/////////////////////////////////////////////////////////////////////////////////// // Copyright (C) 2019-2020, 2022-2023 Edouard Griffiths, F4EXB // // Copyright (C) 2020-2021, 2023 Jon Beniston, M7RCE // // Copyright (C) 2020 Kacper Michajłow // // // // 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_DABDEMODSINK_H #define INCLUDE_DABDEMODSINK_H #include #include "dsp/channelsamplesink.h" #include "dsp/nco.h" #include "dsp/interpolator.h" #include "util/movingaverage.h" #include "util/messagequeue.h" #include "audio/audiofifo.h" #include "dabdemodsettings.h" #include "dabdemoddevice.h" #include #include #define DABDEMOD_CHANNEL_SAMPLE_RATE 2048000 class ChannelAPI; class DABDemod; class DABDemodSink : public ChannelSampleSink { public: DABDemodSink(); ~DABDemodSink(); 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 DABDemodSettings& settings, bool force = false); void applyAudioSampleRate(int sampleRate); void applyDABAudioSampleRate(int sampleRate); int getAudioSampleRate() const { return m_audioSampleRate; } AudioFifo *getAudioFifo() { return &m_audioFifo; } void setAudioFifoLabel(const QString& label) { m_audioFifo.setLabel(label); } void setMessageQueueToChannel(MessageQueue *messageQueue) { m_messageQueueToChannel = messageQueue; } void setChannel(ChannelAPI *channel) { m_channel = channel; } 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; } void reset(); void resetService(); void programAvailable(const QString& programName); // Callbacks void systemData(bool sync, int16_t snr, int32_t freqOffset); void ensembleName(const QString& name, int id); void programName(const QString& name, int id); void programData(int bitrate, const QString& audio, const QString& language, const QString& programType); void audio(int16_t *buffer, int size, int samplerate, bool stereo); void programQuality(int16_t frames, int16_t rs, int16_t aac); void fibQuality(int16_t percent); void data(const QString& data); void motData(const uint8_t *data, int len, const QString& filename, int contentSubType); void tii(int tii); private: struct MagSqLevelsStore { MagSqLevelsStore() : m_magsq(1e-12), m_magsqPeak(1e-12) {} double m_magsq; double m_magsqPeak; }; DABDemodSettings m_settings; ChannelAPI *m_channel; int m_audioSampleRate; // Output device sample rate int m_dabAudioSampleRate; int m_channelSampleRate; int m_channelFrequencyOffset; void *m_dab; DABDemodDevice m_device; audiodata m_ad; API_struct m_api; bool m_programSet; 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; Interpolator m_audioInterpolator; Real m_audioInterpolatorDistance; Real m_audioInterpolatorDistanceRemain; AudioVector m_audioBuffer; AudioFifo m_audioFifo; std::size_t m_audioBufferFill; QVector m_demodBuffer; int m_demodBufferFill; void processOneSample(Complex &ci); void processOneAudioSample(Complex &ci); MessageQueue *getMessageQueueToChannel() { return m_messageQueueToChannel; } void setProgram(const QString& name); }; #endif // INCLUDE_DABDEMODSINK_H