kopia lustrzana https://github.com/f4exb/sdrangel
107 wiersze
4.4 KiB
C++
107 wiersze
4.4 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2012 maintech GmbH, Otto-Hahn-Str. 15, 97204 Hoechberg, Germany //
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// written by Christian Daniel //
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// Copyright (C) 2015-2016, 2018-2019 Edouard Griffiths, F4EXB <f4exb06@gmail.com> //
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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 SDRBASE_AUDIO_AUDIONETSINK_H_
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#define SDRBASE_AUDIO_AUDIONETSINK_H_
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#include "audiofilter.h"
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#include "audiocompressor.h"
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#include "audiog722.h"
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#include "audioopus.h"
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#include "export.h"
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#include <QObject>
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#include <QHostAddress>
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#include <stdint.h>
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class QUdpSocket;
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class RTPSink;
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class QThread;
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class SDRBASE_API AudioNetSink {
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public:
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typedef enum
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{
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SinkUDP,
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SinkRTP
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} SinkType;
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typedef enum
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{
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CodecL16, //!< Linear 16 bit samples (no formatting)
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CodecL8, //!< Linear 8 bit samples
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CodecPCMA, //!< PCM A-law 8 bit samples
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CodecPCMU, //!< PCM Mu-law 8 bit samples
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CodecG722, //!< G722 compressed 8 bit samples 16kS/s in 8kS/s out
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CodecOpus //!< Opus compressed 8 bit samples at 64kbits/s (8kS/s out). Various input sample rates
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} Codec;
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AudioNetSink(QObject *parent); //!< without RTP
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AudioNetSink(QObject *parent, int sampleRate, bool stereo); //!< with RTP
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~AudioNetSink();
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void setDestination(const QString& address, uint16_t port);
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void addDestination(const QString& address, uint16_t port);
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void deleteDestination(const QString& address, uint16_t port);
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void setParameters(Codec codec, bool stereo, int sampleRate);
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void setDecimation(uint32_t decimation);
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void write(qint16 sample);
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void write(qint16 lSample, qint16 rSample);
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bool isRTPCapable() const;
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bool selectType(SinkType type);
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void moveToThread(QThread *thread);
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static const int m_udpBlockSize;
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static const int m_dataBlockSize = 65536; // room for G722 conversion (64000 = 12800*5 largest to date)
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static const int m_g722BlockSize = 12800; // number of resulting G722 bytes (80*20ms frames)
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static const int m_opusBlockSize = 960*2; // provision for 20ms of 2 int16 channels at 48 kS/s
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static const int m_opusOutputSize = 160; // output frame: 20ms of 8 bit data @ 64 kbits/s = 160 bytes
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protected:
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void setNewCodecData(); // actions to take when changes affecting codec dependent data occurs
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void setDecimationFilters(); // set decimation filters limits depending on effective sample rate and codec
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SinkType m_type;
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Codec m_codec;
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QUdpSocket *m_udpSocket;
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RTPSink *m_rtpBufferAudio;
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AudioCompressor m_audioCompressor;
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AudioG722 m_g722;
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AudioOpus m_opus;
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AudioFilter m_audioFilterR;
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AudioFilter m_audioFilterL;
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int m_sampleRate;
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bool m_stereo;
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uint32_t m_decimation;
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uint32_t m_decimationCount;
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char m_data[m_dataBlockSize];
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int16_t m_opusIn[m_opusBlockSize];
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int m_codecInputSize; // codec input block size - for codecs with actual encoding (Opus only for now)
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int m_codecInputIndex; // codec input block fill index
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int m_codecRatio; // codec compression ratio
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unsigned int m_bufferIndex;
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QHostAddress m_address;
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unsigned int m_port;
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};
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#endif /* SDRBASE_AUDIO_AUDIONETSINK_H_ */
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