fix for auto jitter buffer setting

This commit is contained in:
Volker Fischer 2009-08-03 05:50:52 +00:00
parent 36c1bc1da4
commit 5d8d6eecb2

View file

@ -382,19 +382,16 @@ void CClient::ProcessAudioData ( CVector<int16_t>& vecsStereoSndCrd )
} }
} }
// send it through the network // encode current audio frame with CELT encoder
// Socket.SendPacket ( Channel.PrepSendPacket ( vecsNetwork ), celt_encode ( CeltEncoder,
// Channel.GetAddress() ); &vecsNetwork[0],
NULL,
celt_encode ( CeltEncoder, &vecCeltData[0],
&vecsNetwork[0], iCeltNumCodedBytes );
NULL,
&vecCeltData[0],
iCeltNumCodedBytes );
Socket.SendPacket ( vecCeltData, Channel.GetAddress() );
// send coded audio through the network
Socket.SendPacket ( Channel.PrepSendPacket ( vecCeltData ),
Channel.GetAddress() );
// receive a new block // receive a new block
@ -480,13 +477,8 @@ void CClient::UpdateSocketBufferSize()
const double dHysteresis = 0.3; const double dHysteresis = 0.3;
// calculate current buffer setting // calculate current buffer setting
// Use worst case scenario: We add the block size of input and
// output. This is not required if the smaller block size is a
// multiple of the bigger size, but in the general case where
// the block sizes do not have this relation, we require to have
// a minimum buffer size of the sum of both sizes
const double dAudioBufferDurationMs = const double dAudioBufferDurationMs =
( 2 * iMonoBlockSizeSam ) * 1000 / SYSTEM_SAMPLE_RATE; iMonoBlockSizeSam * 1000 / SYSTEM_SAMPLE_RATE;
// accumulate the standard deviations of input network stream and // accumulate the standard deviations of input network stream and
// internal timer, // internal timer,