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490 lines
15 KiB
C++
490 lines
15 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 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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#include <algorithm>
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#include "audionetsink.h"
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#include "util/rtpsink.h"
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#include <QDebug>
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#include <QUdpSocket>
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const int AudioNetSink::m_udpBlockSize = 512;
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AudioNetSink::AudioNetSink(QObject *parent) :
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m_type(SinkUDP),
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m_codec(CodecL16),
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m_rtpBufferAudio(0),
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m_sampleRate(48000),
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m_stereo(false),
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m_decimation(1),
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m_decimationCount(0),
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m_codecInputSize(960),
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m_codecInputIndex(0),
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m_bufferIndex(0),
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m_port(9998)
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{
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std::fill(m_data, m_data+m_dataBlockSize, 0);
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std::fill(m_opusIn, m_opusIn+m_opusBlockSize, 0);
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m_codecRatio = (m_sampleRate / m_decimation) / (AudioOpus::m_bitrate / 8); // compressor ratio
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m_udpSocket = new QUdpSocket(parent);
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}
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AudioNetSink::AudioNetSink(QObject *parent, int sampleRate, bool stereo) :
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m_type(SinkUDP),
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m_codec(CodecL16),
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m_rtpBufferAudio(0),
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m_sampleRate(48000),
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m_stereo(false),
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m_decimation(1),
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m_decimationCount(0),
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m_codecInputSize(960),
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m_codecInputIndex(0),
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m_bufferIndex(0),
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m_port(9998)
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{
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std::fill(m_data, m_data+m_dataBlockSize, 0);
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std::fill(m_opusIn, m_opusIn+m_opusBlockSize, 0);
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m_codecRatio = (m_sampleRate / m_decimation) / (AudioOpus::m_bitrate / 8); // compressor ratio
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m_udpSocket = new QUdpSocket(parent);
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m_rtpBufferAudio = new RTPSink(m_udpSocket, sampleRate, stereo);
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}
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AudioNetSink::~AudioNetSink()
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{
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if (m_rtpBufferAudio) {
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delete m_rtpBufferAudio;
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}
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m_udpSocket->deleteLater(); // this thread is not the owner thread (was moved)
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}
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bool AudioNetSink::isRTPCapable() const
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{
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return m_rtpBufferAudio && m_rtpBufferAudio->isValid();
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}
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bool AudioNetSink::selectType(SinkType type)
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{
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if (type == SinkUDP)
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{
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m_type = SinkUDP;
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}
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else // this is SinkRTP
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{
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m_type = SinkRTP;
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}
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return true;
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}
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void AudioNetSink::setDestination(const QString& address, uint16_t port)
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{
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m_address.setAddress(const_cast<QString&>(address));
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m_port = port;
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if (m_rtpBufferAudio) {
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m_rtpBufferAudio->setDestination(address, port);
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}
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}
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void AudioNetSink::addDestination(const QString& address, uint16_t port)
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{
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if (m_rtpBufferAudio) {
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m_rtpBufferAudio->addDestination(address, port);
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}
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}
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void AudioNetSink::deleteDestination(const QString& address, uint16_t port)
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{
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if (m_rtpBufferAudio) {
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m_rtpBufferAudio->deleteDestination(address, port);
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}
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}
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void AudioNetSink::setParameters(Codec codec, bool stereo, int sampleRate)
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{
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qDebug() << "AudioNetSink::setParameters:"
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<< " codec: " << codec
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<< " stereo: " << stereo
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<< " sampleRate: " << sampleRate;
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m_codec = codec;
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m_stereo = stereo;
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m_sampleRate = sampleRate;
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setNewCodecData();
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if (m_rtpBufferAudio)
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{
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switch (m_codec)
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{
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case CodecPCMA:
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m_audioCompressor.fillALaw();
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m_rtpBufferAudio->setPayloadInformation(RTPSink::PayloadPCMA8, sampleRate);
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break;
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case CodecPCMU:
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m_audioCompressor.fillULaw();
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m_rtpBufferAudio->setPayloadInformation(RTPSink::PayloadPCMU8, sampleRate);
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break;
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case CodecL8:
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m_rtpBufferAudio->setPayloadInformation(RTPSink::PayloadL8, sampleRate);
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break;
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case CodecG722:
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m_rtpBufferAudio->setPayloadInformation(RTPSink::PayloadG722, sampleRate/2);
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break;
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case CodecOpus:
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m_rtpBufferAudio->setPayloadInformation(RTPSink::PayloadOpus, sampleRate);
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break;
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case CodecL16: // actually no codec
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default:
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m_rtpBufferAudio->setPayloadInformation(stereo ? RTPSink::PayloadL16Stereo : RTPSink::PayloadL16Mono, sampleRate);
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break;
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}
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}
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}
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void AudioNetSink::setDecimation(uint32_t decimation)
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{
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m_decimation = decimation < 1 ? 1 : decimation > 6 ? 6 : decimation;
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qDebug() << "AudioNetSink::setDecimation: " << m_decimation << " from: " << decimation;
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setNewCodecData();
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m_decimationCount = 0;
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}
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void AudioNetSink::setNewCodecData()
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{
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if (m_codec == CodecOpus)
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{
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m_codecInputSize = m_sampleRate / (m_decimation * 50); // 20ms = 1/50s - size is per channel
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m_codecInputSize = m_codecInputSize > 960 ? 960 : m_codecInputSize; // hard limit of 48 kS/s
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m_codecRatio = (m_sampleRate / m_decimation) / (AudioOpus::m_bitrate / 8); // compressor ratio
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qDebug() << "AudioNetSink::setNewCodecData: CodecOpus:"
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<< " m_codecInputSize: " << m_codecInputSize
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<< " m_codecRatio: " << m_codecRatio
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<< " Fs: " << m_sampleRate/m_decimation
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<< " stereo: " << m_stereo;
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m_opus.setEncoder(m_sampleRate/m_decimation, m_stereo ? 2 : 1);
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m_codecInputIndex = 0;
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m_bufferIndex = 0;
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}
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setDecimationFilters();
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}
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void AudioNetSink::setDecimationFilters()
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{
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int decimatedSampleRate = m_sampleRate / m_decimation;
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switch (m_codec)
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{
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case CodecPCMA:
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case CodecPCMU:
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m_audioFilterR.setDecimFilters(m_sampleRate, decimatedSampleRate, 3300.0, 300.0);
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m_audioFilterL.setDecimFilters(m_sampleRate, decimatedSampleRate, 3300.0, 300.0);
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break;
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case CodecG722:
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m_audioFilterR.setDecimFilters(m_sampleRate, decimatedSampleRate, 7000.0, 50.0);
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m_audioFilterL.setDecimFilters(m_sampleRate, decimatedSampleRate, 7000.0, 50.0);
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break;
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case CodecOpus:
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case CodecL8:
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case CodecL16:
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default:
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m_audioFilterR.setDecimFilters(m_sampleRate, decimatedSampleRate, 0.45*decimatedSampleRate, 50.0);
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m_audioFilterL.setDecimFilters(m_sampleRate, decimatedSampleRate, 0.45*decimatedSampleRate, 50.0);
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break;
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}
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}
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void AudioNetSink::write(qint16 isample)
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{
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qint16& sample = isample;
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if (m_decimation > 1)
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{
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float lpSample = m_audioFilterR.run(sample / 32768.0f);
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if (m_decimationCount >= m_decimation - 1)
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{
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sample = lpSample * 32768.0f;
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m_decimationCount = 0;
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}
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else
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{
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m_decimationCount++;
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return;
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}
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}
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if (m_type == SinkUDP)
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{
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if (m_codec == CodecG722)
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{
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if (m_bufferIndex >= 2*m_udpBlockSize)
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{
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m_udpSocket->writeDatagram((const char*) m_data, (qint64 ) m_udpBlockSize, m_address, m_port);
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m_bufferIndex = 0;
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}
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}
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else
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{
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if (m_bufferIndex >= m_udpBlockSize)
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{
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m_udpSocket->writeDatagram((const char*) m_data, (qint64 ) m_udpBlockSize, m_address, m_port);
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m_bufferIndex = 0;
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}
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}
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switch(m_codec)
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{
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case CodecPCMA:
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case CodecPCMU:
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{
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qint8 *p = (qint8*) &m_data[m_bufferIndex];
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*p = m_audioCompressor.compress8(sample);
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m_bufferIndex += sizeof(qint8);
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}
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break;
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case CodecL8:
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{
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qint8 *p = (qint8*) &m_data[m_bufferIndex];
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*p = sample / 256;
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m_bufferIndex += sizeof(qint8);
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}
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break;
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case CodecG722:
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{
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qint16 *p = (qint16*) &m_data[m_udpBlockSize + 2*m_bufferIndex];
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*p = sample;
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m_bufferIndex += 1;
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if (m_bufferIndex == 2*m_udpBlockSize) {
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m_g722.encode((uint8_t *) m_data, (const int16_t*) &m_data[m_udpBlockSize], 2*m_udpBlockSize);
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}
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}
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break;
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case CodecOpus:
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{
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if (m_codecInputIndex == m_codecInputSize)
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{
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int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
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nbBytes = nbBytes > m_udpBlockSize ? m_udpBlockSize : nbBytes;
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m_udpSocket->writeDatagram((const char*) m_data, (qint64 ) nbBytes, m_address, m_port);
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m_codecInputIndex = 0;
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}
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m_opusIn[m_codecInputIndex++] = sample;
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}
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break;
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case CodecL16:
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default:
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{
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qint16 *p = (qint16*) &m_data[m_bufferIndex];
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*p = sample;
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m_bufferIndex += sizeof(qint16);
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}
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break;
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}
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}
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else if (m_type == SinkRTP)
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{
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switch(m_codec)
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{
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case CodecPCMA:
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case CodecPCMU:
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{
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qint8 p = m_audioCompressor.compress8(sample);
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m_rtpBufferAudio->write((uint8_t *) &p);
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}
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break;
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case CodecL8:
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{
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qint8 p = sample / 256;
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m_rtpBufferAudio->write((uint8_t *) &p);
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}
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break;
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case CodecG722:
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{
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if (m_bufferIndex >= 2*m_g722BlockSize)
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{
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m_g722.encode((uint8_t *) m_data, (const int16_t*) &m_data[m_g722BlockSize], 2*m_g722BlockSize);
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m_bufferIndex = 0;
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}
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if (m_bufferIndex % 2 == 0) {
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m_rtpBufferAudio->write((uint8_t *) &m_data[m_bufferIndex/2]);
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}
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qint16 *p = (qint16*) &m_data[m_g722BlockSize + 2*m_bufferIndex];
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*p = sample;
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m_bufferIndex += 1;
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}
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break;
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case CodecOpus:
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{
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if (m_codecInputIndex == m_codecInputSize)
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{
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int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
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if (nbBytes != AudioOpus::m_bitrate/400) { // 8 bits for 1/50s (20ms)
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qWarning("AudioNetSink::write: CodecOpus mono: unexpected output frame size: %d bytes", nbBytes);
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}
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m_bufferIndex = 0;
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m_codecInputIndex = 0;
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}
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if (m_codecInputIndex % m_codecRatio == 0) {
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m_rtpBufferAudio->write((uint8_t *) &m_data[m_bufferIndex++]);
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}
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m_opusIn[m_codecInputIndex++] = sample;
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}
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break;
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case CodecL16:
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default:
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m_rtpBufferAudio->write((uint8_t *) &sample);
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break;
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}
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}
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}
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void AudioNetSink::write(qint16 ilSample, qint16 irSample)
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{
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qint16& lSample = ilSample;
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qint16& rSample = irSample;
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if (m_decimation > 1)
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{
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float lpLSample = m_audioFilterL.runLP(lSample / 32768.0f);
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float lpRSample = m_audioFilterR.runLP(rSample / 32768.0f);
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if (m_decimationCount >= m_decimation - 1)
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{
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lSample = lpLSample * 32768.0f;
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rSample = lpRSample * 32768.0f;
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m_decimationCount = 0;
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}
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else
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{
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m_decimationCount++;
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return;
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}
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}
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if (m_type == SinkUDP)
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{
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if (m_bufferIndex >= m_udpBlockSize)
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{
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m_udpSocket->writeDatagram((const char*) m_data, (qint64 ) m_udpBlockSize, m_address, m_port);
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m_bufferIndex = 0;
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}
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switch(m_codec)
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{
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case CodecPCMA:
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case CodecPCMU:
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case CodecG722:
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break; // mono modes - do nothing
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case CodecOpus:
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{
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if (m_codecInputIndex == m_codecInputSize)
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{
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int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
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nbBytes = nbBytes > m_udpBlockSize ? m_udpBlockSize : nbBytes;
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m_udpSocket->writeDatagram((const char*) m_data, (qint64 ) nbBytes, m_address, m_port);
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m_codecInputIndex = 0;
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}
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m_opusIn[2*m_codecInputIndex] = lSample;
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m_opusIn[2*m_codecInputIndex+1] = rSample;
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m_codecInputIndex++;
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}
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break;
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case CodecL8:
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{
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qint8 *p = (qint8*) &m_data[m_bufferIndex];
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*p = lSample / 256;
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m_bufferIndex += sizeof(qint8);
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p = (qint8*) &m_data[m_bufferIndex];
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*p = rSample / 256;
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m_bufferIndex += sizeof(qint8);
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}
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break;
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case CodecL16:
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default:
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{
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qint16 *p = (qint16*) &m_data[m_bufferIndex];
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*p = lSample;
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m_bufferIndex += sizeof(qint16);
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p = (qint16*) &m_data[m_bufferIndex];
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*p = rSample;
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m_bufferIndex += sizeof(qint16);
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}
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break;
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}
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}
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else if (m_type == SinkRTP)
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{
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switch(m_codec)
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{
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case CodecPCMA:
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case CodecPCMU:
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case CodecG722:
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break; // mono modes - do nothing
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case CodecOpus:
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{
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if (m_codecInputIndex == m_codecInputSize)
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{
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int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
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if (nbBytes != AudioOpus::m_bitrate/400) { // 8 bits for 1/50s (20ms)
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qWarning("AudioNetSink::write: CodecOpus stereo: unexpected output frame size: %d bytes", nbBytes);
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}
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m_bufferIndex = 0;
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m_codecInputIndex = 0;
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}
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if (m_codecInputIndex % m_codecRatio == 0) {
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m_rtpBufferAudio->write((uint8_t *) &m_data[m_bufferIndex++]);
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}
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m_opusIn[2*m_codecInputIndex] = lSample;
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m_opusIn[2*m_codecInputIndex+1] = rSample;
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m_codecInputIndex++;
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}
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break;
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case CodecL8:
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{
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qint8 pl = lSample / 256;
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qint8 pr = rSample / 256;
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m_rtpBufferAudio->write((uint8_t *) &pl, (uint8_t *) &pr);
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}
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break;
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case CodecL16:
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default:
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m_rtpBufferAudio->write((uint8_t *) &lSample, (uint8_t *) &rSample);
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break;
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}
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}
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}
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void AudioNetSink::moveToThread(QThread *thread)
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{
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m_udpSocket->moveToThread(thread);
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}
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