kopia lustrzana https://github.com/f4exb/sdrangel
155 wiersze
5.3 KiB
C++
155 wiersze
5.3 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_DABDEMODSINK_H
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#define INCLUDE_DABDEMODSINK_H
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#include <QVector>
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#include "dsp/channelsamplesink.h"
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#include "dsp/nco.h"
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#include "dsp/interpolator.h"
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#include "util/movingaverage.h"
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#include "util/messagequeue.h"
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#include "audio/audiofifo.h"
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#include "dabdemodsettings.h"
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#include "dabdemoddevice.h"
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#include <vector>
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#include <dab-api.h>
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#define DABDEMOD_CHANNEL_SAMPLE_RATE 2048000
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class ChannelAPI;
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class DABDemod;
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class DABDemodSink : public ChannelSampleSink {
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public:
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DABDemodSink(DABDemod *packetDemod);
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~DABDemodSink();
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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 DABDemodSettings& settings, bool force = false);
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void applyAudioSampleRate(int sampleRate);
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void applyDABAudioSampleRate(int sampleRate);
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int getAudioSampleRate() const { return m_audioSampleRate; }
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AudioFifo *getAudioFifo() { return &m_audioFifo; }
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void setAudioFifoLabel(const QString& label) { m_audioFifo.setLabel(label); }
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void setMessageQueueToChannel(MessageQueue *messageQueue) { m_messageQueueToChannel = messageQueue; }
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void setChannel(ChannelAPI *channel) { m_channel = channel; }
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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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void reset();
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void resetService();
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void programAvailable(const QString& programName);
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// Callbacks
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void systemData(bool sync, int16_t snr, int32_t freqOffset);
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void ensembleName(const QString& name, int id);
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void programName(const QString& name, int id);
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void programData(int bitrate, const QString& audio, const QString& language, const QString& programType);
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void audio(int16_t *buffer, int size, int samplerate, bool stereo);
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void programQuality(int16_t frames, int16_t rs, int16_t aac);
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void fibQuality(int16_t percent);
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void data(const QString& data);
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void motData(const uint8_t *data, int len, const QString& filename, int contentSubType);
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void tii(int tii);
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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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DABDemod *m_dabDemod;
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DABDemodSettings m_settings;
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ChannelAPI *m_channel;
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int m_audioSampleRate; // Output device sample rate
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int m_dabAudioSampleRate;
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int m_channelSampleRate;
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int m_channelFrequencyOffset;
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void *m_dab;
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DABDemodDevice m_device;
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audiodata m_ad;
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API_struct m_api;
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bool m_programSet;
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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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Interpolator m_audioInterpolator;
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Real m_audioInterpolatorDistance;
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Real m_audioInterpolatorDistanceRemain;
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AudioVector m_audioBuffer;
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AudioFifo m_audioFifo;
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std::size_t m_audioBufferFill;
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QVector<qint16> m_demodBuffer;
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int m_demodBufferFill;
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void processOneSample(Complex &ci);
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void processOneAudioSample(Complex &ci);
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MessageQueue *getMessageQueueToChannel() { return m_messageQueueToChannel; }
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void setProgram(const QString& name);
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};
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#endif // INCLUDE_DABDEMODSINK_H
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