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sdrangel/sdrbase/audio/audionetsink.h
2023-12-11 01:49:11 +01:00

108 lines
4.4 KiB
C++

///////////////////////////////////////////////////////////////////////////////////
// Copyright (C) 2012 maintech GmbH, Otto-Hahn-Str. 15, 97204 Hoechberg, Germany //
// written by Christian Daniel //
// Copyright (C) 2015-2016, 2018-2019 Edouard Griffiths, F4EXB <f4exb06@gmail.com> //
// //
// 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 SDRBASE_AUDIO_AUDIONETSINK_H_
#define SDRBASE_AUDIO_AUDIONETSINK_H_
#include "dsp/dsptypes.h"
#include "audiofilter.h"
#include "audiocompressor.h"
#include "audiog722.h"
#include "audioopus.h"
#include "export.h"
#include <QObject>
#include <QHostAddress>
#include <stdint.h>
class QUdpSocket;
class RTPSink;
class QThread;
class SDRBASE_API AudioNetSink {
public:
typedef enum
{
SinkUDP,
SinkRTP
} SinkType;
typedef enum
{
CodecL16, //!< Linear 16 bit samples (no formatting)
CodecL8, //!< Linear 8 bit samples
CodecPCMA, //!< PCM A-law 8 bit samples
CodecPCMU, //!< PCM Mu-law 8 bit samples
CodecG722, //!< G722 compressed 8 bit samples 16kS/s in 8kS/s out
CodecOpus //!< Opus compressed 8 bit samples at 64kbits/s (8kS/s out). Various input sample rates
} Codec;
AudioNetSink(QObject *parent); //!< without RTP
AudioNetSink(QObject *parent, int sampleRate, bool stereo); //!< with RTP
~AudioNetSink();
void setDestination(const QString& address, uint16_t port);
void addDestination(const QString& address, uint16_t port);
void deleteDestination(const QString& address, uint16_t port);
void setParameters(Codec codec, bool stereo, int sampleRate);
void setDecimation(uint32_t decimation);
void write(qint16 sample);
void write(qint16 lSample, qint16 rSample);
bool isRTPCapable() const;
bool selectType(SinkType type);
void moveToThread(QThread *thread);
static const int m_udpBlockSize;
static const int m_dataBlockSize = 65536; // room for G722 conversion (64000 = 12800*5 largest to date)
static const int m_g722BlockSize = 12800; // number of resulting G722 bytes (80*20ms frames)
static const int m_opusBlockSize = 960*2; // provision for 20ms of 2 int16 channels at 48 kS/s
static const int m_opusOutputSize = 160; // output frame: 20ms of 8 bit data @ 64 kbits/s = 160 bytes
protected:
void setNewCodecData(); // actions to take when changes affecting codec dependent data occurs
void setDecimationFilters(); // set decimation filters limits depending on effective sample rate and codec
SinkType m_type;
Codec m_codec;
QUdpSocket *m_udpSocket;
RTPSink *m_rtpBufferAudio;
AudioCompressor m_audioCompressor;
AudioG722 m_g722;
AudioOpus m_opus;
AudioFilter m_audioFilterR;
AudioFilter m_audioFilterL;
int m_sampleRate;
bool m_stereo;
uint32_t m_decimation;
uint32_t m_decimationCount;
char m_data[m_dataBlockSize];
int16_t m_opusIn[m_opusBlockSize];
int m_codecInputSize; // codec input block size - for codecs with actual encoding (Opus only for now)
int m_codecInputIndex; // codec input block fill index
int m_codecRatio; // codec compression ratio
unsigned int m_bufferIndex;
QHostAddress m_address;
unsigned int m_port;
};
#endif /* SDRBASE_AUDIO_AUDIONETSINK_H_ */