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some speed optimizations

This commit is contained in:
Volker Fischer 2013-12-14 22:16:20 +00:00
parent d4be4bc038
commit a0c318eeec
1 changed files with 102 additions and 82 deletions
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184
windows/sound.cpp
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@ -499,6 +499,10 @@ void CSound::bufferSwitch ( long index, ASIOBool )
{
int iCurSample;
// get references to class members
int& iASIOBufferSizeMono = pSound->iASIOBufferSizeMono;
CVector<int16_t>& vecsTmpAudioSndCrdStereo = pSound->vecsTmpAudioSndCrdStereo;
// perform the processing for input and output
pSound->ASIOMutex.lock(); // get mutex lock
{
@ -512,42 +516,50 @@ void CSound::bufferSwitch ( long index, ASIOBool )
switch ( pSound->channelInfosInput[pSound->vSelectedInputChannels[i]].type )
{
case ASIOSTInt16LSB:
{
// no type conversion required, just copy operation
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
int16_t* pASIOBuf =
static_cast<int16_t*> ( pSound->bufferInfos[i].buffers[index] );
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
static_cast<int16_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample];
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
pASIOBuf[iCurSample];
}
break;
}
case ASIOSTInt24LSB:
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
int iCurSam = 0;
memcpy ( &iCurSam, ( (char*) pSound->bufferInfos[i].buffers[index] ) + iCurSample * 3, 3 );
iCurSam >>= 8;
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
static_cast<int16_t> ( iCurSam );
}
break;
case ASIOSTInt32LSB:
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
{
int32_t* pASIOBuf =
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] );
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
static_cast<int16_t> ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] >> 16 );
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
static_cast<int16_t> ( pASIOBuf[iCurSample] >> 16 );
}
break;
}
case ASIOSTFloat32LSB: // IEEE 754 32 bit float, as found on Intel x86 architecture
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
static_cast<int16_t> ( static_cast<float*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] * _MAXSHORT );
}
@ -555,9 +567,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTFloat64LSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
static_cast<int16_t> ( static_cast<double*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] * _MAXSHORT );
}
@ -565,9 +577,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt32LSB16: // 32 bit data with 16 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
static_cast<int16_t> ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] & 0xFFFF );
}
@ -575,9 +587,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt32LSB18: // 32 bit data with 18 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
static_cast<int16_t> ( ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] & 0x3FFFF ) >> 2 );
}
@ -585,9 +597,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt32LSB20: // 32 bit data with 20 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
static_cast<int16_t> ( ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] & 0xFFFFF ) >> 4 );
}
@ -595,9 +607,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt32LSB24: // 32 bit data with 24 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
static_cast<int16_t> ( ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] & 0xFFFFFF ) >> 8 );
}
@ -606,9 +618,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt16MSB:
// NOT YET TESTED
// flip bits
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
Flip16Bits ( ( static_cast<int16_t*> (
pSound->bufferInfos[i].buffers[index] ) )[iCurSample] );
}
@ -616,24 +628,24 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt24MSB:
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
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*) pSound->bufferInfos[i].buffers[index] ) + iCurSample * 3, 3 );
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
Flip16Bits ( static_cast<int16_t> ( iCurSam ) );
}
break;
case ASIOSTInt32MSB:
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// flip bits and convert to 16 bit
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
static_cast<int16_t> ( Flip32Bits ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) >> 16 );
}
@ -641,9 +653,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTFloat32MSB: // IEEE 754 32 bit float, as found on Intel x86 architecture
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
static_cast<int16_t> ( static_cast<float> (
Flip32Bits ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
@ -652,9 +664,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTFloat64MSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
static_cast<int16_t> ( static_cast<double> (
Flip64Bits ( static_cast<int64_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
@ -663,9 +675,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
static_cast<int16_t> ( Flip32Bits ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFF );
}
@ -673,9 +685,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt32MSB18: // 32 bit data with 18 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) & 0x3FFFF ) >> 2 );
}
@ -683,9 +695,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt32MSB20: // 32 bit data with 20 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFFF ) >> 4 );
}
@ -693,9 +705,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt32MSB24: // 32 bit data with 24 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFFFF ) >> 8 );
}
@ -704,7 +716,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
}
// call processing callback function
pSound->ProcessCallback ( pSound->vecsTmpAudioSndCrdStereo );
pSound->ProcessCallback ( vecsTmpAudioSndCrdStereo );
// PLAYBACK ------------------------------------------------------------
for ( int i = NUM_IN_OUT_CHANNELS; i < 2 * NUM_IN_OUT_CHANNELS; i++ )
@ -716,21 +728,26 @@ void CSound::bufferSwitch ( long index, ASIOBool )
switch ( pSound->channelInfosOutput[pSound->vSelectedOutputChannels[iOutChNum]].type )
{
case ASIOSTInt16LSB:
{
// no type conversion required, just copy operation
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
int16_t* pASIOBuf =
static_cast<int16_t*> ( pSound->bufferInfos[i].buffers[index] );
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
static_cast<int16_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum];
pASIOBuf[iCurSample] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum];
}
break;
}
case ASIOSTInt24LSB:
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert current sample in 24 bit format
int32_t iCurSam = static_cast<int32_t> (
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
iCurSam <<= 8;
@ -739,24 +756,27 @@ void CSound::bufferSwitch ( long index, ASIOBool )
break;
case ASIOSTInt32LSB:
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
{
int32_t* pASIOBuf =
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] );
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
( iCurSam << 16 );
pASIOBuf[iCurSample] = ( iCurSam << 16 );
}
break;
}
case ASIOSTFloat32LSB: // IEEE 754 32 bit float, as found on Intel x86 architecture
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
const float fCurSam = static_cast<float> (
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
static_cast<float*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
fCurSam / _MAXSHORT;
@ -765,10 +785,10 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTFloat64LSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
const double fCurSam = static_cast<double> (
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
static_cast<double*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
fCurSam / _MAXSHORT;
@ -777,11 +797,11 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt32LSB16: // 32 bit data with 16 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
iCurSam;
@ -790,11 +810,11 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt32LSB18: // 32 bit data with 18 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
( iCurSam << 2 );
@ -803,11 +823,11 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt32LSB20: // 32 bit data with 20 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
( iCurSam << 4 );
@ -816,11 +836,11 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt32LSB24: // 32 bit data with 24 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
( iCurSam << 8 );
@ -830,21 +850,21 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt16MSB:
// NOT YET TESTED
// flip bits
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
( (int16_t*) pSound->bufferInfos[i].buffers[index] )[iCurSample] =
Flip16Bits ( pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
Flip16Bits ( vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
}
break;
case ASIOSTInt24MSB:
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
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 (
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ) );
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ) );
memcpy ( ( (char*) pSound->bufferInfos[i].buffers[index] ) + iCurSample * 3, &iCurSam, 3 );
}
@ -852,11 +872,11 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt32MSB:
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit and flip bits
int iCurSam = static_cast<int32_t> (
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
Flip32Bits ( iCurSam << 16 );
@ -865,10 +885,10 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTFloat32MSB: // IEEE 754 32 bit float, as found on Intel x86 architecture
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
const float fCurSam = static_cast<float> (
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
static_cast<float*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<float> ( Flip32Bits ( static_cast<int32_t> (
@ -878,10 +898,10 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTFloat64MSB: // IEEE 754 64 bit double float, as found on Intel x86 architecture
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
const double fCurSam = static_cast<double> (
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
static_cast<float*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<double> ( Flip64Bits ( static_cast<int64_t> (
@ -891,11 +911,11 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt32MSB16: // 32 bit data with 16 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
Flip32Bits ( iCurSam );
@ -904,11 +924,11 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt32MSB18: // 32 bit data with 18 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
Flip32Bits ( iCurSam << 2 );
@ -917,11 +937,11 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt32MSB20: // 32 bit data with 20 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
Flip32Bits ( iCurSam << 4 );
@ -930,11 +950,11 @@ void CSound::bufferSwitch ( long index, ASIOBool )
case ASIOSTInt32MSB24: // 32 bit data with 24 bit alignment
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
Flip32Bits ( iCurSam << 8 );
@ -943,7 +963,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
}
}
// finally if the driver supports the ASIOOutputReady() optimization,
// Finally if the driver supports the ASIOOutputReady() optimization,
// do it here, all data are in place -----------------------------------
if ( pSound->bASIOPostOutput )
{