/////////////////////////////////////////////////////////////////////////////////// // Copyright (C) 2019 Edouard Griffiths, F4EXB // // Copyright (C) 2021 Jon Beniston, M7RCE // // // // 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_APTDEMODSINK_H #define INCLUDE_APTDEMODSINK_H #include "dsp/channelsamplesink.h" #include "dsp/phasediscri.h" #include "dsp/nco.h" #include "dsp/interpolator.h" #include "dsp/firfilter.h" #include "util/movingaverage.h" #include "util/doublebufferfifo.h" #include "util/messagequeue.h" #include "aptdemodsettings.h" #include #include #include #include // FIXME: Use lower sample rate for better SNR? // Do we want an audio filter? Subcarrier at 2800Hz. Does libaptdec have one? #define APTDEMOD_AUDIO_SAMPLE_RATE 48000 // Lines are 2 per second -> 4160 words per second class APTDemod; class APTDemodSink : public ChannelSampleSink { public: APTDemodSink(APTDemod *packetDemod); ~APTDemodSink(); 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 APTDemodSettings& settings, bool force = false); void setMessageQueueToChannel(MessageQueue *messageQueue) { m_messageQueueToChannel = messageQueue; } double getMagSq() const { return m_magsq; } 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; } int getSamples(float *samples, int count); void resetDecoder(); private: struct MagSqLevelsStore { MagSqLevelsStore() : m_magsq(1e-12), m_magsqPeak(1e-12) {} double m_magsq; double m_magsqPeak; }; APTDemod *m_aptDemod; APTDemodSettings m_settings; int m_channelSampleRate; int m_channelFrequencyOffset; NCO m_nco; Interpolator m_interpolator; Real m_interpolatorDistance; Real m_interpolatorDistanceRemain; double m_magsq; double m_magsqSum; double m_magsqPeak; int m_magsqCount; MagSqLevelsStore m_magSqLevelStore; MessageQueue *m_messageQueueToChannel; MovingAverageUtil m_movingAverage; PhaseDiscriminators m_phaseDiscri; // Audio buffer - should probably use a FIFO float *m_samples; int m_sampleCount; int m_samplesLength; int m_readIdx; int m_writeIdx; int m_row; // Row of image currently being received int m_zenith; // Row number of Zenith void processOneSample(Complex &ci); MessageQueue *getMessageQueueToChannel() { return m_messageQueueToChannel; } }; #endif // INCLUDE_APTDEMODSINK_H