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prepare for special summing channels

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This commit is contained in:
Volker Fischer 2018-03-25 11:21:58 +00:00
parent 29332269c9
commit 7f41f186d4
1 changed files with 95 additions and 101 deletions
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192
windows/sound.cpp
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@ -91,6 +91,7 @@ QString CSound::CheckDeviceCapabilities()
// check the sample rate
const ASIOError CanSaRateReturn = ASIOCanSampleRate ( SYSTEM_SAMPLE_RATE_HZ );
if ( ( CanSaRateReturn == ASE_NoClock ) ||
( CanSaRateReturn == ASE_NotPresent ) )
{
@ -102,6 +103,7 @@ QString CSound::CheckDeviceCapabilities()
// check the number of available channels
ASIOGetChannels ( &lNumInChan, &lNumOutChan );
if ( ( lNumInChan < NUM_IN_OUT_CHANNELS ) ||
( lNumOutChan < NUM_IN_OUT_CHANNELS ) )
{
@ -323,8 +325,7 @@ int CSound::Init ( const int iNewPrefMonoBufferSize )
{
// get the actual sound card buffer size which is supported
// by the audio hardware
iASIOBufferSizeMono =
GetActualBufferSize ( iNewPrefMonoBufferSize );
iASIOBufferSizeMono = GetActualBufferSize ( iNewPrefMonoBufferSize );
// init base class
CSoundBase::Init ( iASIOBufferSizeMono );
@ -342,25 +343,25 @@ int CSound::Init ( const int iNewPrefMonoBufferSize )
// dispose old buffers (if any)
ASIODisposeBuffers();
// init buffer infos, we always want to have two input and
// two output channels
for ( int i = 0; i < NUM_IN_OUT_CHANNELS; i++ )
{
// prepare input channels
for ( int i = 0; i < lNumInChan; i++ )
{
bufferInfos[i].isInput = ASIOTrue;
bufferInfos[i].channelNum = vSelectedInputChannels[i];
bufferInfos[i].channelNum = i;
bufferInfos[i].buffers[0] = 0;
bufferInfos[i].buffers[1] = 0;
// prepare output channels
bufferInfos[NUM_IN_OUT_CHANNELS + i].isInput = ASIOFalse;
bufferInfos[NUM_IN_OUT_CHANNELS + i].channelNum = vSelectedOutputChannels[i];
bufferInfos[NUM_IN_OUT_CHANNELS + i].buffers[0] = 0;
bufferInfos[NUM_IN_OUT_CHANNELS + i].buffers[1] = 0;
}
ASIOCreateBuffers ( bufferInfos,
2 /* in/out */ * NUM_IN_OUT_CHANNELS /* stereo */,
// prepare output channels
for ( int i = 0; i < lNumOutChan; i++ )
{
bufferInfos[lNumInChan + i].isInput = ASIOFalse;
bufferInfos[lNumInChan + i].channelNum = i;
bufferInfos[lNumInChan + i].buffers[0] = 0;
bufferInfos[lNumInChan + i].buffers[1] = 0;
}
ASIOCreateBuffers ( bufferInfos, lNumInChan + lNumOutChan,
iASIOBufferSizeMono, &asioCallbacks );
// query the latency of the driver
@ -529,22 +530,20 @@ void CSound::bufferSwitch ( long index, ASIOBool )
// CAPTURE -------------------------------------------------------------
for ( int i = 0; i < NUM_IN_OUT_CHANNELS; i++ )
{
const int iInChNum = i;
const int iSelCH = pSound->vSelectedInputChannels[i];
// copy new captured block in thread transfer buffer (copy
// mono data interleaved in stereo buffer)
switch ( pSound->channelInfosInput[pSound->vSelectedInputChannels[i]].type )
switch ( pSound->channelInfosInput[iSelCH].type )
{
case ASIOSTInt16LSB:
{
// no type conversion required, just copy operation
int16_t* pASIOBuf =
static_cast<int16_t*> ( pSound->bufferInfos[i].buffers[index] );
int16_t* pASIOBuf = static_cast<int16_t*> ( pSound->bufferInfos[iSelCH].buffers[index] );
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
pASIOBuf[iCurSample];
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] = pASIOBuf[iCurSample];
}
break;
}
@ -554,22 +553,20 @@ void CSound::bufferSwitch ( long index, ASIOBool )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
int iCurSam = 0;
memcpy ( &iCurSam, ( (char*) pSound->bufferInfos[i].buffers[index] ) + iCurSample * 3, 3 );
memcpy ( &iCurSam, ( (char*) pSound->bufferInfos[iSelCH].buffers[index] ) + iCurSample * 3, 3 );
iCurSam >>= 8;
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
static_cast<int16_t> ( iCurSam );
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] = static_cast<int16_t> ( iCurSam );
}
break;
case ASIOSTInt32LSB:
{
int32_t* pASIOBuf =
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] );
int32_t* pASIOBuf = static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] );
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
static_cast<int16_t> ( pASIOBuf[iCurSample] >> 16 );
}
break;
@ -579,9 +576,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
static_cast<int16_t> ( static_cast<float*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] * _MAXSHORT );
pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] * _MAXSHORT );
}
break;
@ -589,9 +586,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
static_cast<int16_t> ( static_cast<double*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] * _MAXSHORT );
pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] * _MAXSHORT );
}
break;
@ -599,9 +596,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
static_cast<int16_t> ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] & 0xFFFF );
pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] & 0xFFFF );
}
break;
@ -609,9 +606,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
static_cast<int16_t> ( ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] & 0x3FFFF ) >> 2 );
pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] & 0x3FFFF ) >> 2 );
}
break;
@ -619,9 +616,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
static_cast<int16_t> ( ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] & 0xFFFFF ) >> 4 );
pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] & 0xFFFFF ) >> 4 );
}
break;
@ -629,9 +626,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
static_cast<int16_t> ( ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] & 0xFFFFFF ) >> 8 );
pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] & 0xFFFFFF ) >> 8 );
}
break;
@ -640,9 +637,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
// flip bits
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
Flip16Bits ( ( static_cast<int16_t*> (
pSound->bufferInfos[i].buffers[index] ) )[iCurSample] );
pSound->bufferInfos[iSelCH].buffers[index] ) )[iCurSample] );
}
break;
@ -653,9 +650,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
// 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 );
memcpy ( &iCurSam, ( (char*) pSound->bufferInfos[iSelCH].buffers[index] ) + iCurSample * 3, 3 );
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
Flip16Bits ( static_cast<int16_t> ( iCurSam ) );
}
break;
@ -665,9 +662,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// flip bits and convert to 16 bit
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
static_cast<int16_t> ( Flip32Bits ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) >> 16 );
pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) >> 16 );
}
break;
@ -675,10 +672,10 @@ void CSound::bufferSwitch ( long index, ASIOBool )
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
static_cast<int16_t> ( static_cast<float> (
Flip32Bits ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
}
break;
@ -686,10 +683,10 @@ void CSound::bufferSwitch ( long index, ASIOBool )
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
static_cast<int16_t> ( static_cast<double> (
Flip64Bits ( static_cast<int64_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
}
break;
@ -697,9 +694,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
static_cast<int16_t> ( Flip32Bits ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFF );
pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) & 0xFFFF );
}
break;
@ -707,9 +704,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) & 0x3FFFF ) >> 2 );
pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) & 0x3FFFF ) >> 2 );
}
break;
@ -717,9 +714,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFFF ) >> 4 );
pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) & 0xFFFFF ) >> 4 );
}
break;
@ -727,9 +724,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
// NOT YET TESTED
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
pSound->bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFFFF ) >> 8 );
pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) & 0xFFFFFF ) >> 8 );
}
break;
}
@ -739,24 +736,22 @@ void CSound::bufferSwitch ( long index, ASIOBool )
pSound->ProcessCallback ( vecsTmpAudioSndCrdStereo );
// PLAYBACK ------------------------------------------------------------
for ( int i = NUM_IN_OUT_CHANNELS; i < 2 * NUM_IN_OUT_CHANNELS; i++ )
for ( int i = 0; i < NUM_IN_OUT_CHANNELS; i++ )
{
const int iOutChNum = i - NUM_IN_OUT_CHANNELS;
const int iSelCH = pSound->lNumInChan + pSound->vSelectedOutputChannels[i];
// copy data from sound card in output buffer (copy
// interleaved stereo data in mono sound card buffer)
switch ( pSound->channelInfosOutput[pSound->vSelectedOutputChannels[iOutChNum]].type )
switch ( pSound->channelInfosOutput[pSound->vSelectedOutputChannels[i]].type )
{
case ASIOSTInt16LSB:
{
// no type conversion required, just copy operation
int16_t* pASIOBuf =
static_cast<int16_t*> ( pSound->bufferInfos[i].buffers[index] );
int16_t* pASIOBuf = static_cast<int16_t*> ( pSound->bufferInfos[iSelCH].buffers[index] );
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
pASIOBuf[iCurSample] =
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum];
pASIOBuf[iCurSample] = vecsTmpAudioSndCrdStereo[2 * iCurSample + i];
}
break;
}
@ -767,24 +762,23 @@ void CSound::bufferSwitch ( long index, ASIOBool )
{
// convert current sample in 24 bit format
int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
iCurSam <<= 8;
memcpy ( ( (char*) pSound->bufferInfos[i].buffers[index] ) + iCurSample * 3, &iCurSam, 3 );
memcpy ( ( (char*) pSound->bufferInfos[iSelCH].buffers[index] ) + iCurSample * 3, &iCurSam, 3 );
}
break;
case ASIOSTInt32LSB:
{
int32_t* pASIOBuf =
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] );
int32_t* pASIOBuf = static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] );
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
pASIOBuf[iCurSample] = ( iCurSam << 16 );
}
@ -796,9 +790,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
const float fCurSam = static_cast<float> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
static_cast<float*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<float*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
fCurSam / _MAXSHORT;
}
break;
@ -808,9 +802,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
const double fCurSam = static_cast<double> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
static_cast<double*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<double*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
fCurSam / _MAXSHORT;
}
break;
@ -821,9 +815,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
iCurSam;
}
break;
@ -834,9 +828,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
( iCurSam << 2 );
}
break;
@ -847,9 +841,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
( iCurSam << 4 );
}
break;
@ -860,9 +854,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
( iCurSam << 8 );
}
break;
@ -872,8 +866,8 @@ void CSound::bufferSwitch ( long index, ASIOBool )
// flip bits
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
( (int16_t*) pSound->bufferInfos[i].buffers[index] )[iCurSample] =
Flip16Bits ( vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
( (int16_t*) pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
Flip16Bits ( vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
}
break;
@ -884,9 +878,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
// 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 + iOutChNum] ) );
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] ) );
memcpy ( ( (char*) pSound->bufferInfos[i].buffers[index] ) + iCurSample * 3, &iCurSam, 3 );
memcpy ( ( (char*) pSound->bufferInfos[iSelCH].buffers[index] ) + iCurSample * 3, &iCurSam, 3 );
}
break;
@ -896,9 +890,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
{
// convert to 32 bit and flip bits
int iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
Flip32Bits ( iCurSam << 16 );
}
break;
@ -908,9 +902,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
const float fCurSam = static_cast<float> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
static_cast<float*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<float*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
static_cast<float> ( Flip32Bits ( static_cast<int32_t> (
fCurSam / _MAXSHORT ) ) );
}
@ -921,9 +915,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
{
const double fCurSam = static_cast<double> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
static_cast<float*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<float*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
static_cast<double> ( Flip64Bits ( static_cast<int64_t> (
fCurSam / _MAXSHORT ) ) );
}
@ -935,9 +929,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
Flip32Bits ( iCurSam );
}
break;
@ -948,9 +942,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
Flip32Bits ( iCurSam << 2 );
}
break;
@ -961,9 +955,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
Flip32Bits ( iCurSam << 4 );
}
break;
@ -974,9 +968,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
{
// convert to 32 bit
const int32_t iCurSam = static_cast<int32_t> (
vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] )[iCurSample] =
static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
Flip32Bits ( iCurSam << 8 );
}
break;