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bug fix, support for 24/32 bit for ASIO interface

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This commit is contained in:
Volker Fischer 2008-07-24 18:15:48 +00:00
parent a80ef7a9ca
commit c067103684
2 changed files with 82 additions and 15 deletions
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6
src/settings.cpp
View file

@ -101,6 +101,12 @@ void CSettings::ReadIniFile ( const QString& sFileName )
pClient->SetNetwBufSizeFactIn ( iValue ); pClient->SetNetwBufSizeFactIn ( iValue );
} }
// network buffer size factor out
if ( GetNumericIniSet ( IniXMLDocument, "client", "netwbusifactout", 1, MAX_NET_BLOCK_SIZE_FACTOR, iValue ) )
{
pClient->SetNetwBufSizeFactOut ( iValue );
}
// flag whether the chat window shall be opened on a new chat message // flag whether the chat window shall be opened on a new chat message
if ( GetFlagIniSet ( IniXMLDocument, "client", "openchatonnewmessage", bValue ) ) if ( GetFlagIniSet ( IniXMLDocument, "client", "openchatonnewmessage", bValue ) )
{ {

73
windows/sound.cpp
View file

@ -356,9 +356,11 @@ void CSound::InitRecordingAndPlayback ( int iNewBufferSize )
ASIOGetChannelInfo ( &channelInfos[i] ); ASIOGetChannelInfo ( &channelInfos[i] );
// only 16 bit is supported // only 16 bit is supported
if ( channelInfos[i].type != ASIOSTInt16LSB ) if ( ( channelInfos[i].type != ASIOSTInt16LSB ) &&
( channelInfos[i].type != ASIOSTInt24LSB ) &&
( channelInfos[i].type != ASIOSTInt32LSB ) )
{ {
throw CGenErr ( "Required audio sample format not available (16 bit LSB)." ); throw CGenErr ( "Required audio sample format not available (16/24/32 bit LSB)." );
} }
} }
@ -593,13 +595,44 @@ void CSound::bufferSwitch ( long index, ASIOBool processNow )
// first check if space in buffer is available // first check if space in buffer is available
if ( !bCaptureBufferOverrun ) if ( !bCaptureBufferOverrun )
{ {
// copy new captured block in thread transfer buffer // copy new captured block in thread transfer buffer (copy
// mono data interleaved in stereo buffer)
switch ( channelInfos[i].type )
{
case ASIOSTInt16LSB:
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ ) for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{ {
// copy mono data interleaved in stereo buffer
psCaptureBuffer[iBufferPosCapture + psCaptureBuffer[iBufferPosCapture +
2 * iCurSample + bufferInfos[i].channelNum] = 2 * iCurSample + bufferInfos[i].channelNum] =
((short*) bufferInfos[i].buffers[index])[iCurSample]; ((short*) bufferInfos[i].buffers[index])[iCurSample];
}
break;
case ASIOSTInt24LSB:
// not yet tested, horrible things might happen with the following code ;-)
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert current sample in 16 bit format
int iCurSam = 0;
memcpy ( &iCurSam, ((char*) bufferInfos[i].buffers[index]) + iCurSample * 3, 3);
iCurSam >>= 8;
psCaptureBuffer[iBufferPosCapture +
2 * iCurSample + bufferInfos[i].channelNum] = static_cast<short> ( iCurSam );
}
break;
case ASIOSTInt32LSB:
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 16 bit
psCaptureBuffer[iBufferPosCapture +
2 * iCurSample + bufferInfos[i].channelNum] =
(((int*) bufferInfos[i].buffers[index])[iCurSample] >> 16);
}
break;
} }
} }
} }
@ -608,12 +641,40 @@ void CSound::bufferSwitch ( long index, ASIOBool processNow )
// PLAYBACK ---------------------------------------------------- // PLAYBACK ----------------------------------------------------
if ( !bPlayBufferUnderrun ) if ( !bPlayBufferUnderrun )
{ {
// copy data from sound card in output buffer // copy data from sound card in output buffer (copy
// interleaved stereo data in mono sound card buffer)
switch ( channelInfos[i].type )
{
case ASIOSTInt16LSB:
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ ) for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{ {
// copy interleaved stereo data in mono sound card buffer
((short*) bufferInfos[i].buffers[index])[iCurSample] = ((short*) bufferInfos[i].buffers[index])[iCurSample] =
psPlayBuffer[2 * iCurSample + bufferInfos[i].channelNum]; psPlayBuffer[2 * iCurSample + bufferInfos[i].channelNum];
}
break;
case ASIOSTInt24LSB:
// not yet tested, horrible things might happen with the following code ;-)
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert current sample in 24 bit format
int iCurSam = static_cast<int> ( psPlayBuffer[2 * iCurSample + bufferInfos[i].channelNum] );
iCurSam <<= 8;
memcpy ( ((char*) bufferInfos[i].buffers[index]) + iCurSample * 3, &iCurSam, 3);
}
break;
case ASIOSTInt32LSB:
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit
int iCurSam = static_cast<int> ( psPlayBuffer[2 * iCurSample + bufferInfos[i].channelNum] );
((int*) bufferInfos[i].buffers[index])[iCurSample] = ( iCurSam << 16 );
}
break;
} }
} }
} }