1
0
mirror of https://github.com/f4exb/sdrangel.git synced 2024-11-04 16:01:14 -05:00
sdrangel/plugins/channelrx/demoddab/dabdemodsink.h

151 lines
5.1 KiB
C
Raw Normal View History

2021-04-16 17:56:15 -04:00
///////////////////////////////////////////////////////////////////////////////////
// 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 <http://www.gnu.org/licenses/>. //
///////////////////////////////////////////////////////////////////////////////////
#ifndef INCLUDE_DABDEMODSINK_H
#define INCLUDE_DABDEMODSINK_H
#include <QVector>
#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 <vector>
#include <dab-api.h>
#define DABDEMOD_CHANNEL_SAMPLE_RATE 2048000
class ChannelAPI;
class DABDemod;
class DABDemodSink : public ChannelSampleSink {
public:
DABDemodSink(DABDemod *packetDemod);
~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 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();
// 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);
2021-04-16 17:56:15 -04:00
private:
struct MagSqLevelsStore
{
MagSqLevelsStore() :
m_magsq(1e-12),
m_magsqPeak(1e-12)
{}
double m_magsq;
double m_magsqPeak;
};
DABDemod *m_dabDemod;
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;
packetdata m_pd;
API_struct m_api;
2021-04-16 17:56:15 -04:00
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<Real, double, 16> m_movingAverage;
Interpolator m_audioInterpolator;
Real m_audioInterpolatorDistance;
Real m_audioInterpolatorDistanceRemain;
AudioVector m_audioBuffer;
AudioFifo m_audioFifo;
uint32_t m_audioBufferFill;
QVector<qint16> m_demodBuffer;
int m_demodBufferFill;
void processOneSample(Complex &ci);
void processOneAudioSample(Complex &ci);
MessageQueue *getMessageQueueToChannel() { return m_messageQueueToChannel; }
};
#endif // INCLUDE_DABDEMODSINK_H