prepare for special summing channels
This commit is contained in:
Volker Fischer
parent
29332269c9
commit
7f41f186d4
1 changed files with 95 additions and 101 deletions
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@ -91,6 +91,7 @@ QString CSound::CheckDeviceCapabilities()
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// check the sample rate
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const ASIOError CanSaRateReturn = ASIOCanSampleRate ( SYSTEM_SAMPLE_RATE_HZ );
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if ( ( CanSaRateReturn == ASE_NoClock ) ||
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( CanSaRateReturn == ASE_NotPresent ) )
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{
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@ -102,6 +103,7 @@ QString CSound::CheckDeviceCapabilities()
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// check the number of available channels
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ASIOGetChannels ( &lNumInChan, &lNumOutChan );
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if ( ( lNumInChan < NUM_IN_OUT_CHANNELS ) ||
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( lNumOutChan < NUM_IN_OUT_CHANNELS ) )
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{
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@ -323,8 +325,7 @@ int CSound::Init ( const int iNewPrefMonoBufferSize )
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{
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// get the actual sound card buffer size which is supported
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// by the audio hardware
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iASIOBufferSizeMono =
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GetActualBufferSize ( iNewPrefMonoBufferSize );
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iASIOBufferSizeMono = GetActualBufferSize ( iNewPrefMonoBufferSize );
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// init base class
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CSoundBase::Init ( iASIOBufferSizeMono );
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@ -342,26 +343,26 @@ int CSound::Init ( const int iNewPrefMonoBufferSize )
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// dispose old buffers (if any)
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ASIODisposeBuffers();
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// init buffer infos, we always want to have two input and
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// two output channels
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for ( int i = 0; i < NUM_IN_OUT_CHANNELS; i++ )
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// prepare input channels
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for ( int i = 0; i < lNumInChan; i++ )
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{
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// prepare input channels
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bufferInfos[i].isInput = ASIOTrue;
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bufferInfos[i].channelNum = vSelectedInputChannels[i];
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bufferInfos[i].channelNum = i;
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bufferInfos[i].buffers[0] = 0;
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bufferInfos[i].buffers[1] = 0;
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// prepare output channels
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bufferInfos[NUM_IN_OUT_CHANNELS + i].isInput = ASIOFalse;
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bufferInfos[NUM_IN_OUT_CHANNELS + i].channelNum = vSelectedOutputChannels[i];
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bufferInfos[NUM_IN_OUT_CHANNELS + i].buffers[0] = 0;
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bufferInfos[NUM_IN_OUT_CHANNELS + i].buffers[1] = 0;
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}
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ASIOCreateBuffers ( bufferInfos,
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2 /* in/out */ * NUM_IN_OUT_CHANNELS /* stereo */,
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iASIOBufferSizeMono, &asioCallbacks );
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// prepare output channels
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for ( int i = 0; i < lNumOutChan; i++ )
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{
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bufferInfos[lNumInChan + i].isInput = ASIOFalse;
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bufferInfos[lNumInChan + i].channelNum = i;
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bufferInfos[lNumInChan + i].buffers[0] = 0;
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bufferInfos[lNumInChan + i].buffers[1] = 0;
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}
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ASIOCreateBuffers ( bufferInfos, lNumInChan + lNumOutChan,
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iASIOBufferSizeMono, &asioCallbacks );
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// query the latency of the driver
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long lInputLatency = 0;
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@ -520,7 +521,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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int iCurSample;
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// get references to class members
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int& iASIOBufferSizeMono = pSound->iASIOBufferSizeMono;
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int& iASIOBufferSizeMono = pSound->iASIOBufferSizeMono;
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CVector<int16_t>& vecsTmpAudioSndCrdStereo = pSound->vecsTmpAudioSndCrdStereo;
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// perform the processing for input and output
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@ -529,22 +530,20 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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// CAPTURE -------------------------------------------------------------
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for ( int i = 0; i < NUM_IN_OUT_CHANNELS; i++ )
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{
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const int iInChNum = i;
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const int iSelCH = pSound->vSelectedInputChannels[i];
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// copy new captured block in thread transfer buffer (copy
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// mono data interleaved in stereo buffer)
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switch ( pSound->channelInfosInput[pSound->vSelectedInputChannels[i]].type )
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switch ( pSound->channelInfosInput[iSelCH].type )
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{
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case ASIOSTInt16LSB:
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{
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// no type conversion required, just copy operation
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int16_t* pASIOBuf =
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static_cast<int16_t*> ( pSound->bufferInfos[i].buffers[index] );
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int16_t* pASIOBuf = static_cast<int16_t*> ( pSound->bufferInfos[iSelCH].buffers[index] );
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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pASIOBuf[iCurSample];
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] = pASIOBuf[iCurSample];
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}
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break;
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}
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@ -554,22 +553,20 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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int iCurSam = 0;
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memcpy ( &iCurSam, ( (char*) pSound->bufferInfos[i].buffers[index] ) + iCurSample * 3, 3 );
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memcpy ( &iCurSam, ( (char*) pSound->bufferInfos[iSelCH].buffers[index] ) + iCurSample * 3, 3 );
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iCurSam >>= 8;
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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static_cast<int16_t> ( iCurSam );
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] = static_cast<int16_t> ( iCurSam );
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}
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break;
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case ASIOSTInt32LSB:
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{
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int32_t* pASIOBuf =
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static_cast<int32_t*> ( pSound->bufferInfos[i].buffers[index] );
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int32_t* pASIOBuf = static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] );
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
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static_cast<int16_t> ( pASIOBuf[iCurSample] >> 16 );
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}
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break;
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@ -579,9 +576,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
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static_cast<int16_t> ( static_cast<float*> (
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pSound->bufferInfos[i].buffers[index] )[iCurSample] * _MAXSHORT );
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pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] * _MAXSHORT );
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}
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break;
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@ -589,9 +586,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
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static_cast<int16_t> ( static_cast<double*> (
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pSound->bufferInfos[i].buffers[index] )[iCurSample] * _MAXSHORT );
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pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] * _MAXSHORT );
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}
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break;
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@ -599,9 +596,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
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static_cast<int16_t> ( static_cast<int32_t*> (
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pSound->bufferInfos[i].buffers[index] )[iCurSample] & 0xFFFF );
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pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] & 0xFFFF );
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}
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break;
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@ -609,9 +606,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
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static_cast<int16_t> ( ( static_cast<int32_t*> (
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pSound->bufferInfos[i].buffers[index] )[iCurSample] & 0x3FFFF ) >> 2 );
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pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] & 0x3FFFF ) >> 2 );
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}
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break;
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@ -619,9 +616,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
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static_cast<int16_t> ( ( static_cast<int32_t*> (
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pSound->bufferInfos[i].buffers[index] )[iCurSample] & 0xFFFFF ) >> 4 );
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pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] & 0xFFFFF ) >> 4 );
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}
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break;
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@ -629,9 +626,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
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static_cast<int16_t> ( ( static_cast<int32_t*> (
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pSound->bufferInfos[i].buffers[index] )[iCurSample] & 0xFFFFFF ) >> 8 );
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pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] & 0xFFFFFF ) >> 8 );
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}
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break;
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@ -640,9 +637,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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// flip bits
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
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Flip16Bits ( ( static_cast<int16_t*> (
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pSound->bufferInfos[i].buffers[index] ) )[iCurSample] );
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pSound->bufferInfos[iSelCH].buffers[index] ) )[iCurSample] );
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}
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break;
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@ -653,9 +650,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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// because the bits are flipped, we do not have to perform the
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// shift by 8 bits
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int iCurSam = 0;
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memcpy ( &iCurSam, ( (char*) pSound->bufferInfos[i].buffers[index] ) + iCurSample * 3, 3 );
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memcpy ( &iCurSam, ( (char*) pSound->bufferInfos[iSelCH].buffers[index] ) + iCurSample * 3, 3 );
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
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Flip16Bits ( static_cast<int16_t> ( iCurSam ) );
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}
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break;
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@ -665,9 +662,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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// flip bits and convert to 16 bit
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
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static_cast<int16_t> ( Flip32Bits ( static_cast<int32_t*> (
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pSound->bufferInfos[i].buffers[index] )[iCurSample] ) >> 16 );
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pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) >> 16 );
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}
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break;
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@ -675,10 +672,10 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
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static_cast<int16_t> ( static_cast<float> (
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Flip32Bits ( static_cast<int32_t*> (
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pSound->bufferInfos[i].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
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pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
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}
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break;
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@ -686,10 +683,10 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
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static_cast<int16_t> ( static_cast<double> (
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Flip64Bits ( static_cast<int64_t*> (
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pSound->bufferInfos[i].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
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pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
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}
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break;
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@ -697,9 +694,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
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static_cast<int16_t> ( Flip32Bits ( static_cast<int32_t*> (
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pSound->bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFF );
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pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) & 0xFFFF );
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}
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break;
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@ -707,9 +704,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
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static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
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pSound->bufferInfos[i].buffers[index] )[iCurSample] ) & 0x3FFFF ) >> 2 );
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pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) & 0x3FFFF ) >> 2 );
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}
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break;
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@ -717,9 +714,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
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static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
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pSound->bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFFF ) >> 4 );
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pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) & 0xFFFFF ) >> 4 );
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}
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break;
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@ -727,9 +724,9 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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// NOT YET TESTED
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] =
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
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static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
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pSound->bufferInfos[i].buffers[index] )[iCurSample] ) & 0xFFFFFF ) >> 8 );
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pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) & 0xFFFFFF ) >> 8 );
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}
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break;
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}
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@ -739,24 +736,22 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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pSound->ProcessCallback ( vecsTmpAudioSndCrdStereo );
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// PLAYBACK ------------------------------------------------------------
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for ( int i = NUM_IN_OUT_CHANNELS; i < 2 * NUM_IN_OUT_CHANNELS; i++ )
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for ( int i = 0; i < NUM_IN_OUT_CHANNELS; i++ )
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{
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const int iOutChNum = i - NUM_IN_OUT_CHANNELS;
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const int iSelCH = pSound->lNumInChan + pSound->vSelectedOutputChannels[i];
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// copy data from sound card in output buffer (copy
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// interleaved stereo data in mono sound card buffer)
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switch ( pSound->channelInfosOutput[pSound->vSelectedOutputChannels[iOutChNum]].type )
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switch ( pSound->channelInfosOutput[pSound->vSelectedOutputChannels[i]].type )
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{
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case ASIOSTInt16LSB:
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{
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// no type conversion required, just copy operation
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int16_t* pASIOBuf =
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static_cast<int16_t*> ( pSound->bufferInfos[i].buffers[index] );
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int16_t* pASIOBuf = static_cast<int16_t*> ( pSound->bufferInfos[iSelCH].buffers[index] );
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for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
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{
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pASIOBuf[iCurSample] =
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum];
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pASIOBuf[iCurSample] = vecsTmpAudioSndCrdStereo[2 * iCurSample + i];
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}
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break;
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}
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@ -767,24 +762,23 @@ void CSound::bufferSwitch ( long index, ASIOBool )
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{
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// convert current sample in 24 bit format
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int32_t iCurSam = static_cast<int32_t> (
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vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] );
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vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
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iCurSam <<= 8;
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memcpy ( ( (char*) pSound->bufferInfos[i].buffers[index] ) + iCurSample * 3, &iCurSam, 3 );
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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;
|
||||
|
|
|
|||
Loading…
Reference in a new issue