/******************************************************************************\
* 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 "client.h"
/* Implementation *************************************************************/
CClient::CClient ( const quint16 iPortNumber,
const QString& strConnOnStartupAddress,
const int iCtrlMIDIChannel,
const bool bNoAutoJackConnect,
const QString& strNClientName ) :
vstrIPAddress ( MAX_NUM_SERVER_ADDR_ITEMS, "" ),
ChannelInfo (),
vecStoredFaderTags ( MAX_NUM_STORED_FADER_SETTINGS, "" ),
vecStoredFaderLevels ( MAX_NUM_STORED_FADER_SETTINGS, AUD_MIX_FADER_MAX ),
vecStoredPanValues ( MAX_NUM_STORED_FADER_SETTINGS, AUD_MIX_PAN_MAX / 2 ),
vecStoredFaderIsSolo ( MAX_NUM_STORED_FADER_SETTINGS, false ),
vecStoredFaderIsMute ( MAX_NUM_STORED_FADER_SETTINGS, false ),
iNewClientFaderLevel ( 100 ),
bConnectDlgShowAllMusicians ( true ),
strClientName ( strNClientName ),
vecWindowPosMain (), // empty array
vecWindowPosSettings (), // empty array
vecWindowPosChat (), // empty array
vecWindowPosProfile (), // empty array
vecWindowPosConnect (), // empty array
bWindowWasShownSettings ( false ),
bWindowWasShownChat ( false ),
bWindowWasShownProfile ( false ),
bWindowWasShownConnect ( false ),
Channel ( false ), /* we need a client channel -> "false" */
CurOpusEncoder ( nullptr ),
CurOpusDecoder ( nullptr ),
eAudioCompressionType ( CT_OPUS ),
iCeltNumCodedBytes ( OPUS_NUM_BYTES_MONO_LOW_QUALITY ),
iOPUSFrameSizeSamples ( DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES ),
eAudioQuality ( AQ_NORMAL ),
eAudioChannelConf ( CC_MONO ),
iNumAudioChannels ( 1 ),
bIsInitializationPhase ( true ),
bMuteOutStream ( false ),
dMuteOutStreamGain ( 1.0 ),
Socket ( &Channel, iPortNumber ),
Sound ( AudioCallback, this, iCtrlMIDIChannel, bNoAutoJackConnect, strNClientName ),
iAudioInFader ( AUD_FADER_IN_MIDDLE ),
bReverbOnLeftChan ( false ),
iReverbLevel ( 0 ),
iSndCrdPrefFrameSizeFactor ( FRAME_SIZE_FACTOR_DEFAULT ),
iSndCrdFrameSizeFactor ( FRAME_SIZE_FACTOR_DEFAULT ),
bSndCrdConversionBufferRequired ( false ),
iSndCardMonoBlockSizeSamConvBuff ( 0 ),
bFraSiFactPrefSupported ( false ),
bFraSiFactDefSupported ( false ),
bFraSiFactSafeSupported ( false ),
eGUIDesign ( GD_ORIGINAL ),
bDisplayChannelLevels ( true ),
bEnableOPUS64 ( false ),
bJitterBufferOK ( true ),
strCentralServerAddress ( "" ),
eCentralServerAddressType ( AT_DEFAULT ),
iServerSockBufNumFrames ( DEF_NET_BUF_SIZE_NUM_BL ),
pSignalHandler ( CSignalHandler::getSingletonP() )
{
int iOpusError;
OpusMode = opus_custom_mode_create ( SYSTEM_SAMPLE_RATE_HZ,
DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES,
&iOpusError );
Opus64Mode = opus_custom_mode_create ( SYSTEM_SAMPLE_RATE_HZ,
SYSTEM_FRAME_SIZE_SAMPLES,
&iOpusError );
// init audio encoders and decoders
OpusEncoderMono = opus_custom_encoder_create ( OpusMode, 1, &iOpusError ); // mono encoder legacy
OpusDecoderMono = opus_custom_decoder_create ( OpusMode, 1, &iOpusError ); // mono decoder legacy
OpusEncoderStereo = opus_custom_encoder_create ( OpusMode, 2, &iOpusError ); // stereo encoder legacy
OpusDecoderStereo = opus_custom_decoder_create ( OpusMode, 2, &iOpusError ); // stereo decoder legacy
Opus64EncoderMono = opus_custom_encoder_create ( Opus64Mode, 1, &iOpusError ); // mono encoder OPUS64
Opus64DecoderMono = opus_custom_decoder_create ( Opus64Mode, 1, &iOpusError ); // mono decoder OPUS64
Opus64EncoderStereo = opus_custom_encoder_create ( Opus64Mode, 2, &iOpusError ); // stereo encoder OPUS64
Opus64DecoderStereo = opus_custom_decoder_create ( Opus64Mode, 2, &iOpusError ); // stereo decoder OPUS64
// we require a constant bit rate
opus_custom_encoder_ctl ( OpusEncoderMono, OPUS_SET_VBR ( 0 ) );
opus_custom_encoder_ctl ( OpusEncoderStereo, OPUS_SET_VBR ( 0 ) );
opus_custom_encoder_ctl ( Opus64EncoderMono, OPUS_SET_VBR ( 0 ) );
opus_custom_encoder_ctl ( Opus64EncoderStereo, OPUS_SET_VBR ( 0 ) );
// for 64 samples frame size we have to adjust the PLC behavior to avoid loud artifacts
opus_custom_encoder_ctl ( Opus64EncoderMono, OPUS_SET_PACKET_LOSS_PERC ( 35 ) );
opus_custom_encoder_ctl ( Opus64EncoderStereo, OPUS_SET_PACKET_LOSS_PERC ( 35 ) );
// we want as low delay as possible
opus_custom_encoder_ctl ( OpusEncoderMono, OPUS_SET_APPLICATION ( OPUS_APPLICATION_RESTRICTED_LOWDELAY ) );
opus_custom_encoder_ctl ( OpusEncoderStereo, OPUS_SET_APPLICATION ( OPUS_APPLICATION_RESTRICTED_LOWDELAY ) );
opus_custom_encoder_ctl ( Opus64EncoderMono, OPUS_SET_APPLICATION ( OPUS_APPLICATION_RESTRICTED_LOWDELAY ) );
opus_custom_encoder_ctl ( Opus64EncoderStereo, OPUS_SET_APPLICATION ( OPUS_APPLICATION_RESTRICTED_LOWDELAY ) );
// set encoder low complexity for legacy 128 samples frame size
opus_custom_encoder_ctl ( OpusEncoderMono, OPUS_SET_COMPLEXITY ( 1 ) );
opus_custom_encoder_ctl ( OpusEncoderStereo, OPUS_SET_COMPLEXITY ( 1 ) );
// Connections -------------------------------------------------------------
// connections for the protocol mechanism
QObject::connect ( &Channel, &CChannel::MessReadyForSending,
this, &CClient::OnSendProtMessage );
QObject::connect ( &Channel, &CChannel::DetectedCLMessage,
this, &CClient::OnDetectedCLMessage );
QObject::connect ( &Channel, &CChannel::ReqJittBufSize,
this, &CClient::OnReqJittBufSize );
QObject::connect ( &Channel, &CChannel::JittBufSizeChanged,
this, &CClient::OnJittBufSizeChanged );
QObject::connect ( &Channel, &CChannel::ReqChanInfo,
this, &CClient::OnReqChanInfo );
QObject::connect ( &Channel, &CChannel::ConClientListMesReceived,
this, &CClient::ConClientListMesReceived );
QObject::connect ( &Channel, &CChannel::Disconnected,
this, &CClient::Disconnected );
QObject::connect ( &Channel, &CChannel::NewConnection,
this, &CClient::OnNewConnection );
QObject::connect ( &Channel, &CChannel::ChatTextReceived,
this, &CClient::ChatTextReceived );
QObject::connect ( &Channel, &CChannel::ClientIDReceived,
this, &CClient::ClientIDReceived );
QObject::connect ( &Channel, &CChannel::MuteStateHasChangedReceived,
this, &CClient::MuteStateHasChangedReceived );
QObject::connect ( &Channel, &CChannel::LicenceRequired,
this, &CClient::LicenceRequired );
QObject::connect ( &Channel, &CChannel::VersionAndOSReceived,
this, &CClient::VersionAndOSReceived );
QObject::connect ( &Channel, &CChannel::RecorderStateReceived,
this, &CClient::RecorderStateReceived );
QObject::connect ( &ConnLessProtocol, &CProtocol::CLMessReadyForSending,
this, &CClient::OnSendCLProtMessage );
QObject::connect ( &ConnLessProtocol, &CProtocol::CLServerListReceived,
this, &CClient::CLServerListReceived );
QObject::connect ( &ConnLessProtocol, &CProtocol::CLConnClientsListMesReceived,
this, &CClient::CLConnClientsListMesReceived );
QObject::connect ( &ConnLessProtocol, &CProtocol::CLPingReceived,
this, &CClient::OnCLPingReceived );
QObject::connect ( &ConnLessProtocol, &CProtocol::CLPingWithNumClientsReceived,
this, &CClient::OnCLPingWithNumClientsReceived );
QObject::connect ( &ConnLessProtocol, &CProtocol::CLDisconnection ,
this, &CClient::OnCLDisconnection );
#ifdef ENABLE_CLIENT_VERSION_AND_OS_DEBUGGING
QObject::connect ( &ConnLessProtocol, &CProtocol::CLVersionAndOSReceived,
this, &CClient::CLVersionAndOSReceived );
#endif
QObject::connect ( &ConnLessProtocol, &CProtocol::CLChannelLevelListReceived,
this, &CClient::CLChannelLevelListReceived );
// other
QObject::connect ( &Sound, &CSound::ReinitRequest,
this, &CClient::OnSndCrdReinitRequest );
QObject::connect ( &Sound, &CSound::ControllerInFaderLevel,
this, &CClient::ControllerInFaderLevel );
QObject::connect ( &Socket, &CHighPrioSocket::InvalidPacketReceived,
this, &CClient::OnInvalidPacketReceived );
QObject::connect ( pSignalHandler, &CSignalHandler::HandledSignal,
this, &CClient::OnHandledSignal );
// start the socket (it is important to start the socket after all
// initializations and connections)
Socket.Start();
// do an immediate start if a server address is given
if ( !strConnOnStartupAddress.isEmpty() )
{
SetServerAddr ( strConnOnStartupAddress );
Start();
}
}
void CClient::OnSendProtMessage ( CVector<uint8_t> vecMessage )
{
// the protocol queries me to call the function to send the message
// send it through the network
Socket.SendPacket ( vecMessage, Channel.GetAddress() );
}
void CClient::OnSendCLProtMessage ( CHostAddress InetAddr,
CVector<uint8_t> vecMessage )
{
// the protocol queries me to call the function to send the message
// send it through the network
Socket.SendPacket ( vecMessage, InetAddr );
}
void CClient::OnInvalidPacketReceived ( CHostAddress RecHostAddr )
{
// message coult not be parsed, check if the packet comes
// from the server we just connected -> if yes, send
// disconnect message since the server may not know that we
// are not connected anymore
if ( Channel.GetAddress() == RecHostAddr )
{
ConnLessProtocol.CreateCLDisconnection ( RecHostAddr );
}
}
void CClient::OnDetectedCLMessage ( CVector<uint8_t> vecbyMesBodyData,
int iRecID,
CHostAddress RecHostAddr )
{
// connection less messages are always processed
ConnLessProtocol.ParseConnectionLessMessageBody ( vecbyMesBodyData,
iRecID,
RecHostAddr );
}
void CClient::OnJittBufSizeChanged ( int iNewJitBufSize )
{
// we received a jitter buffer size changed message from the server,
// only apply this value if auto jitter buffer size is enabled
if ( GetDoAutoSockBufSize() )
{
// Note: Do not use the "SetServerSockBufNumFrames" function for setting
// the new server jitter buffer size since then a message would be sent
// to the server which is incorrect.
iServerSockBufNumFrames = iNewJitBufSize;
}
}
void CClient::OnNewConnection()
{
// a new connection was successfully initiated, send infos and request
// connected clients list
Channel.SetRemoteInfo ( ChannelInfo );
// We have to send a connected clients list request since it can happen
// that we just had connected to the server and then disconnected but
// the server still thinks that we are connected (the server is still
// waiting for the channel time-out). If we now connect again, we would
// not get the list because the server does not know about a new connection.
// Same problem is with the jitter buffer message.
Channel.CreateReqConnClientsList();
CreateServerJitterBufferMessage();
// send opt-in / out for Channel Level updates
Channel.CreateReqChannelLevelListMes ( bDisplayChannelLevels );
}
void CClient::CreateServerJitterBufferMessage()
{
// per definition in the client: if auto jitter buffer is enabled, both,
// the client and server shall use an auto jitter buffer
if ( GetDoAutoSockBufSize() )
{
// in case auto jitter buffer size is enabled, we have to transmit a
// special value
Channel.CreateJitBufMes ( AUTO_NET_BUF_SIZE_FOR_PROTOCOL );
}
else
{
Channel.CreateJitBufMes ( GetServerSockBufNumFrames() );
}
}
void CClient::OnCLPingReceived ( CHostAddress InetAddr,
int iMs )
{
// make sure we are running and the server address is correct
if ( IsRunning() && ( InetAddr == Channel.GetAddress() ) )
{
// take care of wrap arounds (if wrapping, do not use result)
const int iCurDiff = EvaluatePingMessage ( iMs );
if ( iCurDiff >= 0 )
{
emit PingTimeReceived ( iCurDiff );
}
}
}
void CClient::OnCLPingWithNumClientsReceived ( CHostAddress InetAddr,
int iMs,
int iNumClients )
{
// take care of wrap arounds (if wrapping, do not use result)
const int iCurDiff = EvaluatePingMessage ( iMs );
if ( iCurDiff >= 0 )
{
emit CLPingTimeWithNumClientsReceived ( InetAddr,
iCurDiff,
iNumClients );
}
}
int CClient::PreparePingMessage()
{
// transmit the current precise time (in ms)
return PreciseTime.elapsed();
}
int CClient::EvaluatePingMessage ( const int iMs )
{
// calculate difference between received time in ms and current time in ms
return PreciseTime.elapsed() - iMs;
}
void CClient::SetCentralServerAddressType ( const ECSAddType eNCSAT )
{
if ( eCentralServerAddressType != eNCSAT )
{
// update type and emit message to update the server list, too
eCentralServerAddressType = eNCSAT;
emit CentralServerAddressTypeChanged();
}
}
void CClient::SetDoAutoSockBufSize ( const bool bValue )
{
// first, set new value in the channel object
Channel.SetDoAutoSockBufSize ( bValue );
// inform the server about the change
CreateServerJitterBufferMessage();
}
void CClient::SetRemoteChanGain ( const int iId,
const double dGain,
const bool bIsMyOwnFader )
{
// if this gain is for my own channel, apply the value for the Mute Myself function
if ( bIsMyOwnFader )
{
dMuteOutStreamGain = dGain;
}
Channel.SetRemoteChanGain ( iId, dGain );
}
bool CClient::SetServerAddr ( QString strNAddr )
{
CHostAddress HostAddress;
if ( NetworkUtil().ParseNetworkAddress ( strNAddr,
HostAddress ) )
{
// apply address to the channel
Channel.SetAddress ( HostAddress );
return true;
}
else
{
return false; // invalid address
}
}
bool CClient::GetAndResetbJitterBufferOKFlag()
{
// get the socket buffer put status flag and reset it
const bool bSocketJitBufOKFlag = Socket.GetAndResetbJitterBufferOKFlag();
if ( !bJitterBufferOK )
{
// our jitter buffer get status is not OK so the overall status of the
// jitter buffer is also not OK (we do not have to consider the status
// of the socket buffer put status flag)
// reset flag before returning the function
bJitterBufferOK = true;
return false;
}
// the jitter buffer get (our own status flag) is OK, the final status
// now depends on the jitter buffer put status flag from the socket
// since per definition the jitter buffer status is OK if both the
// put and get status are OK
return bSocketJitBufOKFlag;
}
void CClient::SetDisplayChannelLevels ( const bool bNDCL )
{
bDisplayChannelLevels = bNDCL;
// tell any connected server about the change
Channel.CreateReqChannelLevelListMes ( bDisplayChannelLevels );
}
void CClient::SetSndCrdPrefFrameSizeFactor ( const int iNewFactor )
{
// first check new input parameter
if ( ( iNewFactor == FRAME_SIZE_FACTOR_PREFERRED ) ||
( iNewFactor == FRAME_SIZE_FACTOR_DEFAULT ) ||
( iNewFactor == FRAME_SIZE_FACTOR_SAFE ) )
{
// init with new parameter, if client was running then first
// stop it and restart again after new initialization
const bool bWasRunning = Sound.IsRunning();
if ( bWasRunning )
{
Sound.Stop();
}
// set new parameter
iSndCrdPrefFrameSizeFactor = iNewFactor;
// init with new block size index parameter
Init();
if ( bWasRunning )
{
// restart client
Sound.Start();
}
}
}
void CClient::SetEnableOPUS64 ( const bool eNEnableOPUS64 )
{
// init with new parameter, if client was running then first
// stop it and restart again after new initialization
const bool bWasRunning = Sound.IsRunning();
if ( bWasRunning )
{
Sound.Stop();
}
// set new parameter
bEnableOPUS64 = eNEnableOPUS64;
Init();
if ( bWasRunning )
{
Sound.Start();
}
}
void CClient::SetAudioQuality ( const EAudioQuality eNAudioQuality )
{
// init with new parameter, if client was running then first
// stop it and restart again after new initialization
const bool bWasRunning = Sound.IsRunning();
if ( bWasRunning )
{
Sound.Stop();
}
// set new parameter
eAudioQuality = eNAudioQuality;
Init();
if ( bWasRunning )
{
Sound.Start();
}
}
void CClient::SetAudioChannels ( const EAudChanConf eNAudChanConf )
{
// init with new parameter, if client was running then first
// stop it and restart again after new initialization
const bool bWasRunning = Sound.IsRunning();
if ( bWasRunning )
{
Sound.Stop();
}
// set new parameter
eAudioChannelConf = eNAudChanConf;
Init();
if ( bWasRunning )
{
Sound.Start();
}
}
QString CClient::SetSndCrdDev ( const int iNewDev )
{
// if client was running then first
// stop it and restart again after new initialization
const bool bWasRunning = Sound.IsRunning();
if ( bWasRunning )
{
Sound.Stop();
}
const QString strReturn = Sound.SetDev ( iNewDev );
// init again because the sound card actual buffer size might
// be changed on new device
Init();
if ( bWasRunning )
{
// restart client
Sound.Start();
}
return strReturn;
}
void CClient::SetSndCrdLeftInputChannel ( const int iNewChan )
{
// if client was running then first
// stop it and restart again after new initialization
const bool bWasRunning = Sound.IsRunning();
if ( bWasRunning )
{
Sound.Stop();
}
Sound.SetLeftInputChannel ( iNewChan );
Init();
if ( bWasRunning )
{
// restart client
Sound.Start();
}
}
void CClient::SetSndCrdRightInputChannel ( const int iNewChan )
{
// if client was running then first
// stop it and restart again after new initialization
const bool bWasRunning = Sound.IsRunning();
if ( bWasRunning )
{
Sound.Stop();
}
Sound.SetRightInputChannel ( iNewChan );
Init();
if ( bWasRunning )
{
// restart client
Sound.Start();
}
}
void CClient::SetSndCrdLeftOutputChannel ( const int iNewChan )
{
// if client was running then first
// stop it and restart again after new initialization
const bool bWasRunning = Sound.IsRunning();
if ( bWasRunning )
{
Sound.Stop();
}
Sound.SetLeftOutputChannel ( iNewChan );
Init();
if ( bWasRunning )
{
// restart client
Sound.Start();
}
}
void CClient::SetSndCrdRightOutputChannel ( const int iNewChan )
{
// if client was running then first
// stop it and restart again after new initialization
const bool bWasRunning = Sound.IsRunning();
if ( bWasRunning )
{
Sound.Stop();
}
Sound.SetRightOutputChannel ( iNewChan );
Init();
if ( bWasRunning )
{
// restart client
Sound.Start();
}
}
void CClient::OnSndCrdReinitRequest ( int iSndCrdResetType )
{
// in older QT versions, enums cannot easily be used in signals without
// registering them -> workaroud: we use the int type and cast to the enum
const ESndCrdResetType eSndCrdResetType =
static_cast<ESndCrdResetType> ( iSndCrdResetType );
// if client was running then first
// stop it and restart again after new initialization
const bool bWasRunning = Sound.IsRunning();
if ( bWasRunning )
{
Sound.Stop();
}
// perform reinit request as indicated by the request type parameter
if ( eSndCrdResetType != RS_ONLY_RESTART )
{
if ( eSndCrdResetType != RS_ONLY_RESTART_AND_INIT )
{
// reinit the driver if requested
// (we use the currently selected driver)
Sound.SetDev ( Sound.GetDev() );
}
// init client object (must always be performed if the driver
// was changed)
Init();
}
if ( bWasRunning )
{
// restart client
Sound.Start();
}
}
void CClient::OnHandledSignal ( int sigNum )
{
#ifdef _WIN32
// Windows does not actually get OnHandledSignal triggered
QCoreApplication::instance()->exit();
Q_UNUSED ( sigNum )
#else
switch ( sigNum )
{
case SIGINT:
case SIGTERM:
// if connected, terminate connection (needed for headless mode)
if ( IsRunning() )
{
Stop();
}
// this should trigger OnAboutToQuit
QCoreApplication::instance()->exit();
break;
default:
break;
}
#endif
}
void CClient::Start()
{
// init object
Init();
// enable channel
Channel.SetEnable ( true );
// start audio interface
Sound.Start();
}
void CClient::Stop()
{
// stop audio interface
Sound.Stop();
// disable channel
Channel.SetEnable ( false );
// wait for approx. 100 ms to make sure no audio packet is still in the
// network queue causing the channel to be reconnected right after having
// received the disconnect message (seems not to gain much, disconnect is
// still not working reliably)
QTime DieTime = QTime::currentTime().addMSecs ( 100 );
while ( QTime::currentTime() < DieTime )
{
// exclude user input events because if we use AllEvents, it happens
// that if the user initiates a connection and disconnection quickly
// (e.g. quickly pressing enter five times), the software can get into
// an unknown state
QCoreApplication::processEvents (
QEventLoop::ExcludeUserInputEvents, 100 );
}
// Send disconnect message to server (Since we disable our protocol
// receive mechanism with the next command, we do not evaluate any
// respond from the server, therefore we just hope that the message
// gets its way to the server, if not, the old behaviour time-out
// disconnects the connection anyway).
ConnLessProtocol.CreateCLDisconnection ( Channel.GetAddress() );
// reset current signal level and LEDs
bJitterBufferOK = true;
SignalLevelMeter.Reset();
}
void CClient::Init()
{
// check if possible frame size factors are supported
const int iFraSizePreffered = SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_PREFERRED;
const int iFraSizeDefault = SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_DEFAULT;
const int iFraSizeSafe = SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_SAFE;
bFraSiFactPrefSupported = ( Sound.Init ( iFraSizePreffered ) == iFraSizePreffered );
bFraSiFactDefSupported = ( Sound.Init ( iFraSizeDefault ) == iFraSizeDefault );
bFraSiFactSafeSupported = ( Sound.Init ( iFraSizeSafe ) == iFraSizeSafe );
// translate block size index in actual block size
const int iPrefMonoFrameSize = iSndCrdPrefFrameSizeFactor * SYSTEM_FRAME_SIZE_SAMPLES;
// get actual sound card buffer size using preferred size
iMonoBlockSizeSam = Sound.Init ( iPrefMonoFrameSize );
// Calculate the current sound card frame size factor. In case
// the current mono block size is not a multiple of the system
// frame size, we have to use a sound card conversion buffer.
if ( ( ( iMonoBlockSizeSam == ( SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_PREFERRED ) ) && bEnableOPUS64 ) ||
( iMonoBlockSizeSam == ( SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_DEFAULT ) ) ||
( iMonoBlockSizeSam == ( SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_SAFE ) ) )
{
// regular case: one of our predefined buffer sizes is available
iSndCrdFrameSizeFactor = iMonoBlockSizeSam / SYSTEM_FRAME_SIZE_SAMPLES;
// no sound card conversion buffer required
bSndCrdConversionBufferRequired = false;
}
else
{
// An unsupported sound card buffer size is currently used -> we have
// to use a conversion buffer. Per definition we use the smallest buffer
// size as the current frame size.
// store actual sound card buffer size (stereo)
bSndCrdConversionBufferRequired = true;
iSndCardMonoBlockSizeSamConvBuff = iMonoBlockSizeSam;
// overwrite block size factor by using one frame
iSndCrdFrameSizeFactor = 1;
}
// select the OPUS frame size mode depending on current mono block size samples
if ( bSndCrdConversionBufferRequired )
{
if ( ( iSndCardMonoBlockSizeSamConvBuff < DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES ) && bEnableOPUS64 )
{
iMonoBlockSizeSam = SYSTEM_FRAME_SIZE_SAMPLES;
eAudioCompressionType = CT_OPUS64;
}
else
{
iMonoBlockSizeSam = DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES;
eAudioCompressionType = CT_OPUS;
}
}
else
{
if ( iMonoBlockSizeSam < DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES )
{
eAudioCompressionType = CT_OPUS64;
}
else
{
// since we use double size frame size for OPUS, we have to adjust the frame size factor
iSndCrdFrameSizeFactor /= 2;
eAudioCompressionType = CT_OPUS;
}
}
// inits for audio coding
if ( eAudioCompressionType == CT_OPUS )
{
iOPUSFrameSizeSamples = DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES;
if ( eAudioChannelConf == CC_MONO )
{
CurOpusEncoder = OpusEncoderMono;
CurOpusDecoder = OpusDecoderMono;
iNumAudioChannels = 1;
switch ( eAudioQuality )
{
case AQ_LOW: iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_LOW_QUALITY_DBLE_FRAMESIZE; break;
case AQ_NORMAL: iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_NORMAL_QUALITY_DBLE_FRAMESIZE; break;
case AQ_HIGH: iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_HIGH_QUALITY_DBLE_FRAMESIZE; break;
}
}
else
{
CurOpusEncoder = OpusEncoderStereo;
CurOpusDecoder = OpusDecoderStereo;
iNumAudioChannels = 2;
switch ( eAudioQuality )
{
case AQ_LOW: iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_LOW_QUALITY_DBLE_FRAMESIZE; break;
case AQ_NORMAL: iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_NORMAL_QUALITY_DBLE_FRAMESIZE; break;
case AQ_HIGH: iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_HIGH_QUALITY_DBLE_FRAMESIZE; break;
}
}
}
else /* CT_OPUS64 */
{
iOPUSFrameSizeSamples = SYSTEM_FRAME_SIZE_SAMPLES;
if ( eAudioChannelConf == CC_MONO )
{
CurOpusEncoder = Opus64EncoderMono;
CurOpusDecoder = Opus64DecoderMono;
iNumAudioChannels = 1;
switch ( eAudioQuality )
{
case AQ_LOW: iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_LOW_QUALITY; break;
case AQ_NORMAL: iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_NORMAL_QUALITY; break;
case AQ_HIGH: iCeltNumCodedBytes = OPUS_NUM_BYTES_MONO_HIGH_QUALITY; break;
}
}
else
{
CurOpusEncoder = Opus64EncoderStereo;
CurOpusDecoder = Opus64DecoderStereo;
iNumAudioChannels = 2;
switch ( eAudioQuality )
{
case AQ_LOW: iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_LOW_QUALITY; break;
case AQ_NORMAL: iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_NORMAL_QUALITY; break;
case AQ_HIGH: iCeltNumCodedBytes = OPUS_NUM_BYTES_STEREO_HIGH_QUALITY; break;
}
}
}
// calculate stereo (two channels) buffer size
iStereoBlockSizeSam = 2 * iMonoBlockSizeSam;
vecCeltData.Init ( iCeltNumCodedBytes );
vecZeros.Init ( iStereoBlockSizeSam, 0 );
vecsStereoSndCrdMuteStream.Init ( iStereoBlockSizeSam );
dMuteOutStreamGain = 1.0;
opus_custom_encoder_ctl ( CurOpusEncoder,
OPUS_SET_BITRATE (
CalcBitRateBitsPerSecFromCodedBytes (
iCeltNumCodedBytes, iOPUSFrameSizeSamples ) ) );
// inits for network and channel
vecbyNetwData.Init ( iCeltNumCodedBytes );
// set the channel network properties
Channel.SetAudioStreamProperties ( eAudioCompressionType,
iCeltNumCodedBytes,
iSndCrdFrameSizeFactor,
iNumAudioChannels );
// init reverberation
AudioReverb.Init ( eAudioChannelConf,
iStereoBlockSizeSam,
SYSTEM_SAMPLE_RATE_HZ );
// init the sound card conversion buffers
if ( bSndCrdConversionBufferRequired )
{
// inits for conversion buffer (the size of the conversion buffer must
// be the sum of input/output sizes which is the worst case fill level)
const int iSndCardStereoBlockSizeSamConvBuff = 2 * iSndCardMonoBlockSizeSamConvBuff;
const int iConBufSize = iStereoBlockSizeSam + iSndCardStereoBlockSizeSamConvBuff;
SndCrdConversionBufferIn.Init ( iConBufSize );
SndCrdConversionBufferOut.Init ( iConBufSize );
vecDataConvBuf.Init ( iStereoBlockSizeSam );
// the output conversion buffer must be filled with the inner
// block size for initialization (this is the latency which is
// introduced by the conversion buffer) to avoid buffer underruns
SndCrdConversionBufferOut.Put ( vecZeros, iStereoBlockSizeSam );
}
// reset initialization phase flag and mute flag
bIsInitializationPhase = true;
}
void CClient::AudioCallback ( CVector<int16_t>& psData, void* arg )
{
// get the pointer to the object
CClient* pMyClientObj = static_cast<CClient*> ( arg );
// process audio data
pMyClientObj->ProcessSndCrdAudioData ( psData );
/*
// TEST do a soundcard jitter measurement
static CTimingMeas JitterMeas ( 1000, "test2.dat" );
JitterMeas.Measure();
*/
}
void CClient::ProcessSndCrdAudioData ( CVector<int16_t>& vecsStereoSndCrd )
{
// check if a conversion buffer is required or not
if ( bSndCrdConversionBufferRequired )
{
// add new sound card block in conversion buffer
SndCrdConversionBufferIn.Put ( vecsStereoSndCrd, vecsStereoSndCrd.Size() );
// process all available blocks of data
while ( SndCrdConversionBufferIn.GetAvailData() >= iStereoBlockSizeSam )
{
// get one block of data for processing
SndCrdConversionBufferIn.Get ( vecDataConvBuf, iStereoBlockSizeSam );
// process audio data
ProcessAudioDataIntern ( vecDataConvBuf );
SndCrdConversionBufferOut.Put ( vecDataConvBuf, iStereoBlockSizeSam );
}
// get processed sound card block out of the conversion buffer
SndCrdConversionBufferOut.Get ( vecsStereoSndCrd, vecsStereoSndCrd.Size() );
}
else
{
// regular case: no conversion buffer required
// process audio data
ProcessAudioDataIntern ( vecsStereoSndCrd );
}
}
void CClient::ProcessAudioDataIntern ( CVector<int16_t>& vecsStereoSndCrd )
{
int i, j, iUnused;
unsigned char* pCurCodedData;
// Transmit signal ---------------------------------------------------------
// update stereo signal level meter
SignalLevelMeter.Update ( vecsStereoSndCrd );
// add reverberation effect if activated
if ( iReverbLevel != 0 )
{
AudioReverb.Process ( vecsStereoSndCrd,
bReverbOnLeftChan,
static_cast<double> ( iReverbLevel ) / AUD_REVERB_MAX / 4 );
}
// apply pan (audio fader) and mix mono signals
if ( !( ( iAudioInFader == AUD_FADER_IN_MIDDLE ) && ( eAudioChannelConf == CC_STEREO ) ) )
{
// calculate pan gain in the range 0 to 1, where 0.5 is the middle position
const double dPan = static_cast<double> ( iAudioInFader ) / AUD_FADER_IN_MAX;
if ( eAudioChannelConf == CC_STEREO )
{
// for stereo only apply pan attenuation on one channel (same as pan in the server)
const double dGainL = std::min ( 0.5, 1 - dPan ) * 2;
const double dGainR = std::min ( 0.5, dPan ) * 2;
for ( i = 0, j = 0; i < iMonoBlockSizeSam; i++, j += 2 )
{
// note that the gain is always <= 1, therefore a simple cast is
// ok since we never can get an overload
vecsStereoSndCrd[j + 1] = static_cast<int16_t> ( dGainR * vecsStereoSndCrd[j + 1] );
vecsStereoSndCrd[j] = static_cast<int16_t> ( dGainL * vecsStereoSndCrd[j] );
}
}
else
{
// for mono implement a cross-fade between channels and mix them
const double dGainL = 1 - dPan;
const double dGainR = dPan;
for ( i = 0, j = 0; i < iMonoBlockSizeSam; i++, j += 2 )
{
vecsStereoSndCrd[i] = Double2Short (
dGainL * vecsStereoSndCrd[j] + dGainR * vecsStereoSndCrd[j + 1] );
}
}
}
// Support for mono-in/stereo-out mode: Per definition this mode works in
// full stereo mode at the transmission level. The only thing which is done
// is to mix both sound card inputs together and then put this signal on
// both stereo channels to be transmitted to the server.
if ( eAudioChannelConf == CC_MONO_IN_STEREO_OUT )
{
// copy mono data in stereo sound card buffer (note that since the input
// and output is the same buffer, we have to start from the end not to
// overwrite input values)
for ( i = iMonoBlockSizeSam - 1, j = iStereoBlockSizeSam - 2; i >= 0; i--, j -= 2 )
{
vecsStereoSndCrd[j] = vecsStereoSndCrd[j + 1] = vecsStereoSndCrd[i];
}
}
for ( i = 0; i < iSndCrdFrameSizeFactor; i++ )
{
// OPUS encoding
if ( CurOpusEncoder != nullptr )
{
if ( bMuteOutStream )
{
iUnused = opus_custom_encode ( CurOpusEncoder,
&vecZeros[i * iNumAudioChannels * iOPUSFrameSizeSamples],
iOPUSFrameSizeSamples,
&vecCeltData[0],
iCeltNumCodedBytes );
}
else
{
iUnused = opus_custom_encode ( CurOpusEncoder,
&vecsStereoSndCrd[i * iNumAudioChannels * iOPUSFrameSizeSamples],
iOPUSFrameSizeSamples,
&vecCeltData[0],
iCeltNumCodedBytes );
}
}
// send coded audio through the network
Channel.PrepAndSendPacket ( &Socket,
vecCeltData,
iCeltNumCodedBytes );
}
// Receive signal ----------------------------------------------------------
// in case of mute stream, store local data
if ( bMuteOutStream )
{
vecsStereoSndCrdMuteStream = vecsStereoSndCrd;
}
for ( i = 0; i < iSndCrdFrameSizeFactor; i++ )
{
// receive a new block
const bool bReceiveDataOk =
( Channel.GetData ( vecbyNetwData, iCeltNumCodedBytes ) == GS_BUFFER_OK );
// get pointer to coded data and manage the flags
if ( bReceiveDataOk )
{
pCurCodedData = &vecbyNetwData[0];
// on any valid received packet, we clear the initialization phase flag
bIsInitializationPhase = false;
}
else
{
// for lost packets use null pointer as coded input data
pCurCodedData = nullptr;
// invalidate the buffer OK status flag
bJitterBufferOK = false;
}
// OPUS decoding
if ( CurOpusDecoder != nullptr )
{
iUnused = opus_custom_decode ( CurOpusDecoder,
pCurCodedData,
iCeltNumCodedBytes,
&vecsStereoSndCrd[i * iNumAudioChannels * iOPUSFrameSizeSamples],
iOPUSFrameSizeSamples );
}
}
// for muted stream we have to add our local data here
if ( bMuteOutStream )
{
for ( i = 0; i < iStereoBlockSizeSam; i++ )
{
vecsStereoSndCrd[i] = Double2Short (
vecsStereoSndCrd[i] + vecsStereoSndCrdMuteStream[i] * dMuteOutStreamGain );
}
}
// check if channel is connected and if we do not have the initialization phase
if ( Channel.IsConnected() && ( !bIsInitializationPhase ) )
{
if ( eAudioChannelConf == CC_MONO )
{
// copy mono data in stereo sound card buffer (note that since the input
// and output is the same buffer, we have to start from the end not to
// overwrite input values)
for ( i = iMonoBlockSizeSam - 1, j = iStereoBlockSizeSam - 2; i >= 0; i--, j -= 2 )
{
vecsStereoSndCrd[j] = vecsStereoSndCrd[j + 1] = vecsStereoSndCrd[i];
}
}
}
else
{
// if not connected, clear data
vecsStereoSndCrd.Reset ( 0 );
}
// update socket buffer size
Channel.UpdateSocketBufferSize();
Q_UNUSED ( iUnused )
}
int CClient::EstimatedOverallDelay ( const int iPingTimeMs )
{
const double dSystemBlockDurationMs = static_cast<double> ( iOPUSFrameSizeSamples ) /
SYSTEM_SAMPLE_RATE_HZ * 1000;
// If the jitter buffers are set effectively, i.e. they are exactly the
// size of the network jitter, then the delay of the buffer is the buffer
// length. Since that is usually not the case but the buffers are usually
// a bit larger than necessary, we introduce some factor for compensation.
// Consider the jitter buffer on the client and on the server side, too.
const double dTotalJitterBufferDelayMs = dSystemBlockDurationMs *
static_cast<double> ( GetSockBufNumFrames() +
GetServerSockBufNumFrames() ) * 0.7;
// consider delay introduced by the sound card conversion buffer by using
// "GetSndCrdConvBufAdditionalDelayMonoBlSize()"
double dTotalSoundCardDelayMs = GetSndCrdConvBufAdditionalDelayMonoBlSize() *
1000 / SYSTEM_SAMPLE_RATE_HZ;
// try to get the actual input/output sound card delay from the audio
// interface, per definition it is not available if a 0 is returned
const double dSoundCardInputOutputLatencyMs = Sound.GetInOutLatencyMs();
if ( dSoundCardInputOutputLatencyMs == 0.0 )
{
// use an alternative aproach for estimating the sound card delay:
//
// we assume that we have two period sizes for the input and one for the
// output, therefore we have "3 *" instead of "2 *" (for input and output)
// the actual sound card buffer size
// "GetSndCrdConvBufAdditionalDelayMonoBlSize"
dTotalSoundCardDelayMs +=
( 3 * GetSndCrdActualMonoBlSize() ) *
1000 / SYSTEM_SAMPLE_RATE_HZ;
}
else
{
// add the actual sound card latency in ms
dTotalSoundCardDelayMs += dSoundCardInputOutputLatencyMs;
}
// network packets are of the same size as the audio packets per definition
// if no sound card conversion buffer is used
const double dDelayToFillNetworkPacketsMs =
GetSystemMonoBlSize() * 1000 / SYSTEM_SAMPLE_RATE_HZ;
// OPUS additional delay at small frame sizes is half a frame size
const double dAdditionalAudioCodecDelayMs = dSystemBlockDurationMs / 2;
const double dTotalBufferDelayMs =
dDelayToFillNetworkPacketsMs +
dTotalJitterBufferDelayMs +
dTotalSoundCardDelayMs +
dAdditionalAudioCodecDelayMs;
return MathUtils::round ( dTotalBufferDelayMs + iPingTimeMs );
}