/******************************************************************************\ * Copyright (c) 2004-2009 * * 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 ) : Channel ( false ), /* we need a client channel -> "false" */ Sound ( AudioCallback, this ), Socket ( &Channel, iPortNumber ), iAudioInFader ( AUD_FADER_IN_MIDDLE ), iReverbLevel ( 0 ), bReverbOnLeftChan ( false ), vstrIPAddress ( MAX_NUM_SERVER_ADDR_ITEMS, "" ), strName ( "" ), bOpenChatOnNewMessage ( true ), bDoAutoSockBufSize ( true ), iSndCrdPrefFrameSizeFactor ( FRAME_SIZE_FACTOR_DEFAULT ), iSndCrdFrameSizeFactor ( FRAME_SIZE_FACTOR_DEFAULT ) { // init audio endocder/decoder (mono) CeltMode = celt_mode_create ( SYSTEM_SAMPLE_RATE, 1, SYSTEM_FRAME_SIZE_SAMPLES, NULL ); CeltEncoder = celt_encoder_create ( CeltMode ); CeltDecoder = celt_decoder_create ( CeltMode ); // connections ------------------------------------------------------------- // connection for protocol QObject::connect ( &Channel, SIGNAL ( MessReadyForSending ( CVector ) ), this, SLOT ( OnSendProtMessage ( CVector ) ) ); QObject::connect ( &Channel, SIGNAL ( ReqJittBufSize() ), this, SLOT ( OnReqJittBufSize() ) ); QObject::connect ( &Channel, SIGNAL ( ConClientListMesReceived ( CVector ) ), SIGNAL ( ConClientListMesReceived ( CVector ) ) ); QObject::connect ( &Channel, SIGNAL ( NewConnection() ), this, SLOT ( OnNewConnection() ) ); QObject::connect ( &Channel, SIGNAL ( ChatTextReceived ( QString ) ), this, SIGNAL ( ChatTextReceived ( QString ) ) ); QObject::connect ( &Channel, SIGNAL ( PingReceived ( int ) ), this, SLOT ( OnReceivePingMessage ( int ) ) ); QObject::connect ( &Sound, SIGNAL ( ReinitRequest() ), this, SLOT ( OnSndCrdReinitRequest() ) ); } 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::OnReqJittBufSize() { // TODO cant we implement this OnReqJjittBufSize inside the channel object? Channel.CreateJitBufMes ( Channel.GetSockBufNumFrames() ); } void CClient::OnNewConnection() { // a new connection was successfully initiated, send name and request // connected clients list Channel.SetRemoteName ( strName ); // 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. Channel.CreateReqConnClientsList(); } void CClient::OnReceivePingMessage ( int iMs ) { // calculate difference between received time in ms and current time in ms, // take care of wrap arounds (if wrapping, do not use result) const int iCurDiff = PreciseTime.elapsed() - iMs; if ( iCurDiff >= 0 ) { emit PingTimeReceived ( iCurDiff ); } } bool CClient::SetServerAddr ( QString strNAddr ) { QHostAddress InetAddr; quint16 iNetPort = LLCON_DEFAULT_PORT_NUMBER; // parse input address for the type [IP address]:[port number] QString strPort = strNAddr.section ( ":", 1, 1 ); if ( !strPort.isEmpty() ) { // a colon is present in the address string, try to extract port number iNetPort = strPort.toInt(); // extract address port before colon (should be actual internet address) strNAddr = strNAddr.section ( ":", 0, 0 ); } // first try if this is an IP number an can directly applied to QHostAddress if ( !InetAddr.setAddress ( strNAddr ) ) { // it was no vaild IP address, try to get host by name, assuming // that the string contains a valid host name string QHostInfo HostInfo = QHostInfo::fromName ( strNAddr ); if ( HostInfo.error() == QHostInfo::NoError ) { // apply IP address to QT object if ( !HostInfo.addresses().isEmpty() ) { // use the first IP address InetAddr = HostInfo.addresses().first(); } } else { return false; // invalid address } } // apply address (the server port is fixed and always the same) Channel.SetAddress ( CHostAddress ( InetAddr, iNetPort ) ); return true; } void CClient::SetSndCrdPrefFrameSizeFactor ( const int iNewFactor ) { // right now we simply set the internal value 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 ) { 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 ).c_str(); // init again because the sound card actual buffer size might // be changed on new device Init(); if ( bWasRunning ) { Sound.Start(); } return strReturn; } void CClient::OnSndCrdReinitRequest() { // if client was running then first // stop it and restart again after new initialization const bool bWasRunning = Sound.IsRunning(); if ( bWasRunning ) { Sound.Stop(); } // reinit the driver (we use the currently selected driver) and // init client object, too Sound.SetDev ( Sound.GetDev() ); Init(); if ( bWasRunning ) { Sound.Start(); } } void CClient::Start() { // init object Init(); // enable channel Channel.SetEnable ( true ); // start audio interface Sound.Start(); } void CClient::Stop() { // stop audio interface Sound.Stop(); // 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) Channel.CreateDisconnectionMes(); // disable channel Channel.SetEnable ( false ); // reset current signal level and LEDs SignalLevelMeter.Reset(); PostWinMessage ( MS_RESET_ALL, 0 ); } void CClient::AudioCallback ( CVector& psData, void* arg ) { // get the pointer to the object CClient* pMyClientObj = reinterpret_cast ( arg ); // process audio data pMyClientObj->ProcessAudioData ( psData ); } void CClient::Init() { // 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 ); iStereoBlockSizeSam = 2 * iMonoBlockSizeSam; // TEST // calculate actual frame size factor iSndCrdFrameSizeFactor = iMonoBlockSizeSam / SYSTEM_FRAME_SIZE_SAMPLES; vecsAudioSndCrdMono.Init ( iMonoBlockSizeSam ); vecsAudioSndCrdStereo.Init ( iStereoBlockSizeSam ); vecdAudioStereo.Init ( iStereoBlockSizeSam ); // init response time evaluation CycleTimeVariance.Init ( iMonoBlockSizeSam, SYSTEM_SAMPLE_RATE, TIME_MOV_AV_RESPONSE ); CycleTimeVariance.Reset(); // init reverberation AudioReverb.Init ( SYSTEM_SAMPLE_RATE ); // 22: low/normal quality 150 kbsp (128) / 108 kbps (256) // 44: high quality 216 kbps (128) / 174 kbps (256) iCeltNumCodedBytes = 22; vecCeltData.Init ( iCeltNumCodedBytes ); // init network buffers vecsNetwork.Init ( iMonoBlockSizeSam ); vecbyNetwData.Init ( iCeltNumCodedBytes ); // set the channel network properties Channel.SetNetwFrameSizeAndFact ( iCeltNumCodedBytes, iSndCrdFrameSizeFactor ); } void CClient::ProcessAudioData ( 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] = (double) vecsStereoSndCrd[i]; } // add reverberation effect if activated if ( iReverbLevel != 0 ) { // calculate attenuation amplification factor const double dRevLev = (double) iReverbLevel / AUD_REVERB_MAX / 2; if ( bReverbOnLeftChan ) { for ( i = 0; i < iStereoBlockSizeSam; i += 2 ) { // left channel vecdAudioStereo[i] += dRevLev * AudioReverb.ProcessSample ( vecdAudioStereo[i] ); } } else { for ( i = 1; i < iStereoBlockSizeSam; i += 2 ) { // right channel vecdAudioStereo[i] += dRevLev * AudioReverb.ProcessSample ( vecdAudioStereo[i] ); } } } // mix both signals depending on the fading setting, convert // from double to short if ( iAudioInFader == AUD_FADER_IN_MIDDLE ) { // just mix channels together for ( i = 0, j = 0; i < iMonoBlockSizeSam; i++, j += 2 ) { vecsNetwork[i] = Double2Short ( ( vecdAudioStereo[j] + vecdAudioStereo[j + 1] ) / 2 ); } } else { const double dAttFact = (double) ( 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[i] = Double2Short ( ( vecdAudioStereo[j] + dAttFact * vecdAudioStereo[j + 1] ) / 2 ); } } else { for ( i = 0, j = 0; i < iMonoBlockSizeSam; i++, j += 2 ) { // attenuation on left channel vecsNetwork[i] = Double2Short ( ( vecdAudioStereo[j + 1] + dAttFact * vecdAudioStereo[j] ) / 2 ); } } } // TEST // TODO use actual frame size factor, not preferred one!!!! for ( i = 0; i < iSndCrdFrameSizeFactor; i++ ) { // encode current audio frame with CELT encoder celt_encode ( CeltEncoder, &vecsNetwork[i * SYSTEM_FRAME_SIZE_SAMPLES], NULL, &vecCeltData[0], iCeltNumCodedBytes ); // send coded audio through the network Socket.SendPacket ( Channel.PrepSendPacket ( vecCeltData ), Channel.GetAddress() ); } // Receive signal ---------------------------------------------------------- // TEST // TODO use actual frame size factor, not preferred one!!!! 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 ) { celt_decode ( CeltDecoder, &vecbyNetwData[0], iCeltNumCodedBytes, &vecsAudioSndCrdMono[i * SYSTEM_FRAME_SIZE_SAMPLES] ); } else { // lost packet celt_decode ( CeltDecoder, NULL, iCeltNumCodedBytes, &vecsAudioSndCrdMono[i * SYSTEM_FRAME_SIZE_SAMPLES] ); } } // check if channel is connected if ( Channel.IsConnected() ) { // 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 response time measurement and socket buffer size CycleTimeVariance.Update(); UpdateSocketBufferSize(); } void CClient::UpdateSocketBufferSize() { // just update the socket buffer size if auto setting is enabled, otherwise // do nothing if ( bDoAutoSockBufSize ) { // We use the time response measurement for the automatic setting. // Assumptions: // - the audio interface/network jitter is assumed to be Gaussian // - the buffer size is set to 3.3 times the standard deviation of // the jitter (~98% of the jitter should be fit in the // buffer) // - introduce a hysteresis to avoid switching the buffer sizes all the // time in case the time response measurement is close to a bound // - only use time response measurement results if averaging buffer is // completely filled const double dHysteresis = 0.3; // calculate current buffer setting const double dAudioBufferDurationMs = iMonoBlockSizeSam * 1000 / SYSTEM_SAMPLE_RATE; // accumulate the standard deviations of input network stream and // internal timer, // add 0.5 to "round up" -> ceil, // divide by MIN_SERVER_BLOCK_DURATION_MS because this is the size of // one block in the jitter buffer const double dEstCurBufSet = ( dAudioBufferDurationMs + 3.3 * ( Channel.GetTimingStdDev() + CycleTimeVariance.GetStdDev() ) ) / SYSTEM_BLOCK_DURATION_MS_FLOAT + 0.5; // upper/lower hysteresis decision const int iUpperHystDec = LlconMath().round ( dEstCurBufSet - dHysteresis ); const int iLowerHystDec = LlconMath().round ( dEstCurBufSet + dHysteresis ); // if both decisions are equal than use the result if ( iUpperHystDec == iLowerHystDec ) { // set the socket buffer via the main window thread since somehow // it gives a protocol deadlock if we call the SetSocketBufSize() // function directly PostWinMessage ( MS_SET_JIT_BUF_SIZE, iUpperHystDec ); } else { // we are in the middle of the decision region, use // previous setting for determing the new decision if ( !( ( GetSockBufNumFrames() == iUpperHystDec ) || ( GetSockBufNumFrames() == iLowerHystDec ) ) ) { // The old result is not near the new decision, // use per definition the upper decision. // Set the socket buffer via the main window thread since somehow // it gives a protocol deadlock if we call the SetSocketBufSize() // function directly. PostWinMessage ( MS_SET_JIT_BUF_SIZE, iUpperHystDec ); } } } }