From 7f41f186d46a095ba57abf79bca7fb3b57582c6b Mon Sep 17 00:00:00 2001 From: Volker Fischer Date: Sun, 25 Mar 2018 11:21:58 +0000 Subject: [PATCH] prepare for special summing channels --- windows/sound.cpp | 196 ++++++++++++++++++++++------------------------ 1 file changed, 95 insertions(+), 101 deletions(-) diff --git a/windows/sound.cpp b/windows/sound.cpp index 21819dca..77e41971 100755 --- a/windows/sound.cpp +++ b/windows/sound.cpp @@ -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 ); @@ -342,26 +343,26 @@ 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++ ) { - // prepare input channels 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 */, - iASIOBufferSizeMono, &asioCallbacks ); + // 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 long lInputLatency = 0; @@ -520,7 +521,7 @@ void CSound::bufferSwitch ( long index, ASIOBool ) int iCurSample; // get references to class members - int& iASIOBufferSizeMono = pSound->iASIOBufferSizeMono; + int& iASIOBufferSizeMono = pSound->iASIOBufferSizeMono; CVector& vecsTmpAudioSndCrdStereo = pSound->vecsTmpAudioSndCrdStereo; // perform the processing for input and output @@ -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 ( pSound->bufferInfos[i].buffers[index] ); + int16_t* pASIOBuf = static_cast ( 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 ( iCurSam ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + i] = static_cast ( iCurSam ); } break; case ASIOSTInt32LSB: { - int32_t* pASIOBuf = - static_cast ( pSound->bufferInfos[i].buffers[index] ); + int32_t* pASIOBuf = static_cast ( pSound->bufferInfos[iSelCH].buffers[index] ); for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ ) { - vecsTmpAudioSndCrdStereo[2 * iCurSample + iInChNum] = + vecsTmpAudioSndCrdStereo[2 * iCurSample + i] = static_cast ( 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 ( static_cast ( - 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 ( static_cast ( - 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 ( static_cast ( - 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 ( ( static_cast ( - 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 ( ( static_cast ( - 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 ( ( static_cast ( - 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 ( - 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 ( 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 ( Flip32Bits ( static_cast ( - 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 ( static_cast ( Flip32Bits ( static_cast ( - 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 ( static_cast ( Flip64Bits ( static_cast ( - 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 ( Flip32Bits ( static_cast ( - 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 ( ( Flip32Bits ( static_cast ( - 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 ( ( Flip32Bits ( static_cast ( - 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 ( ( Flip32Bits ( static_cast ( - 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 ( pSound->bufferInfos[i].buffers[index] ); + int16_t* pASIOBuf = static_cast ( 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 ( - 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 ( pSound->bufferInfos[i].buffers[index] ); + int32_t* pASIOBuf = static_cast ( pSound->bufferInfos[iSelCH].buffers[index] ); for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ ) { // convert to 32 bit const int32_t iCurSam = static_cast ( - 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 ( - vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + i] ); - static_cast ( pSound->bufferInfos[i].buffers[index] )[iCurSample] = + static_cast ( 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 ( - vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + i] ); - static_cast ( pSound->bufferInfos[i].buffers[index] )[iCurSample] = + static_cast ( 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 ( - vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + i] ); - static_cast ( pSound->bufferInfos[i].buffers[index] )[iCurSample] = + static_cast ( 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 ( - vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + i] ); - static_cast ( pSound->bufferInfos[i].buffers[index] )[iCurSample] = + static_cast ( 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 ( - vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + i] ); - static_cast ( pSound->bufferInfos[i].buffers[index] )[iCurSample] = + static_cast ( 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 ( - vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + i] ); - static_cast ( pSound->bufferInfos[i].buffers[index] )[iCurSample] = + static_cast ( 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 ( 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 ( - vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + i] ); - static_cast ( pSound->bufferInfos[i].buffers[index] )[iCurSample] = + static_cast ( 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 ( - vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + i] ); - static_cast ( pSound->bufferInfos[i].buffers[index] )[iCurSample] = + static_cast ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] = static_cast ( Flip32Bits ( static_cast ( fCurSam / _MAXSHORT ) ) ); } @@ -921,9 +915,9 @@ void CSound::bufferSwitch ( long index, ASIOBool ) for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ ) { const double fCurSam = static_cast ( - vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + i] ); - static_cast ( pSound->bufferInfos[i].buffers[index] )[iCurSample] = + static_cast ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] = static_cast ( Flip64Bits ( static_cast ( fCurSam / _MAXSHORT ) ) ); } @@ -935,9 +929,9 @@ void CSound::bufferSwitch ( long index, ASIOBool ) { // convert to 32 bit const int32_t iCurSam = static_cast ( - vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + i] ); - static_cast ( pSound->bufferInfos[i].buffers[index] )[iCurSample] = + static_cast ( 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 ( - vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + i] ); - static_cast ( pSound->bufferInfos[i].buffers[index] )[iCurSample] = + static_cast ( 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 ( - vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + i] ); - static_cast ( pSound->bufferInfos[i].buffers[index] )[iCurSample] = + static_cast ( 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 ( - vecsTmpAudioSndCrdStereo[2 * iCurSample + iOutChNum] ); + vecsTmpAudioSndCrdStereo[2 * iCurSample + i] ); - static_cast ( pSound->bufferInfos[i].buffers[index] )[iCurSample] = + static_cast ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] = Flip32Bits ( iCurSam << 8 ); } break;