/******************************************************************************\ * 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 ) : ChannelInfo ( ), strClientName ( strNClientName ), 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::OnControllerInFaderLevel ); 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(); } } CClient::~CClient() { // free audio encoders and decoders opus_custom_encoder_destroy ( OpusEncoderMono ); opus_custom_decoder_destroy ( OpusDecoderMono ); opus_custom_encoder_destroy ( OpusEncoderStereo ); opus_custom_decoder_destroy ( OpusDecoderStereo ); opus_custom_encoder_destroy ( Opus64EncoderMono ); opus_custom_decoder_destroy ( Opus64DecoderMono ); opus_custom_encoder_destroy ( Opus64EncoderStereo ); opus_custom_decoder_destroy ( Opus64DecoderStereo ); // free audio modes opus_custom_mode_destroy ( OpusMode ); opus_custom_mode_destroy ( Opus64Mode ); } void CClient::OnSendProtMessage ( CVector 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 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 could 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 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 ( 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::OnControllerInFaderLevel ( int iChannelIdx, int iValue ) { // in case of a headless client the faders cannot be moved so we need // to send the controller information directly to the server #ifdef HEADLESS // only apply new fader level if channel index is valid if ( ( iChannelIdx >= 0 ) && ( iChannelIdx < MAX_NUM_CHANNELS ) ) { SetRemoteChanGain ( iChannelIdx, MathUtils::CalcFaderGain ( iValue ), false ); } #endif emit ControllerInFaderLevel ( iChannelIdx, iValue ); } 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& psData, void* arg ) { // get the pointer to the object CClient* pMyClientObj = static_cast ( 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& 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& vecsStereoSndCrd ) { int i, j, iUnused; unsigned char* pCurCodedData; // Transmit signal --------------------------------------------------------- // update stereo signal level meter (not needed in headless mode) #ifndef HEADLESS SignalLevelMeter.Update ( vecsStereoSndCrd, iMonoBlockSizeSam, true ); #endif // add reverberation effect if activated if ( iReverbLevel != 0 ) { AudioReverb.Process ( vecsStereoSndCrd, bReverbOnLeftChan, static_cast ( 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 ( 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 = MathUtils::GetLeftPan ( dPan, false ); const double dGainR = MathUtils::GetRightPan ( dPan, false ); 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 ( dGainR * vecsStereoSndCrd[j + 1] ); vecsStereoSndCrd[j] = static_cast ( dGainL * vecsStereoSndCrd[j] ); } } else { // for mono implement a cross-fade between channels and mix them, for // mono-in/stereo-out use no attenuation in pan center const double dGainL = MathUtils::GetLeftPan ( dPan, eAudioChannelConf != CC_MONO_IN_STEREO_OUT ); const double dGainR = MathUtils::GetRightPan ( dPan, eAudioChannelConf != CC_MONO_IN_STEREO_OUT ); for ( i = 0, j = 0; i < iMonoBlockSizeSam; i++, j += 2 ) { // note that we need the Double2Short for stereo pan mode 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 ( 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 ( 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 approach 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 ); }