moved the socket buffer size update from the client in the channel because it shall be also used in the server in the future, bug fix with connect/disconnect
This commit is contained in:
parent
a770b75a06
commit
b7ce6c3193
6 changed files with 159 additions and 99 deletions
118
src/channel.cpp
118
src/channel.cpp
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@ -27,12 +27,13 @@
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// CChannel implementation *****************************************************
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CChannel::CChannel ( const bool bNIsServer ) :
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vecdGains ( MAX_NUM_CHANNELS, (double) 1.0 ),
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bIsEnabled ( false ),
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bIsServer ( bNIsServer ),
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vecdGains ( MAX_NUM_CHANNELS, (double) 1.0 ),
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bIsEnabled ( false ),
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bIsServer ( bNIsServer ),
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bDoAutoSockBufSize ( true ),
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iNetwFrameSizeFact ( FRAME_SIZE_FACTOR_PREFERRED ),
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iNetwFrameSize ( 20 ), // must be > 0 and should be close to a valid size
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iNumAudioChannels ( 1 ) // mono
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iNetwFrameSize ( 20 ), // must be > 0 and should be close to a valid size
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iNumAudioChannels ( 1 ) // mono
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{
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// initial value for connection time out counter, we calculate the total
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// number of samples here and subtract the number of samples of the block
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@ -172,14 +173,31 @@ bool CChannel::SetSockBufNumFrames ( const int iNewNumFrames,
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if ( ( iNewNumFrames >= MIN_NET_BUF_SIZE_NUM_BL ) &&
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( iNewNumFrames <= MAX_NET_BUF_SIZE_NUM_BL ) )
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{
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// store new value
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iCurSockBufNumFrames = iNewNumFrames;
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// only apply parameter if new parameter is different from current one
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if ( iCurSockBufNumFrames != iNewNumFrames )
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{
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// store new value
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iCurSockBufNumFrames = iNewNumFrames;
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// the network block size is a multiple of the minimum network
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// block size
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SockBuf.Init ( iNetwFrameSize, iNewNumFrames, bPreserve );
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// the network block size is a multiple of the minimum network
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// block size
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SockBuf.Init ( iNetwFrameSize, iNewNumFrames, bPreserve );
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return false; // -> no error
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// TEST in case we are the client, tell the server that the size has changed
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if ( !bIsServer && bPreserve )
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{
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// TODO this function call causes the lock up of the protocol mechanism when the
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// SetSockBufNumFrames is called directly from UpdateSocketBufferSize
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// -> find and fix the problem!
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CreateJitBufMes ( iNewNumFrames );
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}
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return false; // -> no error
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}
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}
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return true; // set error flag
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@ -549,3 +567,81 @@ int CChannel::GetUploadRateKbps()
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8 /* bits per byte */ *
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SYSTEM_SAMPLE_RATE_HZ / iAudioSizeOut / 1000;
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}
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// TEST
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void CChannel::UpdateSocketBufferSize ( const double dAudioBufferDurationMs,
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const double dLocalStdDev )
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{
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// just update the socket buffer size if auto setting is enabled, otherwise
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// do nothing
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if ( bDoAutoSockBufSize )
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{
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// We use the time response measurement for the automatic setting.
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// Assumptions:
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// - the audio interface/network jitter is assumed to be Gaussian
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// - the buffer size is set to 3.3 times the standard deviation of
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// the jitter (~98% of the jitter should be fit in the
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// buffer)
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// - introduce a hysteresis to avoid switching the buffer sizes all the
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// time in case the time response measurement is close to a bound
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// - only use time response measurement results if averaging buffer is
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// completely filled
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const double dHysteresis = 0.3;
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// jitter introduced in the server by the timer implementation
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// TODO remove this!
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const double dServerJitterMs = 0.666666; // ms
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// accumulate the standard deviations of input network stream and
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// internal timer,
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// add 0.5 to "round up" -> ceil,
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// divide by MIN_SERVER_BLOCK_DURATION_MS because this is the size of
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// one block in the jitter buffer
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const double dEstCurBufSet = ( dAudioBufferDurationMs + dServerJitterMs +
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3.3 * ( GetTimingStdDev() + dLocalStdDev ) ) /
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SYSTEM_BLOCK_DURATION_MS_FLOAT + 0.5;
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// upper/lower hysteresis decision
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const int iUpperHystDec = LlconMath().round ( dEstCurBufSet - dHysteresis );
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const int iLowerHystDec = LlconMath().round ( dEstCurBufSet + dHysteresis );
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// if both decisions are equal than use the result
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if ( iUpperHystDec == iLowerHystDec )
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{
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// set the socket buffer via the main window thread since somehow
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// it gives a protocol deadlock if we call the SetSocketBufSize()
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// function directly
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// TEST
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PostWinMessage ( MS_SET_JIT_BUF_SIZE, iUpperHystDec );
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//SetSockBufNumFrames ( iUpperHystDec, true );
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}
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else
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{
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// we are in the middle of the decision region, use
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// previous setting for determing the new decision
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if ( !( ( GetSockBufNumFrames() == iUpperHystDec ) ||
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( GetSockBufNumFrames() == iLowerHystDec ) ) )
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{
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// The old result is not near the new decision,
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// use per definition the upper decision.
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// Set the socket buffer via the main window thread since somehow
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// it gives a protocol deadlock if we call the SetSocketBufSize()
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// function directly.
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// TEST
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PostWinMessage ( MS_SET_JIT_BUF_SIZE, iUpperHystDec );
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//SetSockBufNumFrames ( iUpperHystDec, true );
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}
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}
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}
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}
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@ -96,6 +96,12 @@ public:
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const bool bPreserve = false );
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int GetSockBufNumFrames() const { return iCurSockBufNumFrames; }
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// TEST
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void UpdateSocketBufferSize ( const double dAudioBufferDurationMs,
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const double dLocalStdDev );
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int GetUploadRateKbps();
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double GetTimingStdDev() { return CycleTimeVariance.GetStdDev(); }
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@ -104,8 +110,13 @@ public:
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void SetAudioStreamProperties ( const int iNewNetwFrameSize,
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const int iNewNetwFrameSizeFact,
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const int iNewNumAudioChannels );
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void SetDoAutoSockBufSize ( const bool bValue ) { bDoAutoSockBufSize = bValue; }
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bool GetDoAutoSockBufSize() const { return bDoAutoSockBufSize; }
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int GetNetwFrameSizeFact() const { return iNetwFrameSizeFact; }
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int GetNetwFrameSize() const { return iNetwFrameSize; }
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int GetNumAudioChannels() const { return iNumAudioChannels; }
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// network protocol interface
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@ -160,6 +171,7 @@ protected:
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bool bIsEnabled;
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bool bIsServer;
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bool bDoAutoSockBufSize;
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int iNetwFrameSizeFact;
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int iNetwFrameSize;
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@ -30,7 +30,6 @@ CClient::CClient ( const quint16 iPortNumber ) :
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vstrIPAddress ( MAX_NUM_SERVER_ADDR_ITEMS, "" ),
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strName ( "" ),
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Channel ( false ), /* we need a client channel -> "false" */
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bDoAutoSockBufSize ( true ),
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iCeltNumCodedBytes ( CELT_NUM_BYTES_MONO_NORMAL_QUALITY ),
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bCeltDoHighQuality ( false ),
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bUseStereo ( false ),
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@ -428,7 +427,12 @@ void CClient::Stop()
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QTime DieTime = QTime::currentTime().addMSecs ( 100 );
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while ( QTime::currentTime() < DieTime )
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{
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QCoreApplication::processEvents ( QEventLoop::AllEvents, 100 );
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// exclude user input events because if we use AllEvents, it happens
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// that if the user initiates a connection and disconnection quickly
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// (e.g. quickly pressing enter five times), the software can get into
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// an unknown state
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QCoreApplication::processEvents (
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QEventLoop::ExcludeUserInputEvents, 100 );
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}
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// Send disconnect message to server (Since we disable our protocol
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@ -877,75 +881,18 @@ void CClient::ProcessAudioDataIntern ( CVector<int16_t>& vecsStereoSndCrd )
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vecsStereoSndCrd.Reset ( 0 );
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}
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// update response time measurement and socket buffer size
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// update response time measurement
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CycleTimeVariance.Update();
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UpdateSocketBufferSize();
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}
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void CClient::UpdateSocketBufferSize()
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{
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// just update the socket buffer size if auto setting is enabled, otherwise
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// do nothing
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if ( bDoAutoSockBufSize )
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{
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// We use the time response measurement for the automatic setting.
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// Assumptions:
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// - the audio interface/network jitter is assumed to be Gaussian
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// - the buffer size is set to 3.3 times the standard deviation of
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// the jitter (~98% of the jitter should be fit in the
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// buffer)
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// - introduce a hysteresis to avoid switching the buffer sizes all the
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// time in case the time response measurement is close to a bound
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// - only use time response measurement results if averaging buffer is
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// completely filled
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const double dHysteresis = 0.3;
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// calculate current buffer setting
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const double dAudioBufferDurationMs =
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( GetSndCrdActualMonoBlSize() +
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GetSndCrdConvBufAdditionalDelayMonoBlSize() ) *
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1000 / SYSTEM_SAMPLE_RATE_HZ;
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// calculate current buffer setting
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const double dAudioBufferDurationMs =
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( GetSndCrdActualMonoBlSize() +
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GetSndCrdConvBufAdditionalDelayMonoBlSize() ) *
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1000 / SYSTEM_SAMPLE_RATE_HZ;
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// jitter introduced in the server by the timer implementation
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const double dServerJitterMs = 0.666666; // ms
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// accumulate the standard deviations of input network stream and
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// internal timer,
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// add 0.5 to "round up" -> ceil,
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// divide by MIN_SERVER_BLOCK_DURATION_MS because this is the size of
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// one block in the jitter buffer
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const double dEstCurBufSet = ( dAudioBufferDurationMs + dServerJitterMs +
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3.3 * ( Channel.GetTimingStdDev() + CycleTimeVariance.GetStdDev() ) ) /
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SYSTEM_BLOCK_DURATION_MS_FLOAT + 0.5;
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// upper/lower hysteresis decision
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const int iUpperHystDec = LlconMath().round ( dEstCurBufSet - dHysteresis );
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const int iLowerHystDec = LlconMath().round ( dEstCurBufSet + dHysteresis );
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// if both decisions are equal than use the result
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if ( iUpperHystDec == iLowerHystDec )
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{
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// set the socket buffer via the main window thread since somehow
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// it gives a protocol deadlock if we call the SetSocketBufSize()
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// function directly
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PostWinMessage ( MS_SET_JIT_BUF_SIZE, iUpperHystDec );
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}
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else
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{
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// we are in the middle of the decision region, use
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// previous setting for determing the new decision
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if ( !( ( GetSockBufNumFrames() == iUpperHystDec ) ||
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( GetSockBufNumFrames() == iLowerHystDec ) ) )
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{
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// The old result is not near the new decision,
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// use per definition the upper decision.
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// Set the socket buffer via the main window thread since somehow
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// it gives a protocol deadlock if we call the SetSocketBufSize()
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// function directly.
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PostWinMessage ( MS_SET_JIT_BUF_SIZE, iUpperHystDec );
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}
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}
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}
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// update and socket buffer size
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Channel.UpdateSocketBufferSize ( dAudioBufferDurationMs,
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CycleTimeVariance.GetStdDev() );
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}
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int CClient::EstimatedOverallDelay ( const int iPingTimeMs )
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39
src/client.h
39
src/client.h
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AudioReverbR.Clear();
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}
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void SetDoAutoSockBufSize ( const bool bValue ) { bDoAutoSockBufSize = bValue; }
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bool GetDoAutoSockBufSize() const { return bDoAutoSockBufSize; }
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void SetDoAutoSockBufSize ( const bool bValue )
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{ Channel.SetDoAutoSockBufSize ( bValue ); }
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bool GetDoAutoSockBufSize() const { return Channel.GetDoAutoSockBufSize(); }
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void SetSockBufNumFrames ( const int iNumBlocks,
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const bool bPreserve = false )
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{
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// only change parameter if new parameter is different from current one
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if ( Channel.GetSockBufNumFrames() != iNumBlocks )
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// set the new socket size (number of frames)
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if ( !Channel.SetSockBufNumFrames ( iNumBlocks, bPreserve ) )
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{
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// set the new socket size (number of frames)
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if ( !Channel.SetSockBufNumFrames ( iNumBlocks, bPreserve ) )
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{
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// if setting of socket buffer size was successful,
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// tell the server that size has changed
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Channel.CreateJitBufMes ( iNumBlocks );
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}
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// if setting of socket buffer size was successful,
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// tell the server that the size has changed
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// TEST is done in channel now
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// Channel.CreateJitBufMes ( iNumBlocks );
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}
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}
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int GetSockBufNumFrames() { return Channel.GetSockBufNumFrames(); }
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protected:
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// callback function must be static, otherwise it does not work
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static void AudioCallback ( CVector<short>& psData, void* arg );
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static void AudioCallback ( CVector<short>& psData, void* arg );
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void Init();
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void ProcessSndCrdAudioData ( CVector<short>& vecsStereoSndCrd );
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void ProcessAudioDataIntern ( CVector<short>& vecsStereoSndCrd );
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void UpdateSocketBufferSize();
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void Init();
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void ProcessSndCrdAudioData ( CVector<short>& vecsStereoSndCrd );
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void ProcessAudioDataIntern ( CVector<short>& vecsStereoSndCrd );
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int PreparePingMessage();
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int EvaluatePingMessage ( const int iMs );
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int PreparePingMessage();
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int EvaluatePingMessage ( const int iMs );
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// only one channel is needed for client application
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CChannel Channel;
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CProtocol ConnLessProtocol;
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bool bDoAutoSockBufSize;
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// audio encoder/decoder
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CELTMode* CeltModeMono;
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@ -79,6 +79,9 @@ CConnectDlg::CConnectDlg ( QWidget* parent, Qt::WindowFlags f )
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lvwServers->setColumnWidth ( 3, 130 );
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lvwServers->clear();
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// make sure the connect button has the focus
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butConnect->setFocus();
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// Connections -------------------------------------------------------------
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// list view
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@ -140,10 +140,11 @@ void CSocket::OnDataReceived()
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// TEST old code -> to be removed because this is not working!!!
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/*
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pChannel->SetEnable ( true );
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pChannel->CreateAndImmSendDisconnectionMes();
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pChannel->SetEnable ( false );
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*/
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// TODO this does not work because for a connected channel at the server, no
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// connection less protocol messages are accepted
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