/////////////////////////////////////////////////////////////////////////////////// // Copyright (C) 2019 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 // // (at your option) any later version. // // // // 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_AMDEMODSINK_H #define INCLUDE_AMDEMODSINK_H #include "dsp/channelsamplesink.h" #include "dsp/nco.h" #include "dsp/interpolator.h" #include "dsp/agc.h" #include "dsp/bandpass.h" #include "dsp/lowpass.h" #include "dsp/phaselockcomplex.h" #include "audio/audiofifo.h" #include "util/movingaverage.h" #include "util/doublebufferfifo.h" #include "amdemodsettings.h" class fftfilt; class AMDemodSink : public ChannelSampleSink { public: AMDemodSink(); ~AMDemodSink(); virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end); void applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force = false); void applySettings(const AMDemodSettings& settings, bool force = false); void applyAudioSampleRate(int sampleRate); 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(); } Real getPllFrequency() const { return m_pll.getFreq(); } AudioFifo *getAudioFifo() { return &m_audioFifo; } void getMagSqLevels(double& avg, double& peak, int& nbSamples) { if (m_magsqCount > 0) { m_magsq = m_magsqSum / m_magsqCount; m_magSqLevelStore.m_magsq = m_magsq; m_magSqLevelStore.m_magsqPeak = m_magsqPeak; } avg = m_magSqLevelStore.m_magsq; peak = m_magSqLevelStore.m_magsqPeak; nbSamples = m_magsqCount == 0 ? 1 : m_magsqCount; m_magsqSum = 0.0f; m_magsqPeak = 0.0f; m_magsqCount = 0; } private: struct MagSqLevelsStore { MagSqLevelsStore() : m_magsq(1e-12), m_magsqPeak(1e-12) {} double m_magsq; double m_magsqPeak; }; enum RateState { RSInitialFill, RSRunning }; int m_channelSampleRate; int m_channelFrequencyOffset; AMDemodSettings m_settings; uint32_t m_audioSampleRate; 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; MagSqLevelsStore m_magSqLevelStore; MovingAverageUtil m_movingAverage; SimpleAGC<4800> m_volumeAGC; Bandpass m_bandpass; Lowpass m_lowpass; Lowpass > m_pllFilt; PhaseLockComplex m_pll; fftfilt* DSBFilter; fftfilt* SSBFilter; Real m_syncAMBuff[2*1024]; uint32_t m_syncAMBuffIndex; MagAGC m_syncAMAGC; AudioVector m_audioBuffer; AudioFifo m_audioFifo; uint32_t m_audioBufferFill; void processOneSample(Complex &ci); }; #endif // INCLUDE_AMDEMODSINK_H