diff --git a/src/client.cpp b/src/client.cpp
index 86e26cf8..515bc012 100755
--- a/src/client.cpp
+++ b/src/client.cpp
@@ -835,6 +835,7 @@ void CClient::Init()
     vecCeltData.Init ( iCeltNumCodedBytes );
     vecZeros.Init ( iStereoBlockSizeSam, 0 );
     vecsStereoSndCrdTMP.Init ( iStereoBlockSizeSam );
+    vecsStereoSndCrdMuteStream.Init ( iStereoBlockSizeSam );
 
     opus_custom_encoder_ctl ( CurOpusEncoder,
                               OPUS_SET_BITRATE (
@@ -883,22 +884,38 @@ void CClient::AudioCallback ( CVector<int16_t>& psData, void* arg )
 
     // process audio data
     pMyClientObj->ProcessSndCrdAudioData ( psData );
-}
-
-void CClient::ProcessSndCrdAudioData ( CVector<int16_t>& vecsStereoSndCrd )
-{
 
 /*
 // TEST do a soundcard jitter measurement
 static CTimingMeas JitterMeas ( 1000, "test2.dat" );
 JitterMeas.Measure();
 */
+}
+
+void CClient::ProcessSndCrdAudioData ( CVector<int16_t>& vecsMultChanAudioSndCrd )
+{
+
+// TODO output mapping from stereo to multi channel: We want to change all the different sound interfaces that they
+// do not select the input and output channels but we do it here at the client. This has the advantage that, e.g.,
+// the special add modes supported for Windows (i.e. if 4 input channels available, you can mix channel 1+3 or 1+4)
+// can then be used for Mac as well without the need of changing anything in the actual Mac sound interface.
+// Since a multichannel signal arrives and must be converted to a stereo signal, we need an additional buffer: vecsStereoSndCrdTMP.
+// TEST input channel selection/mixing
+//const int iNumInCh = 2;
+//for ( int i = 0; i < iNumInCh; i++ )
+//{
+//    for ( int j = 0; j < iMonoBlockSizeSam; j++ )
+//    {
+//        vecsStereoSndCrdTMP[2 * j + i] = vecsMultChanAudioSndCrd[iNumInCh * j + i];
+//    }
+//}
+vecsStereoSndCrdTMP = vecsMultChanAudioSndCrd; // TEST just copy the stereo data for now
 
     // check if a conversion buffer is required or not
     if ( bSndCrdConversionBufferRequired )
     {
         // add new sound card block in conversion buffer
-        SndCrdConversionBufferIn.Put ( vecsStereoSndCrd, vecsStereoSndCrd.Size() );
+        SndCrdConversionBufferIn.Put ( vecsStereoSndCrdTMP, vecsStereoSndCrdTMP.Size() );
 
         // process all available blocks of data
         while ( SndCrdConversionBufferIn.GetAvailData() >= iStereoBlockSizeSam )
@@ -913,14 +930,17 @@ JitterMeas.Measure();
         }
 
         // get processed sound card block out of the conversion buffer
-        SndCrdConversionBufferOut.Get ( vecsStereoSndCrd, vecsStereoSndCrd.Size() );
+        SndCrdConversionBufferOut.Get ( vecsStereoSndCrdTMP, vecsStereoSndCrdTMP.Size() );
     }
     else
     {
         // regular case: no conversion buffer required
         // process audio data
-        ProcessAudioDataIntern ( vecsStereoSndCrd );
+        ProcessAudioDataIntern ( vecsStereoSndCrdTMP );
     }
+
+// TODO output mapping from stereo to multi channel, see comment above for the input mapping
+vecsMultChanAudioSndCrd = vecsStereoSndCrdTMP; // TEST just copy the stereo data for now
 }
 
 void CClient::ProcessAudioDataIntern ( CVector<int16_t>& vecsStereoSndCrd )
@@ -1101,7 +1121,7 @@ void CClient::ProcessAudioDataIntern ( CVector<int16_t>& vecsStereoSndCrd )
     // in case of mute stream, store local data
     if ( bMuteOutStream )
     {
-        vecsStereoSndCrdTMP = vecsStereoSndCrd;
+        vecsStereoSndCrdMuteStream = vecsStereoSndCrd;
     }
 
     for ( i = 0; i < iSndCrdFrameSizeFactor; i++ )
@@ -1158,7 +1178,7 @@ fflush(pFileDelay);
         for ( i = 0; i < iStereoBlockSizeSam; i++ )
         {
             vecsStereoSndCrd[i] = Double2Short (
-                static_cast<double> ( vecsStereoSndCrd[i] ) + vecsStereoSndCrdTMP[i] );
+                static_cast<double> ( vecsStereoSndCrd[i] ) + vecsStereoSndCrdMuteStream[i] );
         }
     }
 
diff --git a/src/client.h b/src/client.h
index bcea8094..15837f00 100755
--- a/src/client.h
+++ b/src/client.h
@@ -304,7 +304,7 @@ protected:
     static void AudioCallback ( CVector<short>& psData, void* arg );
 
     void        Init();
-    void        ProcessSndCrdAudioData ( CVector<short>& vecsStereoSndCrd );
+    void        ProcessSndCrdAudioData ( CVector<short>& vecsMultChanAudioSndCrd );
     void        ProcessAudioDataIntern ( CVector<short>& vecsStereoSndCrd );
 
     int         PreparePingMessage();
@@ -359,6 +359,7 @@ protected:
     CBufferBase<int16_t>    SndCrdConversionBufferOut;
     CVector<int16_t>        vecDataConvBuf;
     CVector<int16_t>        vecsStereoSndCrdTMP;
+    CVector<int16_t>        vecsStereoSndCrdMuteStream;
     CVector<int16_t>        vecZeros;
 
     bool                    bFraSiFactPrefSupported;
diff --git a/src/util.cpp b/src/util.cpp
index f1a23873..225bc594 100755
--- a/src/util.cpp
+++ b/src/util.cpp
@@ -69,7 +69,7 @@ double CStereoSignalLevelMeter::UpdateCurLevel ( double       dCurLevel,
 // TODO Calculate factor from sample rate and frame size (64 or 128 samples frame size).
 //      But tests with 128 and 64 samples frame size have shown that the meter fly back
 //      is ok for both numbers of samples frame size.
-        dCurLevel *= 0.95;
+        dCurLevel *= 0.97;
     }
     else
     {
diff --git a/windows/sound.cpp b/windows/sound.cpp
index d33c0f19..65e9006f 100755
--- a/windows/sound.cpp
+++ b/windows/sound.cpp
@@ -404,7 +404,7 @@ int CSound::Init ( const int iNewPrefMonoBufferSize )
         ASIOSetSampleRate ( SYSTEM_SAMPLE_RATE_HZ );
 
         // create memory for intermediate audio buffer
-        vecsTmpAudioSndCrdStereo.Init ( iASIOBufferSizeStereo );
+        vecsMultChanAudioSndCrd.Init ( iASIOBufferSizeStereo );
 
         // create and activate ASIO buffers (buffer size in samples),
         // dispose old buffers (if any)
@@ -600,8 +600,8 @@ void CSound::bufferSwitch ( long index, ASIOBool )
     int iCurSample;
 
     // get references to class members
-    int&              iASIOBufferSizeMono      = pSound->iASIOBufferSizeMono;
-    CVector<int16_t>& vecsTmpAudioSndCrdStereo = pSound->vecsTmpAudioSndCrdStereo;
+    int&              iASIOBufferSizeMono     = pSound->iASIOBufferSizeMono;
+    CVector<int16_t>& vecsMultChanAudioSndCrd = pSound->vecsMultChanAudioSndCrd;
 
     // perform the processing for input and output
     pSound->ASIOMutex.lock(); // get mutex lock
@@ -625,7 +625,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
 
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
-                    vecsTmpAudioSndCrdStereo[2 * iCurSample + i] = pASIOBuf[iCurSample];
+                    vecsMultChanAudioSndCrd[2 * iCurSample + i] = pASIOBuf[iCurSample];
                 }
 
                 if ( iSelAddCH >= 0 )
@@ -635,8 +635,8 @@ void CSound::bufferSwitch ( long index, ASIOBool )
 
                     for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                     {
-                        vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
-                            Double2Short ( (double) vecsTmpAudioSndCrdStereo[2 * iCurSample + i] +
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] =
+                            Double2Short ( (double) vecsMultChanAudioSndCrd[2 * iCurSample + i] +
                                            (double) pASIOBufAdd[iCurSample] );
                     }
                 }
@@ -650,7 +650,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                     memcpy ( &iCurSam, ( (char*) pSound->bufferInfos[iSelCH].buffers[index] ) + iCurSample * 3, 3 );
                     iCurSam >>= 8;
 
-                    vecsTmpAudioSndCrdStereo[2 * iCurSample + i] = static_cast<int16_t> ( iCurSam );
+                    vecsMultChanAudioSndCrd[2 * iCurSample + i] = static_cast<int16_t> ( iCurSam );
                 }
 
                 if ( iSelAddCH >= 0 )
@@ -662,8 +662,8 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                         memcpy ( &iCurSam, ( (char*) pSound->bufferInfos[iSelAddCH].buffers[index] ) + iCurSample * 3, 3 );
                         iCurSam >>= 8;
 
-                        vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
-                            Double2Short ( (double) vecsTmpAudioSndCrdStereo[2 * iCurSample + i] +
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] =
+                            Double2Short ( (double) vecsMultChanAudioSndCrd[2 * iCurSample + i] +
                                            (double) static_cast<int16_t> ( iCurSam ) );
                     }
                 }
@@ -675,7 +675,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
 
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
-                    vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
+                    vecsMultChanAudioSndCrd[2 * iCurSample + i] =
                         static_cast<int16_t> ( pASIOBuf[iCurSample] >> 16 );
                 }
 
@@ -686,8 +686,8 @@ void CSound::bufferSwitch ( long index, ASIOBool )
 
                     for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                     {
-                        vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
-                            Double2Short ( (double) vecsTmpAudioSndCrdStereo[2 * iCurSample + i] +
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] =
+                            Double2Short ( (double) vecsMultChanAudioSndCrd[2 * iCurSample + i] +
                                            (double) static_cast<int16_t> ( pASIOBufAdd[iCurSample] >> 16 ) );
                     }
                 }
@@ -698,7 +698,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
 // NOT YET TESTED
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
-                    vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
+                    vecsMultChanAudioSndCrd[2 * iCurSample + i] =
                         static_cast<int16_t> ( static_cast<float*> (
                         pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] * _MAXSHORT );
                 }
@@ -708,7 +708,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
 // NOT YET TESTED
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
-                    vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
+                    vecsMultChanAudioSndCrd[2 * iCurSample + i] =
                         static_cast<int16_t> ( static_cast<double*> (
                         pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] * _MAXSHORT );
                 }
@@ -718,7 +718,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
 // NOT YET TESTED
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
-                    vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
+                    vecsMultChanAudioSndCrd[2 * iCurSample + i] =
                         static_cast<int16_t> ( static_cast<int32_t*> (
                         pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] & 0xFFFF );
                 }
@@ -728,7 +728,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
 // NOT YET TESTED
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
-                    vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
+                    vecsMultChanAudioSndCrd[2 * iCurSample + i] =
                         static_cast<int16_t> ( ( static_cast<int32_t*> (
                         pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] & 0x3FFFF ) >> 2 );
                 }
@@ -738,7 +738,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
 // NOT YET TESTED
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
-                    vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
+                    vecsMultChanAudioSndCrd[2 * iCurSample + i] =
                         static_cast<int16_t> ( ( static_cast<int32_t*> (
                         pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] & 0xFFFFF ) >> 4 );
                 }
@@ -748,7 +748,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
 // NOT YET TESTED
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
-                    vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
+                    vecsMultChanAudioSndCrd[2 * iCurSample + i] =
                         static_cast<int16_t> ( ( static_cast<int32_t*> (
                         pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] & 0xFFFFFF ) >> 8 );
                 }
@@ -759,7 +759,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                 // flip bits
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
-                    vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
+                    vecsMultChanAudioSndCrd[2 * iCurSample + i] =
                         Flip16Bits ( ( static_cast<int16_t*> (
                         pSound->bufferInfos[iSelCH].buffers[index] ) )[iCurSample] );
                 }
@@ -774,7 +774,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                     int iCurSam = 0;
                     memcpy ( &iCurSam, ( (char*) pSound->bufferInfos[iSelCH].buffers[index] ) + iCurSample * 3, 3 );
 
-                    vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
+                    vecsMultChanAudioSndCrd[2 * iCurSample + i] =
                         Flip16Bits ( static_cast<int16_t> ( iCurSam ) );
                 }
                 break;
@@ -784,7 +784,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
                     // flip bits and convert to 16 bit
-                    vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
+                    vecsMultChanAudioSndCrd[2 * iCurSample + i] =
                         static_cast<int16_t> ( Flip32Bits ( static_cast<int32_t*> (
                         pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) >> 16 );
                 }
@@ -794,7 +794,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
 // NOT YET TESTED
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
-                    vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
+                    vecsMultChanAudioSndCrd[2 * iCurSample + i] =
                         static_cast<int16_t> ( static_cast<float> (
                         Flip32Bits ( static_cast<int32_t*> (
                         pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
@@ -805,7 +805,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
 // NOT YET TESTED
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
-                    vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
+                    vecsMultChanAudioSndCrd[2 * iCurSample + i] =
                         static_cast<int16_t> ( static_cast<double> (
                         Flip64Bits ( static_cast<int64_t*> (
                         pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) ) * _MAXSHORT );
@@ -816,7 +816,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
 // NOT YET TESTED
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
-                    vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
+                    vecsMultChanAudioSndCrd[2 * iCurSample + i] =
                         static_cast<int16_t> ( Flip32Bits ( static_cast<int32_t*> (
                         pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) & 0xFFFF );
                 }
@@ -826,7 +826,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
 // NOT YET TESTED
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
-                    vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
+                    vecsMultChanAudioSndCrd[2 * iCurSample + i] =
                         static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
                         pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) & 0x3FFFF ) >> 2 );
                 }
@@ -836,7 +836,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
 // NOT YET TESTED
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
-                    vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
+                    vecsMultChanAudioSndCrd[2 * iCurSample + i] =
                         static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
                         pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) & 0xFFFFF ) >> 4 );
                 }
@@ -846,7 +846,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
 // NOT YET TESTED
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
-                    vecsTmpAudioSndCrdStereo[2 * iCurSample + i] =
+                    vecsMultChanAudioSndCrd[2 * iCurSample + i] =
                         static_cast<int16_t> ( ( Flip32Bits ( static_cast<int32_t*> (
                         pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] ) & 0xFFFFFF ) >> 8 );
                 }
@@ -855,7 +855,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
         }
 
         // call processing callback function
-        pSound->ProcessCallback ( vecsTmpAudioSndCrdStereo );
+        pSound->ProcessCallback ( vecsMultChanAudioSndCrd );
 
 
         // PLAYBACK ------------------------------------------------------------
@@ -874,7 +874,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
 
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
-                    pASIOBuf[iCurSample] = vecsTmpAudioSndCrdStereo[2 * iCurSample + i];
+                    pASIOBuf[iCurSample] = vecsMultChanAudioSndCrd[2 * iCurSample + i];
                 }
                 break;
             }
@@ -885,7 +885,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                 {
                     // convert current sample in 24 bit format
                     int32_t iCurSam = static_cast<int32_t> (
-                        vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] );
 
                     iCurSam <<= 8;
 
@@ -901,7 +901,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                 {
                     // convert to 32 bit
                     const int32_t iCurSam = static_cast<int32_t> (
-                        vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] );
 
                     pASIOBuf[iCurSample] = ( iCurSam << 16 );
                 }
@@ -913,7 +913,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
                     const float fCurSam = static_cast<float> (
-                        vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] );
 
                     static_cast<float*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
                         fCurSam / _MAXSHORT;
@@ -925,7 +925,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
                     const double fCurSam = static_cast<double> (
-                        vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] );
 
                     static_cast<double*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
                         fCurSam / _MAXSHORT;
@@ -938,7 +938,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                 {
                     // convert to 32 bit
                     const int32_t iCurSam = static_cast<int32_t> (
-                        vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] );
 
                     static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
                         iCurSam;
@@ -951,7 +951,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                 {
                     // convert to 32 bit
                     const int32_t iCurSam = static_cast<int32_t> (
-                        vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] );
 
                     static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
                         ( iCurSam << 2 );
@@ -964,7 +964,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                 {
                     // convert to 32 bit
                     const int32_t iCurSam = static_cast<int32_t> (
-                        vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] );
 
                     static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
                         ( iCurSam << 4 );
@@ -977,7 +977,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                 {
                     // convert to 32 bit
                     const int32_t iCurSam = static_cast<int32_t> (
-                        vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] );
 
                     static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
                         ( iCurSam << 8 );
@@ -990,7 +990,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
                     ( (int16_t*) pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
-                        Flip16Bits ( vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
+                        Flip16Bits ( vecsMultChanAudioSndCrd[2 * iCurSample + i] );
                 }
                 break;
 
@@ -1001,7 +1001,7 @@ 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 + i] ) );
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] ) );
 
                     memcpy ( ( (char*) pSound->bufferInfos[iSelCH].buffers[index] ) + iCurSample * 3, &iCurSam, 3 );
                 }
@@ -1013,7 +1013,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                 {
                     // convert to 32 bit and flip bits
                     int iCurSam = static_cast<int32_t> (
-                        vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] );
 
                     static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
                         Flip32Bits ( iCurSam << 16 );
@@ -1025,7 +1025,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
                     const float fCurSam = static_cast<float> (
-                        vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] );
 
                     static_cast<float*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
                         static_cast<float> ( Flip32Bits ( static_cast<int32_t> (
@@ -1038,7 +1038,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                 for ( iCurSample = 0; iCurSample < iASIOBufferSizeMono; iCurSample++ )
                 {
                     const double fCurSam = static_cast<double> (
-                        vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] );
 
                     static_cast<float*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
                         static_cast<double> ( Flip64Bits ( static_cast<int64_t> (
@@ -1052,7 +1052,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                 {
                     // convert to 32 bit
                     const int32_t iCurSam = static_cast<int32_t> (
-                        vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] );
 
                     static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
                         Flip32Bits ( iCurSam );
@@ -1065,7 +1065,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                 {
                     // convert to 32 bit
                     const int32_t iCurSam = static_cast<int32_t> (
-                        vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] );
 
                     static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
                         Flip32Bits ( iCurSam << 2 );
@@ -1078,7 +1078,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                 {
                     // convert to 32 bit
                     const int32_t iCurSam = static_cast<int32_t> (
-                        vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] );
 
                     static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
                         Flip32Bits ( iCurSam << 4 );
@@ -1091,7 +1091,7 @@ void CSound::bufferSwitch ( long index, ASIOBool )
                 {
                     // convert to 32 bit
                     const int32_t iCurSam = static_cast<int32_t> (
-                        vecsTmpAudioSndCrdStereo[2 * iCurSample + i] );
+                        vecsMultChanAudioSndCrd[2 * iCurSample + i] );
 
                     static_cast<int32_t*> ( pSound->bufferInfos[iSelCH].buffers[index] )[iCurSample] =
                         Flip32Bits ( iCurSam << 8 );
diff --git a/windows/sound.h b/windows/sound.h
index 21881bb9..5f6d96a7 100755
--- a/windows/sound.h
+++ b/windows/sound.h
@@ -117,7 +117,7 @@ protected:
     CVector<int>     vSelectedInputChannels;
     CVector<int>     vSelectedOutputChannels;
 
-    CVector<int16_t> vecsTmpAudioSndCrdStereo;
+    CVector<int16_t> vecsMultChanAudioSndCrd;
 
     QMutex           ASIOMutex;