eeva/jamulus
1
0
Fork 0
Code Issues Pull requests Releases Wiki Activity

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

Browse source
This commit is contained in:
Volker Fischer 2011-05-22 20:40:29 +00:00
parent a770b75a06
commit b7ce6c3193
6 changed files with 159 additions and 99 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

96
src/channel.cpp
View file

@ -30,6 +30,7 @@ CChannel::CChannel ( const bool bNIsServer ) :
vecdGains ( MAX_NUM_CHANNELS, (double) 1.0 ),
bIsEnabled ( false ),
bIsServer ( bNIsServer ),
bDoAutoSockBufSize ( true ),
iNetwFrameSizeFact ( FRAME_SIZE_FACTOR_PREFERRED ),
iNetwFrameSize ( 20 ), // must be > 0 and should be close to a valid size
iNumAudioChannels ( 1 ) // mono
@ -171,6 +172,9 @@ bool CChannel::SetSockBufNumFrames ( const int iNewNumFrames,
// first check for valid input parameter range
if ( ( iNewNumFrames >= MIN_NET_BUF_SIZE_NUM_BL ) &&
( iNewNumFrames <= MAX_NET_BUF_SIZE_NUM_BL ) )
{
// only apply parameter if new parameter is different from current one
if ( iCurSockBufNumFrames != iNewNumFrames )
{
// store new value
iCurSockBufNumFrames = iNewNumFrames;
@ -179,8 +183,22 @@ bool CChannel::SetSockBufNumFrames ( const int iNewNumFrames,
// block size
SockBuf.Init ( iNetwFrameSize, iNewNumFrames, bPreserve );
// TEST in case we are the client, tell the server that the size has changed
if ( !bIsServer && bPreserve )
{
// TODO this function call causes the lock up of the protocol mechanism when the
// SetSockBufNumFrames is called directly from UpdateSocketBufferSize
// -> find and fix the problem!
CreateJitBufMes ( iNewNumFrames );
}
return false; // -> no error
}
}
return true; // set error flag
}
@ -549,3 +567,81 @@ int CChannel::GetUploadRateKbps()
8 /* bits per byte */ *
SYSTEM_SAMPLE_RATE_HZ / iAudioSizeOut / 1000;
}
// TEST
void CChannel::UpdateSocketBufferSize ( const double dAudioBufferDurationMs,
const double dLocalStdDev )
{
// 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;
// jitter introduced in the server by the timer implementation
// TODO remove this!
const double dServerJitterMs = 0.666666; // ms
// 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 + dServerJitterMs +
3.3 * ( GetTimingStdDev() + dLocalStdDev ) ) /
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
// TEST
PostWinMessage ( MS_SET_JIT_BUF_SIZE, iUpperHystDec );
//SetSockBufNumFrames ( iUpperHystDec, true );
}
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.
// TEST
PostWinMessage ( MS_SET_JIT_BUF_SIZE, iUpperHystDec );
//SetSockBufNumFrames ( iUpperHystDec, true );
}
}
}
}

12
src/channel.h
View file

@ -96,6 +96,12 @@ public:
const bool bPreserve = false );
int GetSockBufNumFrames() const { return iCurSockBufNumFrames; }
// TEST
void UpdateSocketBufferSize ( const double dAudioBufferDurationMs,
const double dLocalStdDev );
int GetUploadRateKbps();
double GetTimingStdDev() { return CycleTimeVariance.GetStdDev(); }
@ -104,8 +110,13 @@ public:
void SetAudioStreamProperties ( const int iNewNetwFrameSize,
const int iNewNetwFrameSizeFact,
const int iNewNumAudioChannels );
void SetDoAutoSockBufSize ( const bool bValue ) { bDoAutoSockBufSize = bValue; }
bool GetDoAutoSockBufSize() const { return bDoAutoSockBufSize; }
int GetNetwFrameSizeFact() const { return iNetwFrameSizeFact; }
int GetNetwFrameSize() const { return iNetwFrameSize; }
int GetNumAudioChannels() const { return iNumAudioChannels; }
// network protocol interface
@ -160,6 +171,7 @@ protected:
bool bIsEnabled;
bool bIsServer;
bool bDoAutoSockBufSize;
int iNetwFrameSizeFact;
int iNetwFrameSize;

73
src/client.cpp
View file

@ -30,7 +30,6 @@ CClient::CClient ( const quint16 iPortNumber ) :
vstrIPAddress ( MAX_NUM_SERVER_ADDR_ITEMS, "" ),
strName ( "" ),
Channel ( false ), /* we need a client channel -> "false" */
bDoAutoSockBufSize ( true ),
iCeltNumCodedBytes ( CELT_NUM_BYTES_MONO_NORMAL_QUALITY ),
bCeltDoHighQuality ( false ),
bUseStereo ( false ),
@ -428,7 +427,12 @@ void CClient::Stop()
QTime DieTime = QTime::currentTime().addMSecs ( 100 );
while ( QTime::currentTime() < DieTime )
{
QCoreApplication::processEvents ( QEventLoop::AllEvents, 100 );
// 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
@ -877,28 +881,8 @@ void CClient::ProcessAudioDataIntern ( CVector<int16_t>& vecsStereoSndCrd )
vecsStereoSndCrd.Reset ( 0 );
}
// update response time measurement and socket buffer size
// update response time measurement
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 =
@ -906,46 +890,9 @@ void CClient::UpdateSocketBufferSize()
GetSndCrdConvBufAdditionalDelayMonoBlSize() ) *
1000 / SYSTEM_SAMPLE_RATE_HZ;
// jitter introduced in the server by the timer implementation
const double dServerJitterMs = 0.666666; // ms
// 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 + dServerJitterMs +
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 );
}
}
}
// update and socket buffer size
Channel.UpdateSocketBufferSize ( dAudioBufferDurationMs,
CycleTimeVariance.GetStdDev() );
}
int CClient::EstimatedOverallDelay ( const int iPingTimeMs )

21
src/client.h
View file

@ -132,21 +132,24 @@ public:
AudioReverbR.Clear();
}
void SetDoAutoSockBufSize ( const bool bValue ) { bDoAutoSockBufSize = bValue; }
bool GetDoAutoSockBufSize() const { return bDoAutoSockBufSize; }
void SetDoAutoSockBufSize ( const bool bValue )
{ Channel.SetDoAutoSockBufSize ( bValue ); }
bool GetDoAutoSockBufSize() const { return Channel.GetDoAutoSockBufSize(); }
void SetSockBufNumFrames ( const int iNumBlocks,
const bool bPreserve = false )
{
// only change parameter if new parameter is different from current one
if ( Channel.GetSockBufNumFrames() != iNumBlocks )
{
// set the new socket size (number of frames)
if ( !Channel.SetSockBufNumFrames ( iNumBlocks, bPreserve ) )
{
// if setting of socket buffer size was successful,
// tell the server that size has changed
Channel.CreateJitBufMes ( iNumBlocks );
}
// tell the server that the size has changed
// TEST is done in channel now
// Channel.CreateJitBufMes ( iNumBlocks );
}
}
int GetSockBufNumFrames() { return Channel.GetSockBufNumFrames(); }
@ -250,7 +253,6 @@ protected:
void Init();
void ProcessSndCrdAudioData ( CVector<short>& vecsStereoSndCrd );
void ProcessAudioDataIntern ( CVector<short>& vecsStereoSndCrd );
void UpdateSocketBufferSize();
int PreparePingMessage();
int EvaluatePingMessage ( const int iMs );
@ -258,7 +260,6 @@ protected:
// only one channel is needed for client application
CChannel Channel;
CProtocol ConnLessProtocol;
bool bDoAutoSockBufSize;
// audio encoder/decoder
CELTMode* CeltModeMono;

3
src/connectdlg.cpp
View file

@ -79,6 +79,9 @@ CConnectDlg::CConnectDlg ( QWidget* parent, Qt::WindowFlags f )
lvwServers->setColumnWidth ( 3, 130 );
lvwServers->clear();
// make sure the connect button has the focus
butConnect->setFocus();
// Connections -------------------------------------------------------------
// list view

3
src/socket.cpp
View file

@ -140,10 +140,11 @@ void CSocket::OnDataReceived()
// TEST old code -> to be removed because this is not working!!!
/*
pChannel->SetEnable ( true );
pChannel->CreateAndImmSendDisconnectionMes();
pChannel->SetEnable ( false );
*/
// TODO this does not work because for a connected channel at the server, no
// connection less protocol messages are accepted