jamulus/src/channel.cpp

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/******************************************************************************\
* Copyright (c) 2004-2011
*
* 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 "channel.h"
// CChannel implementation *****************************************************
CChannel::CChannel ( const bool bNIsServer ) :
vecdGains ( MAX_NUM_CHANNELS, (double) 1.0 ),
bDoAutoSockBufSize ( true ),
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bIsEnabled ( false ),
bIsServer ( bNIsServer ),
iNetwFrameSizeFact ( FRAME_SIZE_FACTOR_PREFERRED ),
iNetwFrameSize ( 20 ), // must be > 0 and should be close to a valid size
iNumAudioChannels ( 1 ) // mono
{
// initial value for connection time out counter, we calculate the total
// number of samples here and subtract the number of samples of the block
// which we take out of the buffer to be independent of block sizes
iConTimeOutStartVal = CON_TIME_OUT_SEC_MAX * SYSTEM_SAMPLE_RATE_HZ;
// init time-out for the buffer with zero -> no connection
iConTimeOut = 0;
// init the socket buffer
SetSockBufNumFrames ( DEF_NET_BUF_SIZE_NUM_BL );
// initialize cycle time variance measurement with defaults
CycleTimeVariance.Init ( SYSTEM_FRAME_SIZE_SAMPLES,
SYSTEM_SAMPLE_RATE_HZ, TIME_MOV_AV_RESPONSE_SECONDS );
// initialize channel name
ResetName();
// Connections -------------------------------------------------------------
QObject::connect ( &Protocol,
SIGNAL ( MessReadyForSending ( CVector<uint8_t> ) ),
this, SLOT ( OnSendProtMessage ( CVector<uint8_t> ) ) );
QObject::connect ( &Protocol,
SIGNAL ( ChangeJittBufSize ( int ) ),
this, SLOT ( OnJittBufSizeChange ( int ) ) );
QObject::connect ( &Protocol,
SIGNAL ( ReqJittBufSize() ),
SIGNAL ( ReqJittBufSize() ) );
QObject::connect ( &Protocol,
SIGNAL ( ReqChanName() ),
SIGNAL ( ReqChanName() ) );
QObject::connect ( &Protocol,
SIGNAL ( ReqConnClientsList() ),
SIGNAL ( ReqConnClientsList() ) );
QObject::connect ( &Protocol,
SIGNAL ( ConClientListMesReceived ( CVector<CChannelShortInfo> ) ),
SIGNAL ( ConClientListMesReceived ( CVector<CChannelShortInfo> ) ) );
QObject::connect( &Protocol, SIGNAL ( ChangeChanGain ( int, double ) ),
this, SLOT ( OnChangeChanGain ( int, double ) ) );
QObject::connect( &Protocol, SIGNAL ( ChangeChanName ( QString ) ),
this, SLOT ( OnChangeChanName ( QString ) ) );
QObject::connect( &Protocol,
SIGNAL ( ChatTextReceived ( QString ) ),
SIGNAL ( ChatTextReceived ( QString ) ) );
QObject::connect ( &Protocol,
SIGNAL ( DetectedCLMessage ( CVector<uint8_t>, int ) ),
SIGNAL ( DetectedCLMessage ( CVector<uint8_t>, int ) ) );
QObject::connect ( &Protocol,
SIGNAL ( NetTranspPropsReceived ( CNetworkTransportProps ) ),
this, SLOT ( OnNetTranspPropsReceived ( CNetworkTransportProps ) ) );
QObject::connect ( &Protocol,
SIGNAL ( ReqNetTranspProps() ),
this, SLOT ( OnReqNetTranspProps() ) );
}
bool CChannel::ProtocolIsEnabled()
{
// for the server, only enable protocol if the channel is connected, i.e.,
// successfully audio packets are received from a client
// for the client, enable protocol if the channel is enabled, i.e., the
// connection button was hit by the user
if ( bIsServer )
{
return IsConnected();
}
else
{
return bIsEnabled;
}
}
void CChannel::SetEnable ( const bool bNEnStat )
{
QMutexLocker locker ( &Mutex );
// set internal parameter
bIsEnabled = bNEnStat;
// if channel is not enabled, reset time out count and protocol
if ( !bNEnStat )
{
iConTimeOut = 0;
Protocol.Reset();
}
}
void CChannel::SetAudioStreamProperties ( const int iNewNetwFrameSize,
const int iNewNetwFrameSizeFact,
const int iNewNumAudioChannels )
{
// this function is intended for the server (not the client)
QMutexLocker locker ( &Mutex );
// store new values
iNumAudioChannels = iNewNumAudioChannels;
iNetwFrameSize = iNewNetwFrameSize;
iNetwFrameSizeFact = iNewNetwFrameSizeFact;
// init socket buffer
SockBuf.Init ( iNetwFrameSize, iCurSockBufNumFrames );
// init conversion buffer
ConvBuf.Init ( iNetwFrameSize * iNetwFrameSizeFact );
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// reset cycle time variance measurement
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CycleTimeVariance.Reset();
// tell the server that audio coding has changed
CreateNetTranspPropsMessFromCurrentSettings();
}
bool CChannel::SetSockBufNumFrames ( const int iNewNumFrames,
const bool bPreserve )
{
QMutexLocker locker ( &Mutex ); // this operation must be done with mutex
// 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;
// the network block size is a multiple of the minimum network
// block size
SockBuf.Init ( iNetwFrameSize, iNewNumFrames, bPreserve );
// only in case we are the server and auto jitter buffer setting is
// enabled, we have to report the current setting to the client
if ( bIsServer && bDoAutoSockBufSize )
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{
// we cannot call the "CreateJitBufMes" function directly since
// this would give us problems with different threads (e.g. the
// timer thread) and the protocol mechanism (problem with
// qRegisterMetaType(), etc.)
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emit ServerAutoSockBufSizeChange ( iNewNumFrames );
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}
return false; // -> no error
}
}
return true; // set error flag
}
void CChannel::SetDoAutoSockBufSize ( const bool bValue )
{
QMutexLocker locker ( &Mutex );
// only act on new value if it is different from the current one
if ( bDoAutoSockBufSize != bValue )
{
if ( bValue )
{
// in case auto socket buffer size was just enabled, reset statistic
CycleTimeVariance.Reset();
}
// store new setting
bDoAutoSockBufSize = bValue;
}
}
void CChannel::SetGain ( const int iChanID,
const double dNewGain )
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{
QMutexLocker locker ( &Mutex );
// set value (make sure channel ID is in range)
if ( ( iChanID >= 0 ) && ( iChanID < MAX_NUM_CHANNELS ) )
{
vecdGains[iChanID] = dNewGain;
}
}
double CChannel::GetGain ( const int iChanID )
{
QMutexLocker locker ( &Mutex );
// get value (make sure channel ID is in range)
if ( ( iChanID >= 0 ) && ( iChanID < MAX_NUM_CHANNELS ) )
{
return vecdGains[iChanID];
}
else
{
return 0;
}
}
void CChannel::SetName ( const QString strNewName )
{
bool bNameHasChanged = false;
Mutex.lock();
{
// apply value (if different from previous name)
if ( sName.compare ( strNewName ) )
{
sName = strNewName;
bNameHasChanged = true;
}
}
Mutex.unlock();
// fire message that name has changed
if ( bNameHasChanged )
{
// the "emit" has to be done outside the mutexed region
emit NameHasChanged();
}
}
QString CChannel::GetName()
{
// make sure the string is not written at the same time when it is
// read here -> use mutex to secure access
QMutexLocker locker ( &Mutex );
return sName;
}
void CChannel::OnSendProtMessage ( CVector<uint8_t> vecMessage )
{
// only send messages if protocol is enabled, otherwise delete complete
// queue
if ( ProtocolIsEnabled() )
{
// emit message to actually send the data
emit MessReadyForSending ( vecMessage );
}
else
{
// delete send message queue
Protocol.Reset();
}
}
void CChannel::OnJittBufSizeChange ( int iNewJitBufSize )
{
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// for server apply setting, for client emit message
if ( bIsServer )
{
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// first check for special case: auto setting
if ( iNewJitBufSize == AUTO_NET_BUF_SIZE_FOR_PROTOCOL )
{
SetDoAutoSockBufSize ( true );
}
else
{
// manual setting is received, turn OFF auto setting and apply new value
SetDoAutoSockBufSize ( false );
SetSockBufNumFrames ( iNewJitBufSize, true );
}
}
else
{
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emit JittBufSizeChanged ( iNewJitBufSize );
}
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}
void CChannel::OnChangeChanGain ( int iChanID,
double dNewGain )
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{
SetGain ( iChanID, dNewGain );
}
void CChannel::OnChangeChanName ( QString strName )
{
SetName ( strName );
}
bool CChannel::GetAddress ( CHostAddress& RetAddr )
{
QMutexLocker locker ( &Mutex );
if ( IsConnected() )
{
RetAddr = InetAddr;
return true;
}
else
{
RetAddr = CHostAddress();
return false;
}
}
void CChannel::OnNetTranspPropsReceived ( CNetworkTransportProps NetworkTransportProps )
{
// only the server shall act on network transport properties message
if ( bIsServer )
{
QMutexLocker locker ( &Mutex );
// store received parameters
iNumAudioChannels = NetworkTransportProps.iNumAudioChannels;
iNetwFrameSizeFact = NetworkTransportProps.iBlockSizeFact;
iNetwFrameSize =
NetworkTransportProps.iBaseNetworkPacketSize;
// update socket buffer (the network block size is a multiple of the
// minimum network frame size
SockBuf.Init ( iNetwFrameSize, iCurSockBufNumFrames );
// init conversion buffer
ConvBuf.Init ( iNetwFrameSize * iNetwFrameSizeFact );
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// reset statistic
CycleTimeVariance.Reset();
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}
}
void CChannel::OnReqNetTranspProps()
{
CreateNetTranspPropsMessFromCurrentSettings();
}
void CChannel::CreateNetTranspPropsMessFromCurrentSettings()
{
CNetworkTransportProps NetworkTransportProps (
iNetwFrameSize,
iNetwFrameSizeFact,
iNumAudioChannels,
SYSTEM_SAMPLE_RATE_HZ,
CT_CELT, // always CELT coding
0, // version of the codec
0 );
// send current network transport properties
Protocol.CreateNetwTranspPropsMes ( NetworkTransportProps );
}
void CChannel::Disconnect()
{
// we only have to disconnect the channel if it is actually connected
if ( IsConnected() )
{
// set time out counter to a small value > 0 so that the next time a
// received audio block is queried, the disconnection is performed
// (assuming that no audio packet is received in the meantime)
iConTimeOut = 1; // a small number > 0
}
}
EPutDataStat CChannel::PutData ( const CVector<uint8_t>& vecbyData,
int iNumBytes )
{
EPutDataStat eRet = PS_GEN_ERROR;
// init flags
bool bIsProtocolPacket = false;
bool bIsAudioPacket = false;
bool bNewConnection = false;
if ( bIsEnabled )
{
// first check if this is protocol data
// only use protocol data if protocol mechanism is enabled
if ( ProtocolIsEnabled() )
{
// parse the message assuming this is a protocol message
if ( !Protocol.ParseMessage ( vecbyData, iNumBytes ) )
{
// set status flags
eRet = PS_PROT_OK;
bIsProtocolPacket = true;
}
}
else
{
// In case we are the server and the current channel is not
// connected, we do not evaluate protocal messages but these
// messages could start the server which is not desired, especially
// not for the disconnect messages.
// We now do not start the server if a valid protocol message
// was received but only start the server on audio packets
if ( Protocol.IsProtocolMessage ( vecbyData, iNumBytes ) )
{
// set status flags
eRet = PS_PROT_OK_MESS_NOT_EVALUATED;
bIsProtocolPacket = true;
}
}
// only try to parse audio if it was not a protocol packet
if ( !bIsProtocolPacket )
{
Mutex.lock();
{
// TODO only process data if network properties protocol message has been arrived
// only process audio if packet has correct size
if ( iNumBytes == ( iNetwFrameSize * iNetwFrameSizeFact ) )
{
// set audio packet flag
bIsAudioPacket = true;
// store new packet in jitter buffer
if ( SockBuf.Put ( vecbyData, iNumBytes ) )
{
eRet = PS_AUDIO_OK;
}
else
{
eRet = PS_AUDIO_ERR;
}
// update cycle time variance measurement (this is only
// used in case auto socket buffer size is enabled)
if ( bDoAutoSockBufSize )
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{
CycleTimeVariance.Update();
}
}
else
{
// the protocol parsing failed and this was no audio block,
// we treat this as protocol error (unkown packet)
eRet = PS_PROT_ERR;
}
// all network packets except of valid llcon protocol messages
// regardless if they are valid or invalid audio packets lead to
// a state change to a connected channel
// this is because protocol messages can only be sent on a
// connected channel and the client has to inform the server
// about the audio packet properties via the protocol
// check if channel was not connected, this is a new connection
bNewConnection = !IsConnected();
// reset time-out counter
ResetTimeOutCounter();
}
Mutex.unlock();
}
if ( bNewConnection )
{
// if this is a new connection and the current network packet is
// neither an audio or protocol packet, we have to query the
// network transport properties for the audio packets
// (this is only required for server since we defined that the
// server has to send with the same properties as sent by
// the client)
// TODO check the conditions: !bIsProtocolPacket should always be true
// since we can only get here if bNewConnection, should we really put
// !bIsAudioPacket in here, because shouldn't we always query the audio
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// properties on a new connection?
if ( bIsServer && ( !bIsProtocolPacket ) && ( !bIsAudioPacket ) )
{
Protocol.CreateReqNetwTranspPropsMes();
}
// reset cycle time variance measurement
CycleTimeVariance.Reset();
// inform other objects that new connection was established
emit NewConnection();
}
}
return eRet;
}
EGetDataStat CChannel::GetData ( CVector<uint8_t>& vecbyData )
{
QMutexLocker locker ( &Mutex );
EGetDataStat eGetStatus;
const bool bSockBufState = SockBuf.Get ( vecbyData );
// decrease time-out counter
if ( iConTimeOut > 0 )
{
// subtract the number of samples of the current block since the
// time out counter is based on samples not on blocks (definition:
// always one atomic block is get by using the GetData() function
// where the atomic block size is "SYSTEM_FRAME_SIZE_SAMPLES")
// TODO this code only works with the above assumption -> better
// implementation so that we are not depending on assumptions
iConTimeOut -= SYSTEM_FRAME_SIZE_SAMPLES;
if ( iConTimeOut <= 0 )
{
// channel is just disconnected
eGetStatus = GS_CHAN_NOW_DISCONNECTED;
iConTimeOut = 0; // make sure we do not have negative values
// emit message
emit Disconnected();
}
else
{
if ( bSockBufState )
{
// everything is ok
eGetStatus = GS_BUFFER_OK;
}
else
{
// channel is not yet disconnected but no data in buffer
eGetStatus = GS_BUFFER_UNDERRUN;
}
}
}
else
{
// channel is disconnected
eGetStatus = GS_CHAN_NOT_CONNECTED;
}
return eGetStatus;
}
CVector<uint8_t> CChannel::PrepSendPacket ( const CVector<uint8_t>& vecbyNPacket )
{
QMutexLocker locker ( &Mutex );
// if the block is not ready we have to initialize with zero length to
// tell the following network send routine that nothing should be sent
CVector<uint8_t> vecbySendBuf ( 0 );
// use conversion buffer to convert sound card block size in network
// block size
if ( ConvBuf.Put ( vecbyNPacket ) )
{
// a packet is ready
vecbySendBuf.Init ( iNetwFrameSize * iNetwFrameSizeFact );
vecbySendBuf = ConvBuf.Get();
}
return vecbySendBuf;
}
int CChannel::GetUploadRateKbps()
{
const int iAudioSizeOut = iNetwFrameSizeFact * SYSTEM_FRAME_SIZE_SAMPLES;
// we assume that the UDP packet which is transported via IP has an
// additional header size of
// 8 (UDP) + 20 (IP without optional fields) = 28 bytes
return ( iNetwFrameSize * iNetwFrameSizeFact + 28 /* header */ ) *
8 /* bits per byte */ *
SYSTEM_SAMPLE_RATE_HZ / iAudioSizeOut / 1000;
}
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void CChannel::UpdateSocketBufferSize ( const double dLocalStdDev )
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{
// just update the socket buffer size if auto setting is enabled, otherwise
// do nothing
if ( bDoAutoSockBufSize )
{
// TEST
SetSockBufNumFrames ( SockBuf.GetAutoSetting(), true );
/*
#ifdef _WIN32
// TEST
static FILE* pFile = fopen ( "c:\\temp\\test.dat", "w" );
fprintf ( pFile, "%d\n", SockBuf.GetAutoSetting() );
fflush ( pFile );
#endif
*/
/*
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// 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
// - we need at least a jitter buffer size of the audio packet duration
// -> add audio buffer duration
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const double dHysteresis = 0.2;
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// 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
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const double dEstCurBufSet = ( SYSTEM_BLOCK_DURATION_MS_FLOAT +
3.3 * ( CycleTimeVariance.GetStdDev() + dLocalStdDev ) ) /
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SYSTEM_BLOCK_DURATION_MS_FLOAT + 0.5;
*/
/*
// TEST
//if (bIsServer) {
static FILE* pFile = fopen ( "c:\\temp\\test.dat", "w" );
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fprintf ( pFile, "%e %e %e\n", CycleTimeVariance.GetStdDev(), dLocalStdDev, dEstCurBufSet );
fflush ( pFile );
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// close;x=read('c:/temp/test.dat',-1,3);plot(x)
//}
*/
/*
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// 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 )
{
// updatet the socket buffer size with the new value
SetSockBufNumFrames ( iUpperHystDec, true );
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}
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.
// updatet the socket buffer size with the new value
SetSockBufNumFrames ( iUpperHystDec, true );
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}
}
*/
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}
}