelizer : public Message { MESSAGE_CLASS_DECLARATION public: int getSampleRate() const { return m_sampleRate; } int getCenterFrequency() const { return m_centerFrequency; } static MsgConfigureChannelizer* create(int sampleRate, int centerFrequency) { return new MsgConfigureChannelizer(sampleRate, centerFrequency); } private: int m_sampleRate; int m_centerFrequency; MsgConfigureChannelizer(int sampleRate, int centerFrequency) : Message(), m_sampleRate(sampleRate), m_centerFrequency(centerFrequency) { } }; SSBDemod(DeviceSourceAPI *deviceAPI); virtual ~SSBDemod(); virtual void destroy() { delete this; } void setSampleSink(BasebandSampleSink* sampleSink) { m_sampleSink = sampleSink; } void configure(MessageQueue* messageQueue, Real Bandwidth, Real LowCutoff, Real volume, int spanLog2, bool audioBinaural, bool audioFlipChannels, bool dsb, bool audioMute, bool agc, bool agcClamping, int agcTimeLog2, int agcPowerThreshold, int agcThresholdGate); virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end, bool positiveOnly); virtual void start(); virtual void stop(); virtual bool handleMessage(const Message& cmd); virtual void getIdentifier(QString& id) { id = objectName(); } virtual void getTitle(QString& title) { title = m_settings.m_title; } virtual qint64 getCenterFrequency() const { return m_settings.m_inputFrequencyOffset; } virtual QByteArray serialize() const; virtual bool deserialize(const QByteArray& data); uint32_t getAudioSampleRate() const { return m_audioSampleRate; } double getMagSq() const { return m_magsq; } bool getAudioActive() const { return m_audioActive; } void getMagSqLevels(double& avg, double& peak, int& nbSamples) { avg = m_magsqCount == 0 ? 1e-10 : m_magsqSum / m_magsqCount; m_magsq = avg; peak = m_magsqPeak == 0.0 ? 1e-10 : m_magsqPeak; nbSamples = m_magsqCount == 0 ? 1 : m_magsqCount; m_magsqSum = 0.0f; m_magsqPeak = 0.0f; m_magsqCount = 0; } static const QString m_channelIdURI; static const QString m_channelId; private: class MsgConfigureSSBDemodPrivate : public Message { MESSAGE_CLASS_DECLARATION public: Real getBandwidth() const { return m_Bandwidth; } Real getLoCutoff() const { return m_LowCutoff; } Real getVolume() const { return m_volume; } int getSpanLog2() const { return m_spanLog2; } bool getAudioBinaural() const { return m_audioBinaural; } bool getAudioFlipChannels() const { return m_audioFlipChannels; } bool getDSB() const { return m_dsb; } bool getAudioMute() const { return m_audioMute; } bool getAGC() const { return m_agc; } bool getAGCClamping() const { return m_agcClamping; } int getAGCTimeLog2() const { return m_agcTimeLog2; } int getAGCPowerThershold() const { return m_agcPowerThreshold; } int getAGCThersholdGate() const { return m_agcThresholdGate; } static MsgConfigureSSBDemodPrivate* create(Real Bandwidth, Real LowCutoff, Real volume, int spanLog2, bool audioBinaural, bool audioFlipChannels, bool dsb, bool audioMute, bool agc, bool agcClamping, int agcTimeLog2, int agcPowerThreshold, int agcThresholdGate) { return new MsgConfigureSSBDemodPrivate( Bandwidth, LowCutoff, volume, spanLog2, audioBinaural, audioFlipChannels, dsb, audioMute, agc, agcClamping, agcTimeLog2, agcPowerThreshold, agcThresholdGate); } private: Real m_Bandwidth; Real m_LowCutoff; Real m_volume; int m_spanLog2; bool m_audioBinaural; bool m_audioFlipChannels; bool m_dsb; bool m_audioMute; bool m_agc; bool m_agcClamping; int m_agcTimeLog2; int m_agcPowerThreshold; int m_agcThresholdGate; MsgConfigureSSBDemodPrivate(Real Bandwidth, Real LowCutoff, Real volume, int spanLog2, bool audioBinaural, bool audioFlipChannels, bool dsb, bool audioMute, bool agc, bool agcClamping, int agcTimeLog2, int agcPowerThreshold, int agcThresholdGate) : Message(), m_Bandwidth(Bandwidth), m_LowCutoff(LowCutoff), m_volume(volume), m_spanLog2(spanLog2), m_audioBinaural(audioBinaural), m_audioFlipChannels(audioFlipChannels), m_dsb(dsb), m_audioMute(audioMute), m_agc(agc), m_agcClamping(agcClamping), m_agcTimeLog2(agcTimeLog2), m_agcPowerThreshold(agcPowerThreshold), m_agcThresholdGate(agcThresholdGate) { } }; DeviceSourceAPI *m_deviceAPI; ThreadedBasebandSampleSink* m_threadedChannelizer; DownChannelizer* m_channelizer; SSBDemodSettings m_settings; Real m_Bandwidth; Real m_LowCutoff; Real m_volume; int m_spanLog2; fftfilt::cmplx m_sum; int m_undersampleCount; int m_inputSampleRate; int m_inputFrequencyOffset; 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; 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 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; BasebandSampleSink* m_sampleSink; SampleVector m_sampleBuffer; AudioVector m_audioBuffer; uint m_audioBufferFill; AudioFifo m_audioFifo; quint32 m_audioSampleRate; QMutex m_settingsMutex; void applyChannelSettings(int inputSampleRate, int inputFrequencyOffset, bool force = false); void applySettings(const SSBDemodSettings& settings, bool force = false); void applyAudioSampleRate(int sampleRate); }; #endif // INCLUDE_SSBDEMOD_H