/////////////////////////////////////////////////////////////////////////////////// // Copyright (C) 2015 Edouard Griffiths, F4EXB. // // // // 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_AMDEMOD_H #define INCLUDE_AMDEMOD_H #include #include #include "dsp/basebandsamplesink.h" #include "channel/channelsinkapi.h" #include "dsp/nco.h" #include "dsp/interpolator.h" #include "util/movingaverage.h" #include "dsp/agc.h" #include "dsp/bandpass.h" #include "dsp/lowpass.h" #include "dsp/phaselockcomplex.h" #include "dsp/fftfilt.h" #include "audio/audiofifo.h" #include "util/message.h" #include "util/doublebufferfifo.h" #include "amdemodsettings.h" class DeviceSourceAPI; class DownChannelizer; class ThreadedBasebandSampleSink; class AMDemod : public BasebandSampleSink, public ChannelSinkAPI { Q_OBJECT public: class MsgConfigureAMDemod : public Message { MESSAGE_CLASS_DECLARATION public: const AMDemodSettings& getSettings() const { return m_settings; } bool getForce() const { return m_force; } static MsgConfigureAMDemod* create(const AMDemodSettings& settings, bool force) { return new MsgConfigureAMDemod(settings, force); } private: AMDemodSettings m_settings; bool m_force; MsgConfigureAMDemod(const AMDemodSettings& 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) { } }; AMDemod(DeviceSourceAPI *deviceAPI); ~AMDemod(); virtual void destroy() { delete this; } virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end, bool po); 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); virtual int webapiSettingsGet( SWGSDRangel::SWGChannelSettings& response, QString& errorMessage); virtual int webapiSettingsPutPatch( bool force, const QStringList& channelSettingsKeys, SWGSDRangel::SWGChannelSettings& response, QString& errorMessage); virtual int webapiReportGet( SWGSDRangel::SWGChannelReport& response, QString& errorMessage); uint32_t getAudioSampleRate() const { return m_audioSampleRate; } double getMagSq() const { return m_magsq; } bool getSquelchOpen() const { return m_squelchOpen; } bool getPllLocked() const { return m_settings.m_pll && m_pll.locked(); } void getMagSqLevels(double& avg, double& peak, int& nbSamples) { avg = m_magsqCount == 0 ? 1e-10 : m_magsqSum / m_magsqCount; 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: enum RateState { RSInitialFill, RSRunning }; DeviceSourceAPI *m_deviceAPI; ThreadedBasebandSampleSink* m_threadedChannelizer; DownChannelizer* m_channelizer; int m_inputSampleRate; int m_inputFrequencyOffset; AMDemodSettings m_settings; uint32_t m_audioSampleRate; bool m_running; NCO m_nco; Interpolator m_interpolator; Real m_interpolatorDistance; Real m_interpolatorDistanceRemain; Real m_squelchLevel; uint32_t m_squelchCount; bool m_squelchOpen; DoubleBufferFIFO m_squelchDelayLine; double m_magsq; double m_magsqSum; double m_magsqPeak; int m_magsqCount; MovingAverageUtil m_movingAverage; SimpleAGC<4096> m_volumeAGC; Bandpass m_bandpass; Lowpass > m_pllFilt; PhaseLockComplex m_pll; fftfilt* DSBFilter; Real m_syncAMBuff[2*1024]; uint32_t m_syncAMBuffIndex; AudioVector m_audioBuffer; uint32_t m_audioBufferFill; AudioFifo m_audioFifo; static const int m_udpBlockSize; QMutex m_settingsMutex; void applyChannelSettings(int inputSampleRate, int inputFrequencyOffset, bool force = false); void applySettings(const AMDemodSettings& settings, bool force = false); void applyAudioSampleRate(int sampleRate); void webapiFormatChannelSettings(SWGSDRangel::SWGChannelSettings& response, const AMDemodSettings& settings); void webapiFormatChannelReport(SWGSDRangel::SWGChannelReport& response); void processOneSample(Complex &ci) { Real re = ci.real() / SDR_RX_SCALEF; Real im = ci.imag() / SDR_RX_SCALEF; Real magsq = re*re + im*im; m_movingAverage(magsq); m_magsq = m_movingAverage.asDouble(); m_magsqSum += magsq; if (magsq > m_magsqPeak) { m_magsqPeak = magsq; } m_magsqCount++; m_squelchDelayLine.write(magsq); if (m_magsq < m_squelchLevel) { if (m_squelchCount > 0) { m_squelchCount--; } } else { if (m_squelchCount < m_audioSampleRate / 10) { m_squelchCount++; } } qint16 sample; m_squelchOpen = (m_squelchCount >= m_audioSampleRate / 20); if (m_squelchOpen && !m_settings.m_audioMute) { Real demod; if (m_settings.m_pll) { std::complex s(re, im); s = m_pllFilt.filter(s); m_pll.feed(s.real(), s.imag()); float yr = re * m_pll.getImag() - im * m_pll.getReal(); float yi = re * m_pll.getReal() + im * m_pll.getImag(); fftfilt::cmplx *sideband; std::complex cs(yr, yi); int n_out = DSBFilter->runDSB(cs, &sideband, false); for (int i = 0; i < n_out; i++) { m_syncAMBuff[i] = (sideband[i].real() + sideband[i].imag()); m_syncAMBuffIndex = 0; } m_syncAMBuffIndex = m_syncAMBuffIndex < 2*1024 ? m_syncAMBuffIndex : 0; demod = m_syncAMBuff[m_syncAMBuffIndex++]*0.7*(SDR_RX_SCALEF/602.0f); m_volumeAGC.feed(demod); demod /= (10.0*m_volumeAGC.getValue()); } else { demod = sqrt(m_squelchDelayLine.readBack(m_audioSampleRate/20)); m_volumeAGC.feed(demod); demod = (demod - m_volumeAGC.getValue()) / m_volumeAGC.getValue(); } if (m_settings.m_bandpassEnable) { demod = m_bandpass.filter(demod); demod /= 301.0f; } Real attack = (m_squelchCount - 0.05f * m_audioSampleRate) / (0.05f * m_audioSampleRate); sample = demod * attack * (m_audioSampleRate/24) * m_settings.m_volume; } else { sample = 0; } m_audioBuffer[m_audioBufferFill].l = sample; m_audioBuffer[m_audioBufferFill].r = sample; ++m_audioBufferFill; if (m_audioBufferFill >= m_audioBuffer.size()) { uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 10); if (res != m_audioBufferFill) { qDebug("AMDemod::processOneSample: %u/%u audio samples written", res, m_audioBufferFill); m_audioFifo.clear(); } m_audioBufferFill = 0; } } }; #endif // INCLUDE_AMDEMOD_H