xref: /dports/audio/jamulus/jamulus-r3_8_1/src/channel.cpp

/******************************************************************************\
 * Copyright (c) 2004-2020
 *
 * Author(s):
 *  Volker Fischer
 *
 ******************************************************************************
 *
 * 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; either version 2 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 for more
 * details.
 *
 * You should have received a copy of the GNU General Public License along with
 * this program; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301, USA
 *
\******************************************************************************/

#include "channel.h"

// CChannel implementation *****************************************************
CChannel::CChannel ( const bool bNIsServer ) :
    vecfGains ( MAX_NUM_CHANNELS, 1.0f ),
    vecfPannings ( MAX_NUM_CHANNELS, 0.5f ),
    iCurSockBufNumFrames ( INVALID_INDEX ),
    bDoAutoSockBufSize ( true ),
    bUseSequenceNumber ( false ), // this is important since in the client we reset on Channel.SetEnable ( false )
    iSendSequenceNumber ( 0 ),
    iFadeInCnt ( 0 ),
    iFadeInCntMax ( FADE_IN_NUM_FRAMES_DBLE_FRAMESIZE ),
    bIsEnabled ( false ),
    bIsServer ( bNIsServer ),
    bIsIdentified ( false ),
    iAudioFrameSizeSamples ( DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES ),
    SignalLevelMeter ( false, 0.5 ) // server mode with mono out and faster smoothing
{
    // reset network transport properties
    ResetNetworkTransportProperties();

    // initial value for connection time out counter, we calculate the total
    // number of samples here and subtract the number of samples of the block
    // which we take out of the buffer to be independent of block sizes
    iConTimeOutStartVal = CON_TIME_OUT_SEC_MAX * SYSTEM_SAMPLE_RATE_HZ;

    // init time-out for the buffer with zero -> no connection
    iConTimeOut = 0;

    // init the socket buffer
    SetSockBufNumFrames ( DEF_NET_BUF_SIZE_NUM_BL );

    // initialize channel info
    ResetInfo();

    // Connections -------------------------------------------------------------

    // clang-format off
// TODO if we later do not fire vectors in the emits, we can remove this again
qRegisterMetaType<CVector<uint8_t> > ( "CVector<uint8_t>" );
qRegisterMetaType<CHostAddress> ( "CHostAddress" );
    // clang-format on

    QObject::connect ( &Protocol, &CProtocol::MessReadyForSending, this, &CChannel::OnSendProtMessage );

    QObject::connect ( &Protocol, &CProtocol::ChangeJittBufSize, this, &CChannel::OnJittBufSizeChange );

    QObject::connect ( &Protocol, &CProtocol::ReqJittBufSize, this, &CChannel::ReqJittBufSize );

    QObject::connect ( &Protocol, &CProtocol::ReqChanInfo, this, &CChannel::ReqChanInfo );

    QObject::connect ( &Protocol, &CProtocol::ReqConnClientsList, this, &CChannel::ReqConnClientsList );

    QObject::connect ( &Protocol, &CProtocol::ConClientListMesReceived, this, &CChannel::ConClientListMesReceived );

    QObject::connect ( &Protocol, &CProtocol::ChangeChanGain, this, &CChannel::OnChangeChanGain );

    QObject::connect ( &Protocol, &CProtocol::ChangeChanPan, this, &CChannel::OnChangeChanPan );

    QObject::connect ( &Protocol, &CProtocol::ClientIDReceived, this, &CChannel::ClientIDReceived );

    QObject::connect ( &Protocol, &CProtocol::MuteStateHasChangedReceived, this, &CChannel::MuteStateHasChangedReceived );

    QObject::connect ( &Protocol, &CProtocol::ChangeChanInfo, this, &CChannel::OnChangeChanInfo );

    QObject::connect ( &Protocol, &CProtocol::ChatTextReceived, this, &CChannel::ChatTextReceived );

    QObject::connect ( &Protocol, &CProtocol::NetTranspPropsReceived, this, &CChannel::OnNetTranspPropsReceived );

    QObject::connect ( &Protocol, &CProtocol::ReqNetTranspProps, this, &CChannel::OnReqNetTranspProps );

    QObject::connect ( &Protocol, &CProtocol::ReqSplitMessSupport, this, &CChannel::OnReqSplitMessSupport );

    QObject::connect ( &Protocol, &CProtocol::SplitMessSupported, this, &CChannel::OnSplitMessSupported );

    QObject::connect ( &Protocol, &CProtocol::LicenceRequired, this, &CChannel::LicenceRequired );

    QObject::connect ( &Protocol, &CProtocol::VersionAndOSReceived, this, &CChannel::OnVersionAndOSReceived );

    QObject::connect ( &Protocol, &CProtocol::RecorderStateReceived, this, &CChannel::RecorderStateReceived );
}

bool CChannel::ProtocolIsEnabled()
{
    // for the server, only enable protocol if the channel is connected, i.e.,
    // successfully audio packets are received from a client
    // for the client, enable protocol if the channel is enabled, i.e., the
    // connection button was hit by the user
    if ( bIsServer )
    {
        return IsConnected();
    }
    else
    {
        return bIsEnabled;
    }
}

void CChannel::SetEnable ( const bool bNEnStat )
{
    QMutexLocker locker ( &Mutex );

    // set internal parameter
    bIsEnabled = bNEnStat;

    // The support for the packet sequence number must be reset if the client
    // disconnects from a server since we do not yet know if the next server we
    // connect to will support the sequence number. We use the SetEnable call in
    // the client for this task since at every start/stop it will call this
    // function. NOTE that it is important to reset this parameter on SetEnable(false)
    // since the SetEnable(true) is set AFTER the Init() in the client -> we
    // simply set it regardless of the state which does not hurt.
    bUseSequenceNumber = false;

    // if channel is not enabled, reset time out count and protocol
    if ( !bNEnStat )
    {
        iConTimeOut = 0;
        Protocol.Reset();
    }
}

void CChannel::OnVersionAndOSReceived ( COSUtil::EOpSystemType eOSType, QString strVersion )
{
    // check if audio packet counter is supported by the server (minimum version is 3.6.0)
#if QT_VERSION >= QT_VERSION_CHECK( 5, 6, 0 )
    if ( QVersionNumber::compare ( QVersionNumber::fromString ( strVersion ), QVersionNumber ( 3, 6, 0 ) ) >= 0 )
    {
        // activate sequence counter and update the audio stream properties (which
        // does all the initialization and tells the server about the change)
        bUseSequenceNumber = true;

        SetAudioStreamProperties ( eAudioCompressionType, iCeltNumCodedBytes, iNetwFrameSizeFact, iNumAudioChannels );
    }
#endif

    emit VersionAndOSReceived ( eOSType, strVersion );
}

void CChannel::SetAudioStreamProperties ( const EAudComprType eNewAudComprType,
                                          const int           iNewCeltNumCodedBytes,
                                          const int           iNewNetwFrameSizeFact,
                                          const int           iNewNumAudioChannels )
{
    /*
        this function is intended for the client (not the server)
    */
    CNetworkTransportProps NetworkTransportProps;

    Mutex.lock();
    {
        // store new values
        eAudioCompressionType = eNewAudComprType;
        iNumAudioChannels     = iNewNumAudioChannels;
        iCeltNumCodedBytes    = iNewCeltNumCodedBytes;
        iNetwFrameSizeFact    = iNewNetwFrameSizeFact;

        // add the size of the optional packet counter
        if ( bUseSequenceNumber )
        {
            iNetwFrameSize = iCeltNumCodedBytes + 1; // per definition 1 byte counter
        }
        else
        {
            iNetwFrameSize = iCeltNumCodedBytes;
        }

        // update audio frame size
        if ( eAudioCompressionType == CT_OPUS )
        {
            iAudioFrameSizeSamples = DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES;
        }
        else
        {
            iAudioFrameSizeSamples = SYSTEM_FRAME_SIZE_SAMPLES;
        }

        MutexSocketBuf.lock();
        {
            // init socket buffer
            SockBuf.SetUseDoubleSystemFrameSize ( eAudioCompressionType == CT_OPUS ); // NOTE must be set BEFORE the init()
            SockBuf.Init ( iCeltNumCodedBytes, iCurSockBufNumFrames, bUseSequenceNumber );
        }
        MutexSocketBuf.unlock();

        MutexConvBuf.lock();
        {
            // init conversion buffer
            ConvBuf.Init ( iNetwFrameSize * iNetwFrameSizeFact, bUseSequenceNumber );
        }
        MutexConvBuf.unlock();

        // fill network transport properties struct
        NetworkTransportProps = GetNetworkTransportPropsFromCurrentSettings();
    }
    Mutex.unlock();

    // tell the server about the new network settings
    Protocol.CreateNetwTranspPropsMes ( NetworkTransportProps );
}

bool CChannel::SetSockBufNumFrames ( const int iNewNumFrames, const bool bPreserve )
{
    bool ReturnValue           = true;  // init with error
    bool bCurDoAutoSockBufSize = false; // we have to init but init values does not matter

    // first check for valid input parameter range
    if ( ( iNewNumFrames >= MIN_NET_BUF_SIZE_NUM_BL ) && ( iNewNumFrames <= MAX_NET_BUF_SIZE_NUM_BL ) )
    {
        // only apply parameter if new parameter is different from current one
        if ( iCurSockBufNumFrames != iNewNumFrames )
        {
            MutexSocketBuf.lock();
            {
                // store new value
                iCurSockBufNumFrames = iNewNumFrames;

                // the network block size is a multiple of the minimum network
                // block size
                SockBuf.Init ( iCeltNumCodedBytes, iNewNumFrames, bUseSequenceNumber, bPreserve );

                // store current auto socket buffer size setting in the mutex
                // region since if we use the current parameter below in the
                // if condition, it may have been changed in between the time
                // when we have left the mutex region and entered the if
                // condition
                bCurDoAutoSockBufSize = bDoAutoSockBufSize;

                ReturnValue = false; // -> no error
            }
            MutexSocketBuf.unlock();
        }
    }

    // only in case there is no error, we are the server and auto jitter buffer
    // setting is enabled, we have to report the current setting to the client
    if ( !ReturnValue && bIsServer && bCurDoAutoSockBufSize )
    {
        // we cannot call the "CreateJitBufMes" function directly since
        // this would give us problems with different threads (e.g. the
        // timer thread) and the protocol mechanism (problem with
        // qRegisterMetaType(), etc.)
        emit ServerAutoSockBufSizeChange ( iNewNumFrames );
    }

    return ReturnValue; // set error flag
}

void CChannel::SetGain ( const int iChanID, const float fNewGain )
{
    QMutexLocker locker ( &Mutex );

    // set value (make sure channel ID is in range)
    if ( ( iChanID >= 0 ) && ( iChanID < MAX_NUM_CHANNELS ) )
    {
        // signal mute change
        if ( ( vecfGains[iChanID] == 0 ) && ( fNewGain > 0 ) )
        {
            emit MuteStateHasChanged ( iChanID, false );
        }
        if ( ( vecfGains[iChanID] > 0 ) && ( fNewGain == 0 ) )
        {
            emit MuteStateHasChanged ( iChanID, true );
        }

        vecfGains[iChanID] = fNewGain;
    }
}

float CChannel::GetGain ( const int iChanID )
{
    QMutexLocker locker ( &Mutex );

    // get value (make sure channel ID is in range)
    if ( ( iChanID >= 0 ) && ( iChanID < MAX_NUM_CHANNELS ) )
    {
        return vecfGains[iChanID];
    }
    else
    {
        return 0;
    }
}

void CChannel::SetPan ( const int iChanID, const float fNewPan )
{
    QMutexLocker locker ( &Mutex );

    // set value (make sure channel ID is in range)
    if ( ( iChanID >= 0 ) && ( iChanID < MAX_NUM_CHANNELS ) )
    {
        vecfPannings[iChanID] = fNewPan;
    }
}

float CChannel::GetPan ( const int iChanID )
{
    QMutexLocker locker ( &Mutex );

    // get value (make sure channel ID is in range)
    if ( ( iChanID >= 0 ) && ( iChanID < MAX_NUM_CHANNELS ) )
    {
        return vecfPannings[iChanID];
    }
    else
    {
        return 0;
    }
}

void CChannel::SetChanInfo ( const CChannelCoreInfo& NChanInf )
{
    bIsIdentified = true; // Indicate we have received channel info

    // apply value (if different from previous one)
    if ( ChannelInfo != NChanInf )
    {
        ChannelInfo = NChanInf;

        // fire message that the channel info has changed
        emit ChanInfoHasChanged();
    }
}

QString CChannel::GetName()
{
    // make sure the string is not written at the same time when it is
    // read here -> use mutex to secure access
    QMutexLocker locker ( &Mutex );

    return ChannelInfo.strName;
}

void CChannel::OnSendProtMessage ( CVector<uint8_t> vecMessage )
{
    // only send messages if protocol is enabled, otherwise delete complete
    // queue
    if ( ProtocolIsEnabled() )
    {
        // emit message to actually send the data
        emit MessReadyForSending ( vecMessage );
    }
    else
    {
        // delete send message queue
        Protocol.Reset();
    }
}

void CChannel::OnJittBufSizeChange ( int iNewJitBufSize )
{
    // for server apply setting, for client emit message
    if ( bIsServer )
    {
        // first check for special case: auto setting
        if ( iNewJitBufSize == AUTO_NET_BUF_SIZE_FOR_PROTOCOL )
        {
            SetDoAutoSockBufSize ( true );
        }
        else
        {
            // manual setting is received, turn OFF auto setting and apply new value
            SetDoAutoSockBufSize ( false );
            SetSockBufNumFrames ( iNewJitBufSize, true );
        }
    }
    else
    {
        emit JittBufSizeChanged ( iNewJitBufSize );
    }
}

void CChannel::OnChangeChanGain ( int iChanID, float fNewGain ) { SetGain ( iChanID, fNewGain ); }

void CChannel::OnChangeChanPan ( int iChanID, float fNewPan ) { SetPan ( iChanID, fNewPan ); }

void CChannel::OnChangeChanInfo ( CChannelCoreInfo ChanInfo ) { SetChanInfo ( ChanInfo ); }

void CChannel::OnNetTranspPropsReceived ( CNetworkTransportProps NetworkTransportProps )
{
    // only the server shall act on network transport properties message
    if ( bIsServer )
    {
        // OPUS and OPUS64 codecs are the only supported codecs right now
        if ( ( NetworkTransportProps.eAudioCodingType != CT_OPUS ) && ( NetworkTransportProps.eAudioCodingType != CT_OPUS64 ) )
        {
            Protocol.CreateOpusSupportedMes();
            return;
        }

        Mutex.lock();
        {
            // store received parameters
            eAudioCompressionType = NetworkTransportProps.eAudioCodingType;
            iNumAudioChannels     = static_cast<int> ( NetworkTransportProps.iNumAudioChannels );
            iNetwFrameSizeFact    = NetworkTransportProps.iBlockSizeFact;
            iNetwFrameSize        = static_cast<int> ( NetworkTransportProps.iBaseNetworkPacketSize );
            bUseSequenceNumber    = ( NetworkTransportProps.eFlags == NF_WITH_COUNTER );

            if ( bUseSequenceNumber )
            {
                iCeltNumCodedBytes = iNetwFrameSize - 1; // per definition 1 byte counter
            }
            else
            {
                iCeltNumCodedBytes = iNetwFrameSize;
            }

            // update maximum number of frames for fade in counter (only needed for server)
            // and audio frame size
            if ( eAudioCompressionType == CT_OPUS )
            {
                iFadeInCntMax          = FADE_IN_NUM_FRAMES_DBLE_FRAMESIZE / iNetwFrameSizeFact;
                iAudioFrameSizeSamples = DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES;
            }
            else
            {
                iFadeInCntMax          = FADE_IN_NUM_FRAMES / iNetwFrameSizeFact;
                iAudioFrameSizeSamples = SYSTEM_FRAME_SIZE_SAMPLES;
            }

            // the fade-in counter maximum value may have changed, make sure the fade-in counter
            // is not larger than the allowed maximum value
            iFadeInCnt = std::min ( iFadeInCnt, iFadeInCntMax );

            MutexSocketBuf.lock();
            {
                // update socket buffer (the network block size is a multiple of the
                // minimum network frame size)
                SockBuf.SetUseDoubleSystemFrameSize ( eAudioCompressionType == CT_OPUS ); // NOTE must be set BEFORE the init()
                SockBuf.Init ( iCeltNumCodedBytes, iCurSockBufNumFrames, bUseSequenceNumber );
            }
            MutexSocketBuf.unlock();

            MutexConvBuf.lock();
            {
                // init conversion buffer
                ConvBuf.Init ( iNetwFrameSize * iNetwFrameSizeFact, bUseSequenceNumber );
            }
            MutexConvBuf.unlock();
        }
        Mutex.unlock();
    }
}

void CChannel::OnReqNetTranspProps()
{
    // fill network transport properties struct from current settings and send it
    Protocol.CreateNetwTranspPropsMes ( GetNetworkTransportPropsFromCurrentSettings() );
}

void CChannel::OnReqSplitMessSupport()
{
    // activate split messages in our protocol (client) and return answer message to the server
    Protocol.SetSplitMessageSupported ( true );
    Protocol.CreateSplitMessSupportedMes();
}

CNetworkTransportProps CChannel::GetNetworkTransportPropsFromCurrentSettings()
{
    // set network flags
    ENetwFlags eFlags = NF_NONE;

    if ( bUseSequenceNumber )
    {
        eFlags = NF_WITH_COUNTER;
    }

    // use current stored settings of the channel to fill the network transport
    // properties structure
    return CNetworkTransportProps ( static_cast<uint32_t> ( iNetwFrameSize ),
                                    static_cast<uint16_t> ( iNetwFrameSizeFact ),
                                    static_cast<uint32_t> ( iNumAudioChannels ),
                                    SYSTEM_SAMPLE_RATE_HZ,
                                    eAudioCompressionType,
                                    eFlags,
                                    0 );
}

void CChannel::Disconnect()
{
    // we only have to disconnect the channel if it is actually connected
    if ( IsConnected() )
    {
        // set time out counter to a small value > 0 so that the next time a
        // received audio block is queried, the disconnection is performed
        // (assuming that no audio packet is received in the meantime)
        iConTimeOut = 1; // a small number > 0
    }
}

void CChannel::PutProtcolData ( const int iRecCounter, const int iRecID, const CVector<uint8_t>& vecbyMesBodyData, const CHostAddress& RecHostAddr )
{
    // Only process protocol message if:
    // - for client only: the packet comes from the server we want to talk to
    // - the channel is enabled
    // - the protocol mechanism is enabled
    if ( ( bIsServer || ( GetAddress() == RecHostAddr ) ) && IsEnabled() && ProtocolIsEnabled() )
    {
        // parse the message assuming this is a regular protocol message
        Protocol.ParseMessageBody ( vecbyMesBodyData, iRecCounter, iRecID );
    }
}

EPutDataStat CChannel::PutAudioData ( const CVector<uint8_t>& vecbyData, const int iNumBytes, CHostAddress RecHostAddr )
{
    // init return state
    EPutDataStat eRet = PS_GEN_ERROR;

    // Only process audio data if:
    // - for client only: the packet comes from the server we want to talk to
    // - the channel is enabled
    if ( ( bIsServer || ( GetAddress() == RecHostAddr ) ) && IsEnabled() )
    {
        MutexSocketBuf.lock();
        {
            // only process audio if packet has correct size
            if ( iNumBytes == ( iNetwFrameSize * iNetwFrameSizeFact ) )
            {
                // store new packet in jitter buffer
                if ( SockBuf.Put ( vecbyData, iNumBytes ) )
                {
                    eRet = PS_AUDIO_OK;
                }
                else
                {
                    eRet = PS_AUDIO_ERR;
                }

                // manage audio fade-in counter, after channel is identified
                if ( iFadeInCnt < iFadeInCntMax && bIsIdentified )
                {
                    iFadeInCnt++;
                }
            }
            else
            {
                // the protocol parsing failed and this was no audio block,
                // we treat this as protocol error (unknown packet)
                eRet = PS_PROT_ERR;
            }

            // All network packets except of valid protocol messages
            // regardless if they are valid or invalid audio packets lead to
            // a state change to a connected channel.
            // This is because protocol messages can only be sent on a
            // connected channel and the client has to inform the server
            // about the audio packet properties via the protocol.

            // check if channel was not connected, this is a new connection
            if ( !IsConnected() )
            {
                // overwrite status
                eRet = PS_NEW_CONNECTION;

                // init audio fade-in counter
                iFadeInCnt = 0;

                // init level meter
                SignalLevelMeter.Reset();
            }

            // reset time-out counter (note that this must be done after the
            // "IsConnected()" query above)
            ResetTimeOutCounter();
        }
        MutexSocketBuf.unlock();
    }
    else
    {
        eRet = PS_AUDIO_INVALID;
    }

    return eRet;
}

EGetDataStat CChannel::GetData ( CVector<uint8_t>& vecbyData, const int iNumBytes )
{
    EGetDataStat eGetStatus;

    MutexSocketBuf.lock();
    {
        // the socket access must be inside a mutex
        const bool bSockBufState = SockBuf.Get ( vecbyData, iNumBytes );

        // decrease time-out counter
        if ( iConTimeOut > 0 )
        {
            // subtract the number of samples of the current block since the
            // time out counter is based on samples not on blocks (definition:
            // always one atomic block is get by using the GetData() function
            // where the atomic block size is "iAudioFrameSizeSamples")
            iConTimeOut -= iAudioFrameSizeSamples;

            if ( iConTimeOut <= 0 )
            {
                // channel is just disconnected
                eGetStatus  = GS_CHAN_NOW_DISCONNECTED;
                iConTimeOut = 0; // make sure we do not have negative values

                // reset network transport properties
                ResetNetworkTransportProperties();
            }
            else
            {
                if ( bSockBufState )
                {
                    // everything is ok
                    eGetStatus = GS_BUFFER_OK;
                }
                else
                {
                    // channel is not yet disconnected but no data in buffer
                    eGetStatus = GS_BUFFER_UNDERRUN;
                }
            }
        }
        else
        {
            // channel is disconnected
            eGetStatus = GS_CHAN_NOT_CONNECTED;
        }
    }
    MutexSocketBuf.unlock();

    // in case we are just disconnected, we have to fire a message
    if ( eGetStatus == GS_CHAN_NOW_DISCONNECTED )
    {
        // reset the protocol
        Protocol.Reset();

        // emit message
        emit Disconnected();
    }

    return eGetStatus;
}

void CChannel::PrepAndSendPacket ( CHighPrioSocket* pSocket, const CVector<uint8_t>& vecbyNPacket, const int iNPacketLen )
{
    // From v3.8.0 onwards, a server will not send audio to a client until that client has sent channel info.
    // This addresses #1243 but means that clients earlier than v3.3.0 (24 Feb 2013) will no longer be compatible.
    if ( bIsServer && !bIsIdentified )
    {
        return;
    }

    QMutexLocker locker ( &MutexConvBuf );

    // use conversion buffer to convert sound card block size in network
    // block size and take care of optional sequence number (note that
    // the sequence number wraps automatically)
    if ( ConvBuf.Put ( vecbyNPacket, iNPacketLen, iSendSequenceNumber++ ) )
    {
        pSocket->SendPacket ( ConvBuf.GetAll(), GetAddress() );
    }
}

double CChannel::UpdateAndGetLevelForMeterdB ( const CVector<short>& vecsAudio, const int iInSize, const bool bIsStereoIn )
{
    // update the signal level meter and immediately return the current value
    SignalLevelMeter.Update ( vecsAudio, iInSize, bIsStereoIn );

    return SignalLevelMeter.GetLevelForMeterdBLeftOrMono();
}

int CChannel::GetUploadRateKbps()
{
    const int iAudioSizeOut = iNetwFrameSizeFact * iAudioFrameSizeSamples;

    // we assume that the UDP packet which is transported via IP has an
    // additional header size of ("Network Music Performance (NMP) in narrow
    // band networks; Carot, Kraemer, Schuller; 2006")
    // 8 (UDP) + 20 (IP without optional fields) = 28 bytes
    // 2 (PPP) + 6 (PPPoE) + 18 (MAC)            = 26 bytes
    // 5 (RFC1483B) + 8 (AAL) + 10 (ATM)         = 23 bytes
    return ( iNetwFrameSize * iNetwFrameSizeFact + 28 + 26 + 23 /* header */ ) * 8 /* bits per byte */ * SYSTEM_SAMPLE_RATE_HZ / iAudioSizeOut / 1000;
}

void CChannel::UpdateSocketBufferSize()
{
    // just update the socket buffer size if auto setting is enabled, otherwise
    // do nothing
    if ( bDoAutoSockBufSize )
    {
        // use auto setting result from channel, make sure we preserve the
        // buffer memory since we just adjust the size here
        SetSockBufNumFrames ( SockBuf.GetAutoSetting(), true );
    }
}