diff --git a/windows/sound.cpp b/windows/sound.cpp index 8759426d..6d444c5c 100755 --- a/windows/sound.cpp +++ b/windows/sound.cpp @@ -499,6 +499,10 @@ void CSound::bufferSwitch ( long index, ASIOBool ) { int iCurSample; + // get references to class members + int& iASIOBufferSizeMono = pSound->iASIOBufferSizeMono; + CVector& 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 ( pSound->bufferInfos[i].buffers[index] ); + + for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ ) { - pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] = - static_cast ( 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 ( iCurSam ); } break; case ASIOSTInt32LSB: -// NOT YET TESTED - for ( iCurSample = 0; iCurSample < pSound->iASIOBufferSizeMono; iCurSample++ ) + { + int32_t* pASIOBuf = + static_cast ( pSound->bufferInfos[i].buffers[index] ); + + for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ ) { - pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] = - static_cast ( static_cast ( - pSound->bufferInfos[i].buffers[index] )[iCurSample] >> 16 ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] = + static_cast ( 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 ( static_cast ( 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 ( static_cast ( 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 ( static_cast ( 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 ( ( static_cast ( 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 ( ( static_cast ( 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 ( ( static_cast ( 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 ( 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 ( 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 ( Flip32Bits ( static_cast ( 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 ( static_cast ( Flip32Bits ( static_cast ( 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 ( static_cast ( Flip64Bits ( static_cast ( 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 ( Flip32Bits ( static_cast ( 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 ( ( Flip32Bits ( static_cast ( 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 ( ( Flip32Bits ( static_cast ( 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 ( ( Flip32Bits ( static_cast ( 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 ( pSound->bufferInfos[i].buffers[index] ); + + for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ ) { - static_cast ( 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 ( - 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 ( pSound->bufferInfos[i].buffers[index] ); + + for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ ) { // convert to 32 bit const int32_t iCurSam = static_cast ( - pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); - static_cast ( 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 ( - pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); static_cast ( 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 ( - pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); static_cast ( 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 ( - pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); static_cast ( 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 ( - pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); static_cast ( 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 ( - pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); static_cast ( 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 ( - pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); static_cast ( 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 ( 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 ( - pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); static_cast ( 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 ( - pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); static_cast ( pSound->bufferInfos[i].buffers[index] )[iCurSample] = static_cast ( Flip32Bits ( static_cast ( @@ -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 ( - pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); static_cast ( pSound->bufferInfos[i].buffers[index] )[iCurSample] = static_cast ( Flip64Bits ( static_cast ( @@ -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 ( - pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); static_cast ( 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 ( - pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); static_cast ( 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 ( - pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); static_cast ( 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 ( - pSound->vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); static_cast ( 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 ) {