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added some more ASIO sample conversions

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This commit is contained in:
Volker Fischer 2010-02-04 20:25:57 +00:00
parent 75b1994257
commit 9f3d07ca67
3 changed files with 121 additions and 21 deletions
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2
src/client.h
View file

@ -142,7 +142,7 @@ public:
// sound card conversion buffer is used or not
if ( bSndCrdConversionBufferRequired )
{
return iSndCardMonoBlockSizeSamConvBuff;
return iSndCardMonoBlockSizeSamConvBuff + iMonoBlockSizeSam;
}
else
{

15
src/clientsettingsdlg.cpp
View file

@ -8,16 +8,16 @@
*
* 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
* 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
* 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
* 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.,
* this program; if not, write to the Free Software Foundation, Inc.,
* 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
\******************************************************************************/
@ -103,12 +103,15 @@ CClientSettingsDlg::CClientSettingsDlg ( CClient* pNCliP, QWidget* parent,
"<li>512 samples: This setting should only be used if only a very slow "
"computer or a slow internet connection is available.</li>"
"</ul>"
"Some sound card driver to not allow the buffer delay to be changed "
"Some sound card driver do not allow the buffer delay to be changed "
"from within the llcon software. In this case the buffer delay setting "
"is disabled. To change the actual buffer delay, this "
"setting has to be changed in the sound card driver. On Windows, press "
"the ASIO Setup button to open the driver settings panel. On Linux, "
"use the Jack configuration tool to change the buffer size.<br>"
"If no buffer size is selected and all settings are disabled, a "
"unsupported bufffer size is used by the driver. The llcon software "
"will still work with this setting but with restricted performannce.<br>"
"The actual buffer delay has influence on the connection status, the "
"current upload rate and the overall delay. The lower the buffer size, "
"the higher the probability of red light in the status indicator (drop "
@ -159,7 +162,7 @@ CClientSettingsDlg::CClientSettingsDlg ( CClient* pNCliP, QWidget* parent,
cbUseHighQualityAudio->setAccessibleName ( tr ( "Use high quality audio "
"check box" ) );
// current connection status parameter
// current connection status parameter
QString strConnStats = tr ( "<b>Current Connection Status "
"Parameter:</b> The ping time is the time required for the audio "
"stream to travel from the client to the server and backwards. This "

125
windows/sound.cpp
View file

@ -639,7 +639,16 @@ void CSound::bufferSwitch ( long index, ASIOBool processNow )
case ASIOSTInt24MSB:
// NOT YET TESTED
// TODO
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// because the bits are flipped, we do not have to perform the
// shift by 8 bits
int iCurSam = 0;
memcpy ( &iCurSam, ( (char*) bufferInfos[i].buffers[index] ) + iCurSample * 3, 3 );
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
Flip16Bits ( static_cast<int16_t> ( iCurSam ) );
}
break;
case ASIOSTInt32MSB:
@ -655,32 +664,64 @@ void CSound::bufferSwitch ( long index, ASIOBool processNow )
case ASIOSTFloat32MSB: // IEEE 754 32 bit float, as found on Intel x86 architecture
// NOT YET TESTED
// TODO
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( static_cast<float> (
Flip32Bits ( static_cast<int32_t*> (
bufferInfos[i].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
}
break;
case ASIOSTFloat64MSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture
// NOT YET TESTED
// TODO
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( static_cast<double> (
Flip64Bits ( static_cast<int64_t*> (
bufferInfos[i].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
}
break;
case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment
// NOT YET TESTED
// TODO
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( Flip32Bits ( static_cast<int32_t*> (
bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFF );
}
break;
case ASIOSTInt32MSB18: // 32 bit data with 18 bit alignment
// NOT YET TESTED
// TODO
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
bufferInfos[i].buffers[index] )[iCurSample] ) & 0x3FFFF ) >> 2 );
}
break;
case ASIOSTInt32MSB20: // 32 bit data with 20 bit alignment
// NOT YET TESTED
// TODO
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFFF ) >> 4 );
}
break;
case ASIOSTInt32MSB24: // 32 bit data with 24 bit alignment
// NOT YET TESTED
// TODO
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] =
static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFFFF ) >> 8 );
}
break;
}
}
@ -823,7 +864,15 @@ void CSound::bufferSwitch ( long index, ASIOBool processNow )
case ASIOSTInt24MSB:
// NOT YET TESTED
// TODO
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// because the bits are flipped, we do not have to perform the
// shift by 8 bits
int32_t iCurSam = static_cast<int32_t> ( Flip16Bits (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] ) );
memcpy ( ( (char*) bufferInfos[i].buffers[index] ) + iCurSample * 3, &iCurSam, 3 );
}
break;
case ASIOSTInt32MSB:
@ -841,32 +890,80 @@ void CSound::bufferSwitch ( long index, ASIOBool processNow )
case ASIOSTFloat32MSB: // IEEE 754 32 bit float, as found on Intel x86 architecture
// NOT YET TESTED
// TODO
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
const float fCurSam = static_cast<float> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<float*> ( bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<float> ( Flip32Bits ( static_cast<int32_t> (
fCurSam / _MAXSHORT ) ) );
}
break;
case ASIOSTFloat64MSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture
// NOT YET TESTED
// TODO
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
const double fCurSam = static_cast<double> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<float*> ( bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<double> ( Flip64Bits ( static_cast<int64_t> (
fCurSam / _MAXSHORT ) ) );
}
break;
case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment
// NOT YET TESTED
// TODO
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
Flip32Bits ( iCurSam );
}
break;
case ASIOSTInt32MSB18: // 32 bit data with 18 bit alignment
// NOT YET TESTED
// TODO
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
Flip32Bits ( iCurSam << 2 );
}
break;
case ASIOSTInt32MSB20: // 32 bit data with 20 bit alignment
// NOT YET TESTED
// TODO
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
Flip32Bits ( iCurSam << 4 );
}
break;
case ASIOSTInt32MSB24: // 32 bit data with 24 bit alignment
// NOT YET TESTED
// TODO
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + bufferInfos[i].channelNum] );
static_cast<int32_t*> ( bufferInfos[i].buffers[index] )[iCurSample] =
Flip32Bits ( iCurSam << 8 );
}
break;
}
}