AudioNetSink Class Reference

#include <audionetsink.h>

Collaboration diagram for AudioNetSink:

Public Types
enum	SinkType { SinkUDP, SinkRTP }
enum	Codec {   CodecL16, CodecL8, CodecPCMA, CodecPCMU,   CodecG722, CodecOpus }

Public Member Functions
	AudioNetSink (QObject *parent)
	without RTP More...
	AudioNetSink (QObject *parent, int sampleRate, bool stereo)
	with RTP More...
	~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 Public Attributes
static const int	m_udpBlockSize = 512
static const int	m_dataBlockSize = 65536
static const int	m_g722BlockSize = 12800
static const int	m_opusBlockSize = 960*2
static const int	m_opusOutputSize = 160

Protected Member Functions
void	setNewCodecData ()
void	setDecimationFilters ()

Protected Attributes
SinkType	m_type
Codec	m_codec
QUdpSocket *	m_udpSocket
RTPSink *	m_rtpBufferAudio
AudioCompressor	m_audioCompressor
AudioG722	m_g722
AudioOpus	m_opus
AudioFilter	m_audioFilter
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
int	m_codecInputIndex
int	m_codecRatio
unsigned int	m_bufferIndex
QHostAddress	m_address
unsigned int	m_port

Detailed Description

Definition at line 37 of file audionetsink.h.

Member Enumeration Documentation

Codec

enum AudioNetSink::Codec

Enumerator
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.

Definition at line 45 of file audionetsink.h.

    {
        CodecL16,  
        CodecL8,   
        CodecPCMA, 
        CodecPCMU, 
        CodecG722, 
        CodecOpus  
    } Codec;

SinkType

enum AudioNetSink::SinkType

Enumerator
SinkUDP
SinkRTP

Definition at line 39 of file audionetsink.h.

    {
        SinkUDP,
        SinkRTP
    } SinkType;

Constructor & Destructor Documentation

AudioNetSink() [1/2]

AudioNetSink::AudioNetSink

(

QObject *

parent

)

without RTP

Definition at line 29 of file audionetsink.cpp.

References AudioOpus::m_bitrate, m_codecRatio, m_data, m_dataBlockSize, m_decimation, m_opusBlockSize, m_opusIn, m_sampleRate, and m_udpSocket.

                                          :
    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() [2/2]

AudioNetSink::AudioNetSink	(	QObject *	parent,
		int	sampleRate,
		bool	stereo
	)

with RTP

Definition at line 48 of file audionetsink.cpp.

References AudioOpus::m_bitrate, m_codecRatio, m_data, m_dataBlockSize, m_decimation, m_opusBlockSize, m_opusIn, m_rtpBufferAudio, m_sampleRate, and m_udpSocket.

                                                                       :
    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::~AudioNetSink

(

)

Definition at line 68 of file audionetsink.cpp.

References m_rtpBufferAudio, and m_udpSocket.

{
    if (m_rtpBufferAudio) {
        delete m_rtpBufferAudio;
    }

    m_udpSocket->deleteLater(); // this thread is not the owner thread (was moved)
}

Member Function Documentation

addDestination()

void AudioNetSink::addDestination	(	const QString &	address,
		uint16_t	port
	)

Definition at line 106 of file audionetsink.cpp.

References RTPSink::addDestination(), and m_rtpBufferAudio.

{
    if (m_rtpBufferAudio) {
        m_rtpBufferAudio->addDestination(address, port);
    }
}

Here is the call graph for this function:

deleteDestination()

void AudioNetSink::deleteDestination	(	const QString &	address,
		uint16_t	port
	)

Definition at line 113 of file audionetsink.cpp.

References RTPSink::deleteDestination(), and m_rtpBufferAudio.

{
    if (m_rtpBufferAudio) {
        m_rtpBufferAudio->deleteDestination(address, port);
    }
}

Here is the call graph for this function:

isRTPCapable()

bool AudioNetSink::isRTPCapable

(

)

const

Definition at line 77 of file audionetsink.cpp.

References RTPSink::isValid(), and m_rtpBufferAudio.

{
    return m_rtpBufferAudio && m_rtpBufferAudio->isValid();
}

Here is the call graph for this function:

moveToThread()

void AudioNetSink::moveToThread

(

QThread *

thread

)

Definition at line 483 of file audionetsink.cpp.

References m_udpSocket.

{
    m_udpSocket->moveToThread(thread);
}

selectType()

bool AudioNetSink::selectType

(

SinkType

type

)

Definition at line 82 of file audionetsink.cpp.

References m_type, SinkRTP, and SinkUDP.

Referenced by AudioOutput::setUdpUseRTP().

{
    if (type == SinkUDP)
    {
        m_type = SinkUDP;
    }
    else // this is SinkRTP
    {
        m_type = SinkRTP;
    }

    return true;
}

Here is the caller graph for this function:

setDecimation()

void AudioNetSink::setDecimation

(

uint32_t

decimation

)

Definition at line 162 of file audionetsink.cpp.

References decimation(), m_decimation, m_decimationCount, and setNewCodecData().

Referenced by AudioOutput::setUdpDecimation().

{
    m_decimation = decimation < 1 ? 1 : decimation > 6 ? 6 : decimation;
    qDebug() << "AudioNetSink::setDecimation: " << m_decimation << " from: " << decimation;
    setNewCodecData();
    m_decimationCount = 0;
}

Here is the call graph for this function:

Here is the caller graph for this function:

setDecimationFilters()

void AudioNetSink::setDecimationFilters ( )

protected

Definition at line 190 of file audionetsink.cpp.

References CodecG722, CodecL16, CodecL8, CodecOpus, CodecPCMA, CodecPCMU, m_audioFilter, m_codec, m_decimation, m_sampleRate, and AudioFilter::setDecimFilters().

Referenced by setNewCodecData().

{
    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;
    }
}

Here is the call graph for this function:

Here is the caller graph for this function:

setDestination()

void AudioNetSink::setDestination	(	const QString &	address,
		uint16_t	port
	)

Definition at line 96 of file audionetsink.cpp.

References m_address, m_port, m_rtpBufferAudio, and RTPSink::setDestination().

Referenced by AudioOutput::setUdpDestination().

{
    m_address.setAddress(const_cast<QString&>(address));
    m_port = port;

    if (m_rtpBufferAudio) {
        m_rtpBufferAudio->setDestination(address, port);
    }
}

Here is the call graph for this function:

Here is the caller graph for this function:

setNewCodecData()

void AudioNetSink::setNewCodecData ( )

protected

Definition at line 170 of file audionetsink.cpp.

References CodecOpus, AudioOpus::m_bitrate, m_bufferIndex, m_codec, m_codecInputIndex, m_codecInputSize, m_codecRatio, m_decimation, m_opus, m_sampleRate, m_stereo, setDecimationFilters(), and AudioOpus::setEncoder().

Referenced by setDecimation(), and setParameters().

{
    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();
}

Here is the call graph for this function:

Here is the caller graph for this function:

setParameters()

void AudioNetSink::setParameters	(	Codec	codec,
		bool	stereo,
		int	sampleRate
	)

Definition at line 120 of file audionetsink.cpp.

References CodecG722, CodecL16, CodecL8, CodecOpus, CodecPCMA, CodecPCMU, AudioCompressor::fillALaw(), AudioCompressor::fillULaw(), m_audioCompressor, m_codec, m_rtpBufferAudio, m_sampleRate, m_stereo, RTPSink::PayloadG722, RTPSink::PayloadL16Mono, RTPSink::PayloadL16Stereo, RTPSink::PayloadL8, RTPSink::PayloadOpus, RTPSink::PayloadPCMA8, RTPSink::PayloadPCMU8, setNewCodecData(), and RTPSink::setPayloadInformation().

Referenced by AudioOutput::setUdpChannelFormat().

{
    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;
        }
    }
}

Here is the call graph for this function:

Here is the caller graph for this function:

write() [1/2]

void AudioNetSink::write

(

qint16

sample

)

Definition at line 212 of file audionetsink.cpp.

References CodecG722, CodecL16, CodecL8, CodecOpus, CodecPCMA, CodecPCMU, AudioCompressor::compress8(), AudioG722::encode(), AudioOpus::encode(), m_address, m_audioCompressor, m_audioFilter, AudioOpus::m_bitrate, m_bufferIndex, m_codec, m_codecInputIndex, m_codecInputSize, m_codecRatio, m_data, m_decimation, m_decimationCount, m_g722, m_g722BlockSize, m_opus, m_opusIn, m_port, m_rtpBufferAudio, m_type, m_udpBlockSize, m_udpSocket, AudioFilter::run(), SinkRTP, SinkUDP, and RTPSink::write().

Referenced by AudioOutput::readData().

{
    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;
        }
    }
}

Here is the call graph for this function:

Here is the caller graph for this function:

write() [2/2]

void AudioNetSink::write	(	qint16	lSample,
		qint16	rSample
	)

Definition at line 364 of file audionetsink.cpp.

References CodecG722, CodecL16, CodecL8, CodecOpus, CodecPCMA, CodecPCMU, AudioOpus::encode(), m_address, m_audioFilter, AudioOpus::m_bitrate, m_bufferIndex, m_codec, m_codecInputIndex, m_codecInputSize, m_codecRatio, m_data, m_decimation, m_decimationCount, m_opus, m_opusIn, m_port, m_rtpBufferAudio, m_type, m_udpBlockSize, m_udpSocket, AudioFilter::runLP(), SinkRTP, SinkUDP, and RTPSink::write().

{
    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;
        }
    }
}

Here is the call graph for this function:

Member Data Documentation

m_address

QHostAddress AudioNetSink::m_address

protected

Definition at line 101 of file audionetsink.h.

Referenced by setDestination(), and write().

m_audioCompressor

AudioCompressor AudioNetSink::m_audioCompressor

protected

Definition at line 87 of file audionetsink.h.

Referenced by setParameters(), and write().

m_audioFilter

AudioFilter AudioNetSink::m_audioFilter

protected

Definition at line 90 of file audionetsink.h.

Referenced by setDecimationFilters(), and write().

m_bufferIndex

unsigned int AudioNetSink::m_bufferIndex

protected

Definition at line 100 of file audionetsink.h.

Referenced by setNewCodecData(), and write().

m_codec

Codec AudioNetSink::m_codec

protected

Definition at line 84 of file audionetsink.h.

Referenced by setDecimationFilters(), setNewCodecData(), setParameters(), and write().

m_codecInputIndex

int AudioNetSink::m_codecInputIndex

protected

Definition at line 98 of file audionetsink.h.

Referenced by setNewCodecData(), and write().

m_codecInputSize

int AudioNetSink::m_codecInputSize

protected

Definition at line 97 of file audionetsink.h.

Referenced by setNewCodecData(), and write().

m_codecRatio

int AudioNetSink::m_codecRatio

protected

Definition at line 99 of file audionetsink.h.

Referenced by AudioNetSink(), setNewCodecData(), and write().

m_data

char AudioNetSink::m_data[m_dataBlockSize]

protected

Definition at line 95 of file audionetsink.h.

Referenced by AudioNetSink(), and write().

m_dataBlockSize

const int AudioNetSink::m_dataBlockSize = 65536

static

Definition at line 74 of file audionetsink.h.

Referenced by AudioNetSink().

m_decimation

uint32_t AudioNetSink::m_decimation

protected

Definition at line 93 of file audionetsink.h.

Referenced by AudioNetSink(), setDecimation(), setDecimationFilters(), setNewCodecData(), and write().

m_decimationCount

uint32_t AudioNetSink::m_decimationCount

protected

Definition at line 94 of file audionetsink.h.

Referenced by setDecimation(), and write().

m_g722

AudioG722 AudioNetSink::m_g722

protected

Definition at line 88 of file audionetsink.h.

Referenced by write().

m_g722BlockSize

const int AudioNetSink::m_g722BlockSize = 12800

static

Definition at line 75 of file audionetsink.h.

Referenced by write().

m_opus

AudioOpus AudioNetSink::m_opus

protected

Definition at line 89 of file audionetsink.h.

Referenced by setNewCodecData(), and write().

m_opusBlockSize

const int AudioNetSink::m_opusBlockSize = 960*2

static

Definition at line 76 of file audionetsink.h.

Referenced by AudioNetSink().

m_opusIn

int16_t AudioNetSink::m_opusIn[m_opusBlockSize]

protected

Definition at line 96 of file audionetsink.h.

Referenced by AudioNetSink(), and write().

m_opusOutputSize

const int AudioNetSink::m_opusOutputSize = 160

static

Definition at line 77 of file audionetsink.h.

m_port

unsigned int AudioNetSink::m_port

protected

Definition at line 102 of file audionetsink.h.

Referenced by setDestination(), and write().

m_rtpBufferAudio

RTPSink* AudioNetSink::m_rtpBufferAudio

protected

Definition at line 86 of file audionetsink.h.

Referenced by addDestination(), AudioNetSink(), deleteDestination(), isRTPCapable(), setDestination(), setParameters(), write(), and ~AudioNetSink().

m_sampleRate

int AudioNetSink::m_sampleRate

protected

Definition at line 91 of file audionetsink.h.

Referenced by AudioNetSink(), setDecimationFilters(), setNewCodecData(), and setParameters().

m_stereo

bool AudioNetSink::m_stereo

protected

Definition at line 92 of file audionetsink.h.

Referenced by setNewCodecData(), and setParameters().

m_type

SinkType AudioNetSink::m_type

protected

Definition at line 83 of file audionetsink.h.

Referenced by selectType(), and write().

m_udpBlockSize

const int AudioNetSink::m_udpBlockSize = 512

static

Definition at line 73 of file audionetsink.h.

Referenced by write().

m_udpSocket

QUdpSocket* AudioNetSink::m_udpSocket

protected

Definition at line 85 of file audionetsink.h.

Referenced by AudioNetSink(), moveToThread(), write(), and ~AudioNetSink().

---

The documentation for this class was generated from the following files:

• sdrbase/audio/audionetsink.h
• sdrbase/audio/audionetsink.cpp