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127 lines
4.3 KiB
C++
127 lines
4.3 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2019 Edouard Griffiths, F4EXB //
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// Copyright (C) 2021 Jon Beniston, M7RCE //
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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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// (at your option) any later version. //
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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_APTDEMODSINK_H
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#define INCLUDE_APTDEMODSINK_H
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#include "dsp/channelsamplesink.h"
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#include "dsp/phasediscri.h"
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#include "dsp/nco.h"
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#include "dsp/interpolator.h"
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#include "dsp/firfilter.h"
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#include "util/movingaverage.h"
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#include "util/doublebufferfifo.h"
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#include "util/messagequeue.h"
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#include "aptdemodsettings.h"
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#include <apt.h>
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#include <vector>
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#include <iostream>
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#include <fstream>
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// FIXME: Use lower sample rate for better SNR?
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// Do we want an audio filter? Subcarrier at 2800Hz. Does libaptdec have one?
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#define APTDEMOD_AUDIO_SAMPLE_RATE 48000
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// Lines are 2 per second -> 4160 words per second
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class APTDemod;
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class APTDemodSink : public ChannelSampleSink {
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public:
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APTDemodSink(APTDemod *packetDemod);
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~APTDemodSink();
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virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end);
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void applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force = false);
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void applySettings(const APTDemodSettings& settings, bool force = false);
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void setMessageQueueToChannel(MessageQueue *messageQueue) { m_messageQueueToChannel = messageQueue; }
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double getMagSq() const { return m_magsq; }
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void getMagSqLevels(double& avg, double& peak, int& nbSamples)
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{
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if (m_magsqCount > 0)
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{
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m_magsq = m_magsqSum / m_magsqCount;
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m_magSqLevelStore.m_magsq = m_magsq;
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m_magSqLevelStore.m_magsqPeak = m_magsqPeak;
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}
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avg = m_magSqLevelStore.m_magsq;
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peak = m_magSqLevelStore.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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int getSamples(float *samples, int count);
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void resetDecoder();
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private:
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struct MagSqLevelsStore
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{
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MagSqLevelsStore() :
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m_magsq(1e-12),
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m_magsqPeak(1e-12)
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{}
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double m_magsq;
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double m_magsqPeak;
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};
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APTDemod *m_aptDemod;
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APTDemodSettings m_settings;
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int m_channelSampleRate;
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int m_channelFrequencyOffset;
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NCO 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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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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MagSqLevelsStore m_magSqLevelStore;
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MessageQueue *m_messageQueueToChannel;
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MovingAverageUtil<Real, double, 16> m_movingAverage;
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PhaseDiscriminators m_phaseDiscri;
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// Audio buffer - should probably use a FIFO
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float *m_samples;
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int m_sampleCount;
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int m_samplesLength;
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int m_readIdx;
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int m_writeIdx;
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int m_row; // Row of image currently being received
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int m_zenith; // Row number of Zenith
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void processOneSample(Complex &ci);
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MessageQueue *getMessageQueueToChannel() { return m_messageQueueToChannel; }
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};
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#endif // INCLUDE_APTDEMODSINK_H
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