/******************************************************************************\ * Copyright (c) 2004-2013 * * 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., * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * \******************************************************************************/ #include "client.h" /* Implementation *************************************************************/ CClient::CClient ( const quint16 iPortNumber ) : vstrIPAddress ( MAX_NUM_SERVER_ADDR_ITEMS, "" ), ChannelInfo (), vecStoredFaderTags ( MAX_NUM_STORED_FADER_LEVELS, "" ), vecStoredFaderLevels ( MAX_NUM_STORED_FADER_LEVELS, AUD_MIX_FADER_MAX ), Channel ( false ), /* we need a client channel -> "false" */ eAudioCompressionType ( CT_OPUS ), iCeltNumCodedBytes ( CELT_NUM_BYTES_MONO_NORMAL_QUALITY ), bCeltDoHighQuality ( false ), bUseStereo ( false ), Socket ( &Channel, &ConnLessProtocol, iPortNumber ), Sound ( AudioCallback, this ), iAudioInFader ( AUD_FADER_IN_MIDDLE ), bReverbOnLeftChan ( false ), iReverbLevel ( 0 ), iSndCrdPrefFrameSizeFactor ( FRAME_SIZE_FACTOR_PREFERRED ), iSndCrdFrameSizeFactor ( FRAME_SIZE_FACTOR_PREFERRED ), bSndCrdConversionBufferRequired ( false ), iSndCardMonoBlockSizeSamConvBuff ( 0 ), bFraSiFactPrefSupported ( false ), bFraSiFactDefSupported ( false ), bFraSiFactSafeSupported ( false ), bOpenChatOnNewMessage ( true ), eGUIDesign ( GD_ORIGINAL ), strCentralServerAddress ( "" ), bUseDefaultCentralServerAddress ( true ), iServerSockBufNumFrames ( DEF_NET_BUF_SIZE_NUM_BL ) { int iOpusError; // init audio encoder/decoder (mono) CeltModeMono = cc6_celt_mode_create ( SYSTEM_SAMPLE_RATE_HZ, 1, SYSTEM_FRAME_SIZE_SAMPLES, NULL ); CeltEncoderMono = cc6_celt_encoder_create ( CeltModeMono ); CeltDecoderMono = cc6_celt_decoder_create ( CeltModeMono ); #ifdef USE_LOW_COMPLEXITY_CELT_ENC // set encoder low complexity cc6_celt_encoder_ctl ( CeltEncoderMono, cc6_CELT_SET_COMPLEXITY ( 1 ) ); #endif OpusMode = opus_custom_mode_create ( SYSTEM_SAMPLE_RATE_HZ, SYSTEM_FRAME_SIZE_SAMPLES, &iOpusError ); OpusEncoderMono = opus_custom_encoder_create ( OpusMode, 1, &iOpusError ); OpusDecoderMono = opus_custom_decoder_create ( OpusMode, 1, &iOpusError ); // we require a constant bit rate opus_custom_encoder_ctl ( OpusEncoderMono, OPUS_SET_VBR ( 0 ) ); // we expect packet loss, tell the encoder about it opus_custom_encoder_ctl ( OpusEncoderMono, OPUS_SET_PACKET_LOSS_PERC ( OPUS_EXPECTED_PACKET_LOSS_PERC ) ); // we want as low delay as possible opus_custom_encoder_ctl ( OpusEncoderMono, OPUS_SET_APPLICATION ( OPUS_APPLICATION_RESTRICTED_LOWDELAY ) ); #ifdef USE_LOW_COMPLEXITY_CELT_ENC // set encoder low complexity opus_custom_encoder_ctl ( OpusEncoderMono, OPUS_SET_COMPLEXITY ( 1 ) ); #endif // init audio encoder/decoder (stereo) CeltModeStereo = cc6_celt_mode_create ( SYSTEM_SAMPLE_RATE_HZ, 2, SYSTEM_FRAME_SIZE_SAMPLES, NULL ); CeltEncoderStereo = cc6_celt_encoder_create ( CeltModeStereo ); CeltDecoderStereo = cc6_celt_decoder_create ( CeltModeStereo ); #ifdef USE_LOW_COMPLEXITY_CELT_ENC // set encoder low complexity cc6_celt_encoder_ctl ( CeltEncoderStereo, cc6_CELT_SET_COMPLEXITY ( 1 ) ); #endif OpusEncoderStereo = opus_custom_encoder_create ( OpusMode, 2, &iOpusError ); OpusDecoderStereo = opus_custom_decoder_create ( OpusMode, 2, &iOpusError ); // we require a constant bit rate opus_custom_encoder_ctl ( OpusEncoderStereo, OPUS_SET_VBR ( 0 ) ); // we expect packet loss, tell the encoder about it opus_custom_encoder_ctl ( OpusEncoderStereo, OPUS_SET_PACKET_LOSS_PERC ( OPUS_EXPECTED_PACKET_LOSS_PERC ) ); // we want as low delay as possible opus_custom_encoder_ctl ( OpusEncoderStereo, OPUS_SET_APPLICATION ( OPUS_APPLICATION_RESTRICTED_LOWDELAY ) ); #ifdef USE_LOW_COMPLEXITY_CELT_ENC // set encoder low complexity opus_custom_encoder_ctl ( OpusEncoderStereo, OPUS_SET_COMPLEXITY ( 1 ) ); #endif // Connections ------------------------------------------------------------- // connection for protocol QObject::connect ( &Channel, SIGNAL ( MessReadyForSending ( CVector ) ), this, SLOT ( OnSendProtMessage ( CVector ) ) ); QObject::connect ( &Channel, SIGNAL ( DetectedCLMessage ( CVector, int ) ), this, SLOT ( OnDetectedCLMessage ( CVector, int ) ) ); QObject::connect ( &Channel, SIGNAL ( ReqJittBufSize() ), this, SLOT ( OnReqJittBufSize() ) ); QObject::connect ( &Channel, SIGNAL ( JittBufSizeChanged ( int ) ), this, SLOT ( OnJittBufSizeChanged ( int ) ) ); QObject::connect ( &Channel, SIGNAL ( ReqChanInfo() ), this, SLOT ( OnReqChanInfo() ) ); QObject::connect ( &Channel, SIGNAL ( ConClientListNameMesReceived ( CVector ) ), SIGNAL ( ConClientListNameMesReceived ( CVector ) ) ); QObject::connect ( &Channel, SIGNAL ( ConClientListMesReceived ( CVector ) ), SIGNAL ( ConClientListMesReceived ( CVector ) ) ); QObject::connect ( &Channel, SIGNAL ( Disconnected() ), SIGNAL ( Disconnected() ) ); QObject::connect ( &Channel, SIGNAL ( NewConnection() ), this, SLOT ( OnNewConnection() ) ); QObject::connect ( &Channel, SIGNAL ( ChatTextReceived ( QString ) ), SIGNAL ( ChatTextReceived ( QString ) ) ); // #### COMPATIBILITY OLD VERSION, TO BE REMOVED #### QObject::connect ( &Channel, SIGNAL ( OpusSupported() ), this, SLOT ( OnOpusSupported() ) ); QObject::connect ( &ConnLessProtocol, SIGNAL ( CLMessReadyForSending ( CHostAddress, CVector ) ), this, SLOT ( OnSendCLProtMessage ( CHostAddress, CVector ) ) ); QObject::connect ( &ConnLessProtocol, SIGNAL ( CLServerListReceived ( CHostAddress, CVector ) ), SIGNAL ( CLServerListReceived ( CHostAddress, CVector ) ) ); QObject::connect ( &ConnLessProtocol, SIGNAL ( CLPingReceived ( CHostAddress, int ) ), this, SLOT ( OnCLPingReceived ( CHostAddress, int ) ) ); QObject::connect ( &ConnLessProtocol, SIGNAL ( CLPingWithNumClientsReceived ( CHostAddress, int, int ) ), this, SLOT ( OnCLPingWithNumClientsReceived ( CHostAddress, int, int ) ) ); QObject::connect ( &Sound, SIGNAL ( ReinitRequest ( int ) ), this, SLOT ( OnSndCrdReinitRequest ( int ) ) ); } 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::OnDetectedCLMessage ( CVector vecbyData, int iNumBytes ) { // this is a special case: we received a connection less message but we are // in a connection ConnLessProtocol.ParseConnectionLessMessage ( vecbyData, iNumBytes, Channel.GetAddress() ); } 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(); } 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::SetDoAutoSockBufSize ( const bool bValue ) { // first, set new value in the channel object Channel.SetDoAutoSockBufSize ( bValue ); // inform the server about the change CreateServerJitterBufferMessage(); } bool CClient::SetServerAddr ( QString strNAddr ) { CHostAddress HostAddress; if ( LlconNetwUtil().ParseNetworkAddress ( strNAddr, HostAddress ) ) { // apply address to the channel Channel.SetAddress ( HostAddress ); return true; } else { return false; // invalid address } } 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(); } } } // #### COMPATIBILITY OLD VERSION, TO BE REMOVED #### void CClient::OnOpusSupported() { if ( eAudioCompressionType != CT_OPUS ) { SetAudoCompressiontype ( CT_OPUS ); } } // #### COMPATIBILITY OLD VERSION, TO BE REMOVED #### void CClient::SetAudoCompressiontype ( const EAudComprType eNAudCompressionType ) { // 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 eAudioCompressionType = eNAudCompressionType; Init(); if ( bWasRunning ) { Sound.Start(); } } void CClient::SetCELTHighQuality ( const bool bNCeltHighQualityFlag ) { // 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 bCeltDoHighQuality = bNCeltHighQualityFlag; Init(); if ( bWasRunning ) { Sound.Start(); } } void CClient::SetUseStereo ( const bool bNUseStereo ) { // 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 bUseStereo = bNUseStereo; 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::Start() { // #### COMPATIBILITY OLD VERSION, TO BE REMOVED #### // our first attempt is always to use the old code eAudioCompressionType = CT_CELT; // 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 SignalLevelMeter.Reset(); PostWinMessage ( MS_RESET_ALL, 0 ); } void CClient::Init() { // check if possible frame size factors are supported const int iFraSizePreffered = FRAME_SIZE_FACTOR_PREFERRED * SYSTEM_FRAME_SIZE_SAMPLES; bFraSiFactPrefSupported = ( Sound.Init ( iFraSizePreffered ) == iFraSizePreffered ); const int iFraSizeDefault = FRAME_SIZE_FACTOR_DEFAULT * SYSTEM_FRAME_SIZE_SAMPLES; bFraSiFactDefSupported = ( Sound.Init ( iFraSizeDefault ) == iFraSizeDefault ); const int iFraSizeSafe = FRAME_SIZE_FACTOR_SAFE * SYSTEM_FRAME_SIZE_SAMPLES; 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 ) ) || ( 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 llcon frame size // store actual sound card buffer size (stereo) iSndCardMonoBlockSizeSamConvBuff = iMonoBlockSizeSam; const int iSndCardStereoBlockSizeSamConvBuff = 2 * iMonoBlockSizeSam; // overwrite block size by smallest supported llcon buffer size iSndCrdFrameSizeFactor = FRAME_SIZE_FACTOR_PREFERRED; iMonoBlockSizeSam = SYSTEM_FRAME_SIZE_SAMPLES * FRAME_SIZE_FACTOR_PREFERRED; iStereoBlockSizeSam = 2 * iMonoBlockSizeSam; // 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 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 const CVector vZeros ( iStereoBlockSizeSam, 0 ); SndCrdConversionBufferOut.Put ( vZeros, vZeros.Size() ); bSndCrdConversionBufferRequired = true; } // calculate stereo (two channels) buffer size iStereoBlockSizeSam = 2 * iMonoBlockSizeSam; vecsAudioSndCrdMono.Init ( iMonoBlockSizeSam ); vecdAudioStereo.Init ( iStereoBlockSizeSam ); // init reverberation AudioReverbL.Init ( SYSTEM_SAMPLE_RATE_HZ ); AudioReverbR.Init ( SYSTEM_SAMPLE_RATE_HZ ); // inits for CELT coding if ( bCeltDoHighQuality ) { if ( bUseStereo ) { iCeltNumCodedBytes = CELT_NUM_BYTES_STEREO_HIGH_QUALITY; } else { iCeltNumCodedBytes = CELT_NUM_BYTES_MONO_HIGH_QUALITY; } } else { if ( bUseStereo ) { iCeltNumCodedBytes = CELT_NUM_BYTES_STEREO_NORMAL_QUALITY; } else { iCeltNumCodedBytes = CELT_NUM_BYTES_MONO_NORMAL_QUALITY; } } vecCeltData.Init ( iCeltNumCodedBytes ); if ( bUseStereo ) { opus_custom_encoder_ctl ( OpusEncoderStereo, OPUS_SET_BITRATE ( CalcBitRateBitsPerSecFromCodedBytes ( iCeltNumCodedBytes ) ) ); } else { opus_custom_encoder_ctl ( OpusEncoderMono, OPUS_SET_BITRATE ( CalcBitRateBitsPerSecFromCodedBytes ( iCeltNumCodedBytes ) ) ); } // inits for network and channel vecbyNetwData.Init ( iCeltNumCodedBytes ); if ( bUseStereo ) { vecsNetwork.Init ( iStereoBlockSizeSam ); // set the channel network properties Channel.SetAudioStreamProperties ( eAudioCompressionType, iCeltNumCodedBytes, iSndCrdFrameSizeFactor, 2 ); } else { vecsNetwork.Init ( iMonoBlockSizeSam ); // set the channel network properties Channel.SetAudioStreamProperties ( eAudioCompressionType, iCeltNumCodedBytes, iSndCrdFrameSizeFactor, 1 ); } } void CClient::AudioCallback ( CVector& psData, void* arg ) { // get the pointer to the object CClient* pMyClientObj = reinterpret_cast ( arg ); // process audio data pMyClientObj->ProcessSndCrdAudioData ( psData ); } 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 ); // process audio data ProcessAudioDataIntern ( vecDataConvBuf ); SndCrdConversionBufferOut.Put ( vecDataConvBuf, vecDataConvBuf.Size() ); } // get processed sound card block out of the conversion buffer SndCrdConversionBufferOut.Get ( vecsStereoSndCrd ); } else { // regular case: no conversion buffer required // process audio data ProcessAudioDataIntern ( vecsStereoSndCrd ); } } void CClient::ProcessAudioDataIntern ( CVector& vecsStereoSndCrd ) { int i, j; // Transmit signal --------------------------------------------------------- // update stereo signal level meter SignalLevelMeter.Update ( vecsStereoSndCrd ); // convert data from short to double for ( i = 0; i < iStereoBlockSizeSam; i++ ) { vecdAudioStereo[i] = static_cast ( vecsStereoSndCrd[i] ); } // add reverberation effect if activated if ( iReverbLevel != 0 ) { // calculate attenuation amplification factor const double dRevLev = static_cast ( iReverbLevel ) / AUD_REVERB_MAX / 2; if ( bUseStereo ) { // for stereo always apply reverberation effect on both channels for ( i = 0; i < iStereoBlockSizeSam; i += 2 ) { // left channel vecdAudioStereo[i] += dRevLev * AudioReverbL.ProcessSample ( vecdAudioStereo[i] ); // right channel vecdAudioStereo[i + 1] += dRevLev * AudioReverbR.ProcessSample ( vecdAudioStereo[i + 1] ); } } else { if ( bReverbOnLeftChan ) { for ( i = 0; i < iStereoBlockSizeSam; i += 2 ) { // left channel vecdAudioStereo[i] += dRevLev * AudioReverbL.ProcessSample ( vecdAudioStereo[i] ); } } else { for ( i = 1; i < iStereoBlockSizeSam; i += 2 ) { // right channel vecdAudioStereo[i] += dRevLev * AudioReverbR.ProcessSample ( vecdAudioStereo[i] ); } } } } // mix both signals depending on the fading setting, convert // from double to short if ( iAudioInFader == AUD_FADER_IN_MIDDLE ) { if ( bUseStereo ) { // perform type conversion for ( i = 0; i < iStereoBlockSizeSam; i++ ) { vecsNetwork[i] = Double2Short ( vecdAudioStereo[i] ); } } else { // mix channels together for ( i = 0, j = 0; i < iMonoBlockSizeSam; i++, j += 2 ) { vecsNetwork[i] = Double2Short ( ( vecdAudioStereo[j] + vecdAudioStereo[j + 1] ) / 2 ); } } } else { if ( bUseStereo ) { // stereo const double dAttFactStereo = static_cast ( AUD_FADER_IN_MIDDLE - abs ( AUD_FADER_IN_MIDDLE - iAudioInFader ) ) / AUD_FADER_IN_MIDDLE; if ( iAudioInFader > AUD_FADER_IN_MIDDLE ) { for ( i = 0, j = 0; i < iMonoBlockSizeSam; i++, j += 2 ) { // attenuation on right channel vecsNetwork[j] = Double2Short ( vecdAudioStereo[j] ); vecsNetwork[j + 1] = Double2Short ( dAttFactStereo * vecdAudioStereo[j + 1] ); } } else { for ( i = 0, j = 0; i < iMonoBlockSizeSam; i++, j += 2 ) { // attenuation on left channel vecsNetwork[j] = Double2Short ( dAttFactStereo * vecdAudioStereo[j] ); vecsNetwork[j + 1] = Double2Short ( vecdAudioStereo[j + 1] ); } } } else { // mono // make sure that in the middle position the two channels are // amplified by 1/2, if the pan is set to one channel, this // channel should have an amplification of 1 const double dAttFactMono = static_cast ( AUD_FADER_IN_MIDDLE - abs ( AUD_FADER_IN_MIDDLE - iAudioInFader ) ) / AUD_FADER_IN_MIDDLE / 2; const double dAmplFactMono = 0.5 + static_cast ( abs ( AUD_FADER_IN_MIDDLE - iAudioInFader ) ) / AUD_FADER_IN_MIDDLE / 2; if ( iAudioInFader > AUD_FADER_IN_MIDDLE ) { for ( i = 0, j = 0; i < iMonoBlockSizeSam; i++, j += 2 ) { // attenuation on right channel vecsNetwork[i] = Double2Short ( dAmplFactMono * vecdAudioStereo[j] + dAttFactMono * vecdAudioStereo[j + 1] ); } } else { for ( i = 0, j = 0; i < iMonoBlockSizeSam; i++, j += 2 ) { // attenuation on left channel vecsNetwork[i] = Double2Short ( dAmplFactMono * vecdAudioStereo[j + 1] + dAttFactMono * vecdAudioStereo[j] ); } } } } for ( i = 0; i < iSndCrdFrameSizeFactor; i++ ) { if ( bUseStereo ) { // encode current audio frame if ( eAudioCompressionType == CT_CELT ) { cc6_celt_encode ( CeltEncoderStereo, &vecsNetwork[i * 2 * SYSTEM_FRAME_SIZE_SAMPLES], NULL, &vecCeltData[0], iCeltNumCodedBytes ); } else { opus_custom_encode ( OpusEncoderStereo, &vecsNetwork[i * 2 * SYSTEM_FRAME_SIZE_SAMPLES], SYSTEM_FRAME_SIZE_SAMPLES, &vecCeltData[0], iCeltNumCodedBytes ); } } else { // encode current audio frame if ( eAudioCompressionType == CT_CELT ) { cc6_celt_encode ( CeltEncoderMono, &vecsNetwork[i * SYSTEM_FRAME_SIZE_SAMPLES], NULL, &vecCeltData[0], iCeltNumCodedBytes ); } else { opus_custom_encode ( OpusEncoderMono, &vecsNetwork[i * SYSTEM_FRAME_SIZE_SAMPLES], SYSTEM_FRAME_SIZE_SAMPLES, &vecCeltData[0], iCeltNumCodedBytes ); } } // send coded audio through the network Socket.SendPacket ( Channel.PrepSendPacket ( vecCeltData ), Channel.GetAddress() ); } // Receive signal ---------------------------------------------------------- for ( i = 0; i < iSndCrdFrameSizeFactor; i++ ) { // receive a new block const bool bReceiveDataOk = ( Channel.GetData ( vecbyNetwData ) == GS_BUFFER_OK ); if ( bReceiveDataOk ) { PostWinMessage ( MS_JIT_BUF_GET, MUL_COL_LED_GREEN ); } else { PostWinMessage ( MS_JIT_BUF_GET, MUL_COL_LED_RED ); } // CELT decoding if ( bReceiveDataOk ) { if ( bUseStereo ) { if ( eAudioCompressionType == CT_CELT ) { cc6_celt_decode ( CeltDecoderStereo, &vecbyNetwData[0], iCeltNumCodedBytes, &vecsStereoSndCrd[i * 2 * SYSTEM_FRAME_SIZE_SAMPLES] ); } else { opus_custom_decode ( OpusDecoderStereo, &vecbyNetwData[0], iCeltNumCodedBytes, &vecsStereoSndCrd[i * 2 * SYSTEM_FRAME_SIZE_SAMPLES], SYSTEM_FRAME_SIZE_SAMPLES ); } } else { if ( eAudioCompressionType == CT_CELT ) { cc6_celt_decode ( CeltDecoderMono, &vecbyNetwData[0], iCeltNumCodedBytes, &vecsAudioSndCrdMono[i * SYSTEM_FRAME_SIZE_SAMPLES] ); } else { opus_custom_decode ( OpusDecoderMono, &vecbyNetwData[0], iCeltNumCodedBytes, &vecsAudioSndCrdMono[i * SYSTEM_FRAME_SIZE_SAMPLES], SYSTEM_FRAME_SIZE_SAMPLES ); } } } else { // lost packet if ( bUseStereo ) { if ( eAudioCompressionType == CT_CELT ) { cc6_celt_decode ( CeltDecoderStereo, NULL, 0, &vecsStereoSndCrd[i * 2 * SYSTEM_FRAME_SIZE_SAMPLES] ); } else { opus_custom_decode ( OpusDecoderStereo, NULL, iCeltNumCodedBytes, &vecsStereoSndCrd[i * 2 * SYSTEM_FRAME_SIZE_SAMPLES], SYSTEM_FRAME_SIZE_SAMPLES ); } } else { if ( eAudioCompressionType == CT_CELT ) { cc6_celt_decode ( CeltDecoderMono, NULL, 0, &vecsAudioSndCrdMono[i * SYSTEM_FRAME_SIZE_SAMPLES] ); } else { opus_custom_decode ( OpusDecoderMono, NULL, iCeltNumCodedBytes, &vecsAudioSndCrdMono[i * SYSTEM_FRAME_SIZE_SAMPLES], SYSTEM_FRAME_SIZE_SAMPLES ); } } } } /* // TEST // fid=fopen('v.dat','r');x=fread(fid,'int16');fclose(fid); static FILE* pFileDelay = fopen("c:\\temp\\test2.dat", "wb"); short sData[2]; for (i = 0; i < iMonoBlockSizeSam; i++) { sData[0] = (short) vecsAudioSndCrdMono[i]; fwrite(&sData, size_t(2), size_t(1), pFileDelay); } fflush(pFileDelay); */ // check if channel is connected if ( Channel.IsConnected() ) { if ( !bUseStereo ) { // copy mono data in stereo sound card buffer for ( i = 0, j = 0; i < iMonoBlockSizeSam; i++, j += 2 ) { vecsStereoSndCrd[j] = vecsStereoSndCrd[j + 1] = vecsAudioSndCrdMono[i]; } } } else { // if not connected, clear data vecsStereoSndCrd.Reset ( 0 ); } // update socket buffer size Channel.UpdateSocketBufferSize(); } int CClient::EstimatedOverallDelay ( const int iPingTimeMs ) { /* For estimating the overall delay, use the following assumptions: - the mean delay of a cyclic buffer is half the buffer size (since for the average it is assumed that the buffer is half filled) - consider the jitter buffer on the server side, too */ // the buffer sizes at client and server divided by 2 (half the buffer // for the delay) is the total socket buffer size const double dTotalJitterBufferDelayMs = SYSTEM_BLOCK_DURATION_MS_FLOAT * static_cast ( GetSockBufNumFrames() + GetServerSockBufNumFrames() ) / 2; // 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, also consider delay introduced by // sound card conversion buffer by using // "GetSndCrdConvBufAdditionalDelayMonoBlSize" const double dTotalSoundCardDelayMs = ( 3 * GetSndCrdActualMonoBlSize() + GetSndCrdConvBufAdditionalDelayMonoBlSize() ) * 1000 / SYSTEM_SAMPLE_RATE_HZ; // 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; // CELT additional delay at small frame sizes is half a frame size const double dAdditionalAudioCodecDelayMs = SYSTEM_BLOCK_DURATION_MS_FLOAT / 2; const double dTotalBufferDelayMs = dDelayToFillNetworkPacketsMs + dTotalJitterBufferDelayMs + dTotalSoundCardDelayMs + dAdditionalAudioCodecDelayMs; return LlconMath::round ( dTotalBufferDelayMs + iPingTimeMs ); }