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sdrangel/sdrbase/audio/audionetsink.cpp

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///////////////////////////////////////////////////////////////////////////////////
// Copyright (C) 2018 F4EXB //
// written by Edouard Griffiths //
// //
// 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/>. //
///////////////////////////////////////////////////////////////////////////////////
#include <algorithm>
#include "audionetsink.h"
#include "util/rtpsink.h"
#include <QDebug>
#include <QUdpSocket>
const int AudioNetSink::m_udpBlockSize = 512;
AudioNetSink::AudioNetSink(QObject *parent) :
m_type(SinkUDP),
m_codec(CodecL16),
m_rtpBufferAudio(0),
m_sampleRate(48000),
m_stereo(false),
m_decimation(1),
m_decimationCount(0),
m_codecInputSize(960),
m_codecInputIndex(0),
m_bufferIndex(0),
m_port(9998)
{
std::fill(m_data, m_data+m_dataBlockSize, 0);
std::fill(m_opusIn, m_opusIn+m_opusBlockSize, 0);
m_codecRatio = (m_sampleRate / m_decimation) / (AudioOpus::m_bitrate / 8); // compressor ratio
m_udpSocket = new QUdpSocket(parent);
}
AudioNetSink::AudioNetSink(QObject *parent, int sampleRate, bool stereo) :
m_type(SinkUDP),
m_codec(CodecL16),
m_rtpBufferAudio(0),
m_sampleRate(48000),
m_stereo(false),
m_decimation(1),
m_decimationCount(0),
m_codecInputSize(960),
m_codecInputIndex(0),
m_bufferIndex(0),
m_port(9998)
{
std::fill(m_data, m_data+m_dataBlockSize, 0);
std::fill(m_opusIn, m_opusIn+m_opusBlockSize, 0);
m_codecRatio = (m_sampleRate / m_decimation) / (AudioOpus::m_bitrate / 8); // compressor ratio
m_udpSocket = new QUdpSocket(parent);
m_rtpBufferAudio = new RTPSink(m_udpSocket, sampleRate, stereo);
}
AudioNetSink::~AudioNetSink()
{
if (m_rtpBufferAudio) {
delete m_rtpBufferAudio;
}
m_udpSocket->deleteLater(); // this thread is not the owner thread (was moved)
}
bool AudioNetSink::isRTPCapable() const
{
return m_rtpBufferAudio && m_rtpBufferAudio->isValid();
}
bool AudioNetSink::selectType(SinkType type)
{
if (type == SinkUDP)
{
m_type = SinkUDP;
}
else // this is SinkRTP
{
m_type = SinkRTP;
}
return true;
}
void AudioNetSink::setDestination(const QString& address, uint16_t port)
{
m_address.setAddress(const_cast<QString&>(address));
m_port = port;
if (m_rtpBufferAudio) {
m_rtpBufferAudio->setDestination(address, port);
}
}
void AudioNetSink::addDestination(const QString& address, uint16_t port)
{
if (m_rtpBufferAudio) {
m_rtpBufferAudio->addDestination(address, port);
}
}
void AudioNetSink::deleteDestination(const QString& address, uint16_t port)
{
if (m_rtpBufferAudio) {
m_rtpBufferAudio->deleteDestination(address, port);
}
}
void AudioNetSink::setParameters(Codec codec, bool stereo, int sampleRate)
{
qDebug() << "AudioNetSink::setParameters:"
<< " codec: " << codec
<< " stereo: " << stereo
<< " sampleRate: " << sampleRate;
m_codec = codec;
m_stereo = stereo;
m_sampleRate = sampleRate;
setNewCodecData();
if (m_rtpBufferAudio)
{
switch (m_codec)
{
case CodecPCMA:
m_audioCompressor.fillALaw();
m_rtpBufferAudio->setPayloadInformation(RTPSink::PayloadPCMA8, sampleRate);
break;
case CodecPCMU:
m_audioCompressor.fillULaw();
m_rtpBufferAudio->setPayloadInformation(RTPSink::PayloadPCMU8, sampleRate);
break;
case CodecL8:
m_rtpBufferAudio->setPayloadInformation(RTPSink::PayloadL8, sampleRate);
break;
case CodecG722:
m_rtpBufferAudio->setPayloadInformation(RTPSink::PayloadG722, sampleRate/2);
break;
case CodecOpus:
m_rtpBufferAudio->setPayloadInformation(RTPSink::PayloadOpus, sampleRate);
break;
case CodecL16: // actually no codec
default:
m_rtpBufferAudio->setPayloadInformation(stereo ? RTPSink::PayloadL16Stereo : RTPSink::PayloadL16Mono, sampleRate);
break;
}
}
}
void AudioNetSink::setDecimation(uint32_t decimation)
{
m_decimation = decimation < 1 ? 1 : decimation > 6 ? 6 : decimation;
qDebug() << "AudioNetSink::setDecimation: " << m_decimation << " from: " << decimation;
setNewCodecData();
m_decimationCount = 0;
}
void AudioNetSink::setNewCodecData()
{
if (m_codec == CodecOpus)
{
m_codecInputSize = m_sampleRate / (m_decimation * 50); // 20ms = 1/50s - size is per channel
m_codecInputSize = m_codecInputSize > 960 ? 960 : m_codecInputSize; // hard limit of 48 kS/s
m_codecRatio = (m_sampleRate / m_decimation) / (AudioOpus::m_bitrate / 8); // compressor ratio
qDebug() << "AudioNetSink::setNewCodecData: CodecOpus:"
<< " m_codecInputSize: " << m_codecInputSize
<< " m_codecRatio: " << m_codecRatio
<< " Fs: " << m_sampleRate/m_decimation
<< " stereo: " << m_stereo;
m_opus.setEncoder(m_sampleRate/m_decimation, m_stereo ? 2 : 1);
m_codecInputIndex = 0;
m_bufferIndex = 0;
}
setDecimationFilters();
}
void AudioNetSink::setDecimationFilters()
{
int decimatedSampleRate = m_sampleRate / m_decimation;
switch (m_codec)
{
case CodecPCMA:
case CodecPCMU:
m_audioFilter.setDecimFilters(m_sampleRate, decimatedSampleRate, 3300.0, 300.0);
break;
case CodecG722:
m_audioFilter.setDecimFilters(m_sampleRate, decimatedSampleRate, 7000.0, 50.0);
break;
case CodecOpus:
case CodecL8:
case CodecL16:
default:
m_audioFilter.setDecimFilters(m_sampleRate, decimatedSampleRate, 0.45*decimatedSampleRate, 50.0);
break;
}
}
void AudioNetSink::write(qint16 isample)
{
qint16& sample = isample;
if (m_decimation > 1)
{
float lpSample = m_audioFilter.run(sample / 32768.0f);
if (m_decimationCount >= m_decimation - 1)
{
sample = lpSample * 32768.0f;
m_decimationCount = 0;
}
else
{
m_decimationCount++;
return;
}
}
if (m_type == SinkUDP)
{
if (m_codec == CodecG722)
{
if (m_bufferIndex >= 2*m_udpBlockSize)
{
m_udpSocket->writeDatagram((const char*) m_data, (qint64 ) m_udpBlockSize, m_address, m_port);
m_bufferIndex = 0;
}
}
else
{
if (m_bufferIndex >= m_udpBlockSize)
{
m_udpSocket->writeDatagram((const char*) m_data, (qint64 ) m_udpBlockSize, m_address, m_port);
m_bufferIndex = 0;
}
}
switch(m_codec)
{
case CodecPCMA:
case CodecPCMU:
{
qint8 *p = (qint8*) &m_data[m_bufferIndex];
*p = m_audioCompressor.compress8(sample);
m_bufferIndex += sizeof(qint8);
}
break;
case CodecL8:
{
qint8 *p = (qint8*) &m_data[m_bufferIndex];
*p = sample / 256;
m_bufferIndex += sizeof(qint8);
}
break;
case CodecG722:
{
qint16 *p = (qint16*) &m_data[m_udpBlockSize + 2*m_bufferIndex];
*p = sample;
m_bufferIndex += 1;
if (m_bufferIndex == 2*m_udpBlockSize) {
m_g722.encode((uint8_t *) m_data, (const int16_t*) &m_data[m_udpBlockSize], 2*m_udpBlockSize);
}
}
break;
case CodecOpus:
{
if (m_codecInputIndex == m_codecInputSize)
{
int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
nbBytes = nbBytes > m_udpBlockSize ? m_udpBlockSize : nbBytes;
m_udpSocket->writeDatagram((const char*) m_data, (qint64 ) nbBytes, m_address, m_port);
m_codecInputIndex = 0;
}
m_opusIn[m_codecInputIndex++] = sample;
}
break;
case CodecL16:
default:
{
qint16 *p = (qint16*) &m_data[m_bufferIndex];
*p = sample;
m_bufferIndex += sizeof(qint16);
}
break;
}
}
else if (m_type == SinkRTP)
{
switch(m_codec)
{
case CodecPCMA:
case CodecPCMU:
{
qint8 p = m_audioCompressor.compress8(sample);
m_rtpBufferAudio->write((uint8_t *) &p);
}
break;
case CodecL8:
{
qint8 p = sample / 256;
m_rtpBufferAudio->write((uint8_t *) &p);
}
break;
case CodecG722:
{
if (m_bufferIndex >= 2*m_g722BlockSize)
{
m_g722.encode((uint8_t *) m_data, (const int16_t*) &m_data[m_g722BlockSize], 2*m_g722BlockSize);
m_bufferIndex = 0;
}
if (m_bufferIndex % 2 == 0) {
m_rtpBufferAudio->write((uint8_t *) &m_data[m_bufferIndex/2]);
}
qint16 *p = (qint16*) &m_data[m_g722BlockSize + 2*m_bufferIndex];
*p = sample;
m_bufferIndex += 1;
}
break;
case CodecOpus:
{
if (m_codecInputIndex == m_codecInputSize)
{
int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
if (nbBytes != AudioOpus::m_bitrate/400) { // 8 bits for 1/50s (20ms)
qWarning("AudioNetSink::write: CodecOpus mono: unexpected output frame size: %d bytes", nbBytes);
}
m_bufferIndex = 0;
m_codecInputIndex = 0;
}
if (m_codecInputIndex % m_codecRatio == 0) {
m_rtpBufferAudio->write((uint8_t *) &m_data[m_bufferIndex++]);
}
m_opusIn[m_codecInputIndex++] = sample;
}
break;
case CodecL16:
default:
m_rtpBufferAudio->write((uint8_t *) &sample);
break;
}
}
}
void AudioNetSink::write(qint16 ilSample, qint16 irSample)
{
qint16& lSample = ilSample;
qint16& rSample = irSample;
if (m_decimation > 1)
{
float lpLSample = m_audioFilter.runLP(lSample / 32768.0f);
float lpRSample = m_audioFilter.runLP(rSample / 32768.0f);
if (m_decimationCount >= m_decimation - 1)
{
lSample = lpLSample * 32768.0f;
rSample = lpRSample * 32768.0f;
m_decimationCount = 0;
}
else
{
m_decimationCount++;
return;
}
}
if (m_type == SinkUDP)
{
if (m_bufferIndex >= m_udpBlockSize)
{
m_udpSocket->writeDatagram((const char*) m_data, (qint64 ) m_udpBlockSize, m_address, m_port);
m_bufferIndex = 0;
}
switch(m_codec)
{
case CodecPCMA:
case CodecPCMU:
case CodecG722:
break; // mono modes - do nothing
case CodecOpus:
{
if (m_codecInputIndex == m_codecInputSize)
{
int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
nbBytes = nbBytes > m_udpBlockSize ? m_udpBlockSize : nbBytes;
m_udpSocket->writeDatagram((const char*) m_data, (qint64 ) nbBytes, m_address, m_port);
m_codecInputIndex = 0;
}
m_opusIn[2*m_codecInputIndex] = lSample;
m_opusIn[2*m_codecInputIndex+1] = rSample;
m_codecInputIndex++;
}
break;
case CodecL8:
{
qint8 *p = (qint8*) &m_data[m_bufferIndex];
*p = lSample / 256;
m_bufferIndex += sizeof(qint8);
p = (qint8*) &m_data[m_bufferIndex];
*p = rSample / 256;
m_bufferIndex += sizeof(qint8);
}
break;
case CodecL16:
default:
{
qint16 *p = (qint16*) &m_data[m_bufferIndex];
*p = lSample;
m_bufferIndex += sizeof(qint16);
p = (qint16*) &m_data[m_bufferIndex];
*p = rSample;
m_bufferIndex += sizeof(qint16);
}
break;
}
}
else if (m_type == SinkRTP)
{
switch(m_codec)
{
case CodecPCMA:
case CodecPCMU:
case CodecG722:
break; // mono modes - do nothing
case CodecOpus:
{
if (m_codecInputIndex == m_codecInputSize)
{
int nbBytes = m_opus.encode(m_codecInputSize, m_opusIn, (uint8_t *) m_data);
if (nbBytes != AudioOpus::m_bitrate/400) { // 8 bits for 1/50s (20ms)
qWarning("AudioNetSink::write: CodecOpus stereo: unexpected output frame size: %d bytes", nbBytes);
}
m_bufferIndex = 0;
m_codecInputIndex = 0;
}
if (m_codecInputIndex % m_codecRatio == 0) {
m_rtpBufferAudio->write((uint8_t *) &m_data[m_bufferIndex++]);
}
m_opusIn[2*m_codecInputIndex] = lSample;
m_opusIn[2*m_codecInputIndex+1] = rSample;
m_codecInputIndex++;
}
break;
case CodecL8:
{
qint8 pl = lSample / 256;
qint8 pr = rSample / 256;
m_rtpBufferAudio->write((uint8_t *) &pl, (uint8_t *) &pr);
}
break;
case CodecL16:
default:
m_rtpBufferAudio->write((uint8_t *) &lSample, (uint8_t *) &rSample);
break;
}
}
}
void AudioNetSink::moveToThread(QThread *thread)
{
m_udpSocket->moveToThread(thread);
}
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