Updates to OS-X for multi-stream support (GS).

1 files changed, 5290 insertions, 5189 deletions

diff --git a/RtAudio.cpp b/RtAudio.cpp
index 05e3afa..9b186cc 100644
--- a/RtAudio.cpp
+++ b/RtAudio.cpp
@@ -4,7 +4,7 @@
 
     RtAudio provides a common API (Application Programming Interface)
     for realtime audio input/output across Linux (native ALSA, Jack,
-    and OSS), SGI, Macintosh OS X (CoreAudio and Jack), and Windows
+    and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
     (DirectSound and ASIO) operating systems.
 
     RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
@@ -401,7 +401,8 @@ unsigned int RtApi :: getStreamSampleRate( void )
 struct CoreHandle {
   AudioDeviceID id[2];    // device ids
   AudioDeviceIOProcID procId[2];
-  UInt32 iStream[2];      // device stream index (first for mono mode)
+  UInt32 iStream[2];      // device stream index (or first if using multiple)
+  UInt32 nStreams[2];     // number of streams to use
   bool xrun[2];
   char *deviceBuffer;
   pthread_cond_t condition;
@@ -409,7 +410,7 @@ struct CoreHandle {
   bool internalDrain;     // Indicates if stop is initiated from callback or not.
 
   CoreHandle()
-    :deviceBuffer(0), drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
+    :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
 };
 
 RtApiCore :: RtApiCore()
@@ -813,69 +814,73 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
     return FAILURE;
   }
 
-  // Search for a stream that contains the desired number of
+  // Search for one or more streams that contain the desired number of
   // channels. CoreAudio devices can have an arbitrary number of
   // streams and each stream can have an arbitrary number of channels.
   // For each stream, a single buffer of interleaved samples is
-  // provided.  RtAudio currently only supports the use of one stream
-  // of interleaved data or multiple consecutive single-channel
-  // streams.  Thus, our search below is limited to these two
-  // contexts.
-  unsigned int streamChannels = 0, nStreams = 0;
-  UInt32 iChannel = 0, iStream = 0;
-  unsigned int offsetCounter = firstChannel;
-  stream_.deviceInterleaved[mode] = true;
-  nStreams = bufferList->mNumberBuffers;
+  // provided.  RtAudio prefers the use of one stream of interleaved
+  // data or multiple consecutive single-channel streams.  However, we
+  // now support multiple consecutive multi-channel streams of
+  // interleaved data as well.
+  UInt32 iStream, offsetCounter = firstChannel;
+  UInt32 nStreams = bufferList->mNumberBuffers;
+  bool monoMode = false;
   bool foundStream = false;
 
+  // First check that the device supports the requested number of
+  // channels.
+  UInt32 deviceChannels = 0;
+  for ( iStream=0; iStream<nStreams; iStream++ )
+    deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;
+
+  if ( deviceChannels < ( channels + firstChannel ) ) {
+    free( bufferList );
+    errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  // Look for a single stream meeting our needs.
+  UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;
   for ( iStream=0; iStream<nStreams; iStream++ ) {
     streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
     if ( streamChannels >= channels + offsetCounter ) {
-      iChannel += offsetCounter;
+      firstStream = iStream;
+      channelOffset = offsetCounter;
       foundStream = true;
       break;
     }
     if ( streamChannels > offsetCounter ) break;
     offsetCounter -= streamChannels;
-    iChannel += streamChannels;
   }
 
-  // If we didn't find a single stream above, see if we can meet
-  // the channel specification in mono mode (i.e. using separate
-  // non-interleaved buffers).  This can only work if there are N
-  // consecutive one-channel streams, where N is the number of
-  // desired channels (+ channel offset).
+  // If we didn't find a single stream above, then we should be able
+  // to meet the channel specification with multiple streams.
   if ( foundStream == false ) {
-    unsigned int counter = 0;
+    monoMode = true;
     offsetCounter = firstChannel;
-    iChannel = 0;
     for ( iStream=0; iStream<nStreams; iStream++ ) {
       streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
-      if ( offsetCounter ) {
-        if ( streamChannels > offsetCounter ) break;
-        offsetCounter -= streamChannels;
-      }
-      else if ( streamChannels == 1 )
-        counter++;
-      else
-        counter = 0;
-      if ( counter == channels ) {
-        iStream -= channels - 1;
-        iChannel -= channels - 1;
-        stream_.deviceInterleaved[mode] = false;
-        foundStream = true;
-        break;
-      }
-      iChannel += streamChannels;
+      if ( streamChannels > offsetCounter ) break;
+      offsetCounter -= streamChannels;
+    }
+
+    firstStream = iStream;
+    channelOffset = offsetCounter;
+    Int32 channelCounter = channels + offsetCounter - streamChannels;
+
+    if ( streamChannels > 1 ) monoMode = false;
+    while ( channelCounter > 0 ) {
+      streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;
+      if ( streamChannels > 1 ) monoMode = false;
+      channelCounter -= streamChannels;
+      streamCount++;
     }
   }
+
   free( bufferList );
 
-  if ( foundStream == false ) {
-    errorStream_ << "RtApiCore::probeDeviceOpen: unable to find OS-X stream on device (" << device << ") for requested channels.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+  std::cout << "deviceStreams = " << nStreams << ", firstStream = " << firstStream << ", streamCount = " << streamCount << ", channelOffset = " << channelOffset << std::endl;
 
   // Determine the buffer size.
   AudioValueRange	bufferRange;
@@ -893,8 +898,8 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
   else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;
   if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;
 
-  // Set the buffer size.  For mono mode, I'm assuming we only need to
-  // make this setting for the master channel.
+  // Set the buffer size.  For multiple streams, I'm assuming we only
+  // need to make this setting for the master channel.
   UInt32 theSize = (UInt32) *bufferSize;
   dataSize = sizeof( UInt32 );
   result = AudioDeviceSetProperty( id, NULL, 0, isInput,
@@ -919,8 +924,8 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
   stream_.bufferSize = *bufferSize;
   stream_.nBuffers = 1;
 
-  // Get the stream ID(s) so we can set the stream format.  In mono
-  // mode, we'll have to do this for each stream (channel).
+  // Get the stream ID(s) so we can set the stream format.  We'll have
+  // to do this for each stream.
   AudioStreamID streamIDs[ nStreams ];
   dataSize = nStreams * sizeof( AudioStreamID );
   result = AudioDeviceGetProperty( id, 0, isInput,
@@ -936,13 +941,11 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
   // device and change that if necessary.
   AudioStreamBasicDescription	description;
   dataSize = sizeof( AudioStreamBasicDescription );
-  if ( stream_.deviceInterleaved[mode] ) nStreams = 1;
-  else nStreams = channels;
 
   bool updateFormat;
-  for ( unsigned int i=0; i<nStreams; i++ ) {
+  for ( UInt32 i=0; i<streamCount; i++ ) {
 
-    result = AudioStreamGetProperty( streamIDs[iStream+i], 0,
+    result = AudioStreamGetProperty( streamIDs[firstStream+i], 0,
                                      kAudioStreamPropertyVirtualFormat,
                                      &dataSize, &description );
 
@@ -967,7 +970,7 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
     }
 
     if ( updateFormat ) {
-      result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0,
+      result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0,
                                        kAudioStreamPropertyVirtualFormat,
                                        dataSize, &description );
       if ( result != noErr ) {
@@ -978,7 +981,7 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
     }
 
     // Now check the physical format.
-    result = AudioStreamGetProperty( streamIDs[iStream+i], 0,
+    result = AudioStreamGetProperty( streamIDs[firstStream+i], 0,
                                      kAudioStreamPropertyPhysicalFormat,
                                      &dataSize, &description );
     if ( result != noErr ) {
@@ -996,32 +999,32 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
       testDescription.mBitsPerChannel = 32;
       formatFlags = description.mFormatFlags | kLinearPCMFormatFlagIsFloat & ~kLinearPCMFormatFlagIsSignedInteger;
       testDescription.mFormatFlags = formatFlags;
-      result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
+      result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
       if ( result == noErr ) continue;
 
       testDescription = description;
       testDescription.mBitsPerChannel = 32;
       formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger) & ~kLinearPCMFormatFlagIsFloat;
       testDescription.mFormatFlags = formatFlags;
-      result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
+      result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
       if ( result == noErr ) continue;
 
       testDescription = description;
       testDescription.mBitsPerChannel = 24;
       testDescription.mFormatFlags = formatFlags;
-      result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
+      result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
       if ( result == noErr ) continue;
 
       testDescription = description;
       testDescription.mBitsPerChannel = 16;
       testDescription.mFormatFlags = formatFlags;
-      result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
+      result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
       if ( result == noErr ) continue;
 
       testDescription = description;
       testDescription.mBitsPerChannel = 8;
       testDescription.mFormatFlags = formatFlags;
-      result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
+      result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
       if ( result != noErr ) {
         errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
         errorText_ = errorStream_.str();
@@ -1034,14 +1037,12 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
   // and the stream.  First, attempt to get the device latency on the
   // master channel or the first open channel.  Errors that might
   // occur here are not deemed critical.
+
+  // ***** CHECK THIS ***** //
   UInt32 latency, channel = 0;
   dataSize = sizeof( UInt32 );
   AudioDevicePropertyID property = kAudioDevicePropertyLatency;
-  for ( int i=0; i<2; i++ ) {
-    if ( hasProperty( id, channel, isInput, property ) == true ) break;
-    channel = iChannel + 1 + i;
-  }
-  if ( channel <= iChannel + 1 ) {
+  if ( hasProperty( id, channel, isInput, property ) == true ) {
     result = AudioDeviceGetProperty( id, channel, isInput, property, &dataSize, &latency );
     if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
     else {
@@ -1051,9 +1052,9 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
     }
   }
 
-  // Now try to get the stream latency.  For "mono" mode, I assume the
-  // latency is equal for all single-channel streams.
-  result = AudioStreamGetProperty( streamIDs[iStream], 0, property, &dataSize, &latency );
+  // Now try to get the stream latency.  For multiple streams, I assume the
+  // latency is equal for each.
+  result = AudioStreamGetProperty( streamIDs[firstStream], 0, property, &dataSize, &latency );
   if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] += latency;
   else {
     errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream latency for device (" << device << ").";
@@ -1071,14 +1072,16 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
   stream_.userFormat = format;
   stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
 
-  if ( stream_.deviceInterleaved[mode] )
+  if ( streamCount == 1 )
     stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
-  else // mono mode
+  else // multiple streams
     stream_.nDeviceChannels[mode] = channels;
   stream_.nUserChannels[mode] = channels;
-  stream_.channelOffset[mode] = iChannel;  // offset within a CoreAudio stream
+  stream_.channelOffset[mode] = channelOffset;  // offset within a CoreAudio stream
   if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
   else stream_.userInterleaved = true;
+  stream_.deviceInterleaved[mode] = true;
+  if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;
 
   // Set flags for buffer conversion.
   stream_.doConvertBuffer[mode] = false;
@@ -1086,10 +1089,16 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
     stream_.doConvertBuffer[mode] = true;
   if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
     stream_.doConvertBuffer[mode] = true;
-  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
-       stream_.nUserChannels[mode] > 1 )
+  if ( streamCount == 1 ) {
+    if ( stream_.nUserChannels[mode] > 1 &&
+         stream_.userInterleaved != stream_.deviceInterleaved[mode] )
+      stream_.doConvertBuffer[mode] = true;
+  }
+  else if ( monoMode && stream_.userInterleaved )
     stream_.doConvertBuffer[mode] = true;
 
+  std::cout << "doConvert = " << stream_.doConvertBuffer[mode] << ", userInterleaved = " << stream_.userInterleaved << ", deviceInterleaved = " << stream_.deviceInterleaved[mode] << std::endl;
+
   // Allocate our CoreHandle structure for the stream.
   CoreHandle *handle = 0;
   if ( stream_.apiHandle == 0 ) {
@@ -1109,7 +1118,8 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
   }
   else
     handle = (CoreHandle *) stream_.apiHandle;
-  handle->iStream[mode] = iStream;
+  handle->iStream[mode] = firstStream;
+  handle->nStreams[mode] = streamCount;
   handle->id[mode] = id;
 
   // Allocate necessary internal buffers.
@@ -1122,9 +1132,9 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
   }
 
   // If possible, we will make use of the CoreAudio stream buffers as
-  // "device buffers".  However, we can't do this if the device
-  // buffers are non-interleaved ("mono" mode).
-  if ( !stream_.deviceInterleaved[mode] && stream_.doConvertBuffer[mode] ) {
+  // "device buffers".  However, we can't do this if using multiple
+  // streams.
+  if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {
 
     bool makeBuffer = true;
     bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
@@ -1143,13 +1153,6 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
         errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
         goto error;
       }
-
-      // Save a pointer to our own device buffer in the CoreHandle
-      // structure because we may need to use the stream_.deviceBuffer
-      // variable to point to the CoreAudio buffer before buffer
-      // conversion (if we have a duplex stream with two different
-      // conversion schemes).
-      handle->deviceBuffer = stream_.deviceBuffer;
     }
   }
 
@@ -1158,23 +1161,10 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
   stream_.state = STREAM_STOPPED;
   stream_.callbackInfo.object = (void *) this;
 
-  // Setup the buffer conversion information structure.  We override
-  // the channel offset value and perform our own setting for that
-  // here.
+  // Setup the buffer conversion information structure.
   if ( stream_.doConvertBuffer[mode] ) {
-    setConvertInfo( mode, 0 );
-
-    // Add channel offset for interleaved channels.
-    if ( firstChannel > 0 && stream_.deviceInterleaved[mode] ) {
-      if ( mode == OUTPUT ) {
-        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
-          stream_.convertInfo[mode].outOffset[k] += firstChannel;
-      }
-      else {
-        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
-          stream_.convertInfo[mode].inOffset[k] += firstChannel;
-      }
-    }
+    if ( streamCount > 1 ) setConvertInfo( mode, 0 );
+    else setConvertInfo( mode, channelOffset );
   }
 
   if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
@@ -1265,8 +1255,8 @@ void RtApiCore :: closeStream( void )
     }
   }
 
-  if ( handle->deviceBuffer ) {
-    free( handle->deviceBuffer );
+  if ( stream_.deviceBuffer ) {
+    free( stream_.deviceBuffer );
     stream_.deviceBuffer = 0;
   }
 
@@ -1442,48 +1432,96 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
 
     if ( handle->drainCounter > 1 ) { // write zeros to the output stream
 
-      if ( stream_.deviceInterleaved[0] ) {
+      if ( handle->nStreams[0] == 1 ) {
         memset( outBufferList->mBuffers[handle->iStream[0]].mData,
                 0,
                 outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
       }
-      else {
-        for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+      else { // fill multiple streams with zeros
+        for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
           memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
                   0,
                   outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
         }
       }
     }
-    else if ( stream_.doConvertBuffer[0] ) {
-
-      if ( stream_.deviceInterleaved[0] )
-        stream_.deviceBuffer = (char *) outBufferList->mBuffers[handle->iStream[0]].mData;
-      else
-        stream_.deviceBuffer = handle->deviceBuffer;
-
-      convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
-
-      if ( !stream_.deviceInterleaved[0] ) {
-        UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
-        for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
-          memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
-                  &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
-        }
+    else if ( handle->nStreams[0] == 1 ) {
+      if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
+        convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
+                       stream_.userBuffer[0], stream_.convertInfo[0] );
       }
-
-    }
-    else {
-      if ( stream_.deviceInterleaved[0] ) {
+      else { // copy from user buffer
         memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
                 stream_.userBuffer[0],
                 outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
       }
-      else {
+    }
+    else { // fill multiple streams
+      Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
+      if ( stream_.doConvertBuffer[0] ) {
+        convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+        inBuffer = (Float32 *) stream_.deviceBuffer;
+      }
+
+      if ( stream_.deviceInterleaved[0] == false ) { // mono mode
         UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
-        for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+        for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
           memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
-                  &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
+                  &inBuffer[i*bufferBytes], bufferBytes );
+        }
+      }
+      else { // fill multiple multi-channel streams with interleaved data
+        UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
+        Float32 *out, *in;
+
+        bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
+        UInt32 inChannels = stream_.nUserChannels[0];
+        if ( stream_.doConvertBuffer[0] ) {
+          inInterleaved = true; // device buffer will always be interleaved for nStreams > 1
+          inChannels = stream_.nDeviceChannels[0];
+        }
+
+        if ( inInterleaved ) inOffset = 1;
+        else inOffset = stream_.bufferSize;
+
+        channelsLeft = inChannels;
+        for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
+          in = inBuffer;
+          out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
+          streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;
+
+          outJump = 0;
+          // Account for possible channel offset in first stream
+          if ( i == 0 && stream_.channelOffset[0] > 0 ) {
+            streamChannels -= stream_.channelOffset[0];
+            outJump = stream_.channelOffset[0];
+            out += outJump;
+          }
+
+          // Account for possible unfilled channels at end of the last stream
+          if ( streamChannels > channelsLeft ) {
+            outJump = streamChannels - channelsLeft;
+            streamChannels = channelsLeft;
+          }
+
+          // Determine input buffer offsets and skips
+          if ( inInterleaved ) {
+            inJump = inChannels;
+            in += inChannels - channelsLeft;
+          }
+          else {
+            inJump = 1;
+            in += (inChannels - channelsLeft) * inOffset;
+          }
+
+          for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
+            for ( unsigned int j=0; j<streamChannels; j++ ) {
+              *out++ = in[j*inOffset];
+            }
+            out += outJump;
+            in += inJump;
+          }
+          channelsLeft -= streamChannels;
         }
       }
     }
@@ -1498,26 +1536,89 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
   inputDevice = handle->id[1];
   if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
 
-    if ( stream_.doConvertBuffer[1] ) {
+    if ( handle->nStreams[1] == 1 ) {
+      if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer
+        convertBuffer( stream_.userBuffer[1],
+                       (char *) inBufferList->mBuffers[handle->iStream[1]].mData,
+                       stream_.convertInfo[1] );
+      }
+      else { // copy to user buffer
+        memcpy( stream_.userBuffer[1],
+                inBufferList->mBuffers[handle->iStream[1]].mData,
+                inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
+      }
+    }
+    else { // read from multiple streams
+      Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];
+      if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;
 
-      if ( stream_.deviceInterleaved[1] )
-        stream_.deviceBuffer = (char *) inBufferList->mBuffers[handle->iStream[1]].mData;
-      else {
-        stream_.deviceBuffer = (char *) handle->deviceBuffer;
+      if ( stream_.deviceInterleaved[1] == false ) { // mono mode
         UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
-        for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
-          memcpy( &stream_.deviceBuffer[i*bufferBytes],
+        for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+          memcpy( &outBuffer[i*bufferBytes],
                   inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
         }
       }
+      else { // read from multiple multi-channel streams
+        UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;
+        Float32 *out, *in;
+
+        bool outInterleaved = ( stream_.userInterleaved ) ? true : false;
+        UInt32 outChannels = stream_.nUserChannels[1];
+        if ( stream_.doConvertBuffer[1] ) {
+          outInterleaved = true; // device buffer will always be interleaved for nStreams > 1
+          outChannels = stream_.nDeviceChannels[1];
+        }
 
-      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+        if ( outInterleaved ) outOffset = 1;
+        else outOffset = stream_.bufferSize;
+
+        channelsLeft = outChannels;
+        for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {
+          out = outBuffer;
+          in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;
+          streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;
+
+          inJump = 0;
+          // Account for possible channel offset in first stream
+          if ( i == 0 && stream_.channelOffset[1] > 0 ) {
+            streamChannels -= stream_.channelOffset[1];
+            inJump = stream_.channelOffset[1];
+            in += inJump;
+          }
 
-    }
-    else {
-      memcpy( stream_.userBuffer[1],
-              inBufferList->mBuffers[handle->iStream[1]].mData,
-              inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
+          // Account for possible unread channels at end of the last stream
+          if ( streamChannels > channelsLeft ) {
+            inJump = streamChannels - channelsLeft;
+            streamChannels = channelsLeft;
+          }
+
+          // Determine output buffer offsets and skips
+          if ( outInterleaved ) {
+            outJump = outChannels;
+            out += outChannels - channelsLeft;
+          }
+          else {
+            outJump = 1;
+            out += (outChannels - channelsLeft) * outOffset;
+          }
+
+          for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
+            for ( unsigned int j=0; j<streamChannels; j++ ) {
+              out[j*outOffset] = *in++;
+            }
+            out += outJump;
+            in += inJump;
+          }
+          channelsLeft -= streamChannels;
+        }
+      }
+      
+      if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
+        convertBuffer( stream_.userBuffer[1],
+                       stream_.deviceBuffer,
+                       stream_.convertInfo[1] );
+      }
     }
   }
 
@@ -1528,799 +1629,799 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
   return SUCCESS;
 }
 
-const char* RtApiCore :: getErrorCode( OSStatus code )
-{
-	switch( code ) {
+  const char* RtApiCore :: getErrorCode( OSStatus code )
+  {
+    switch( code ) {
 
-  case kAudioHardwareNotRunningError:
-    return "kAudioHardwareNotRunningError";
+    case kAudioHardwareNotRunningError:
+      return "kAudioHardwareNotRunningError";
 
-  case kAudioHardwareUnspecifiedError:
-    return "kAudioHardwareUnspecifiedError";
+    case kAudioHardwareUnspecifiedError:
+      return "kAudioHardwareUnspecifiedError";
 
-  case kAudioHardwareUnknownPropertyError:
-    return "kAudioHardwareUnknownPropertyError";
+    case kAudioHardwareUnknownPropertyError:
+      return "kAudioHardwareUnknownPropertyError";
 
-  case kAudioHardwareBadPropertySizeError:
-    return "kAudioHardwareBadPropertySizeError";
+    case kAudioHardwareBadPropertySizeError:
+      return "kAudioHardwareBadPropertySizeError";
 
-  case kAudioHardwareIllegalOperationError:
-    return "kAudioHardwareIllegalOperationError";
+    case kAudioHardwareIllegalOperationError:
+      return "kAudioHardwareIllegalOperationError";
 
-  case kAudioHardwareBadObjectError:
-    return "kAudioHardwareBadObjectError";
+    case kAudioHardwareBadObjectError:
+      return "kAudioHardwareBadObjectError";
 
-  case kAudioHardwareBadDeviceError:
-    return "kAudioHardwareBadDeviceError";
+    case kAudioHardwareBadDeviceError:
+      return "kAudioHardwareBadDeviceError";
 
-  case kAudioHardwareBadStreamError:
-    return "kAudioHardwareBadStreamError";
+    case kAudioHardwareBadStreamError:
+      return "kAudioHardwareBadStreamError";
 
-  case kAudioHardwareUnsupportedOperationError:
-    return "kAudioHardwareUnsupportedOperationError";
+    case kAudioHardwareUnsupportedOperationError:
+      return "kAudioHardwareUnsupportedOperationError";
 
-  case kAudioDeviceUnsupportedFormatError:
-    return "kAudioDeviceUnsupportedFormatError";
+    case kAudioDeviceUnsupportedFormatError:
+      return "kAudioDeviceUnsupportedFormatError";
 
-  case kAudioDevicePermissionsError:
-    return "kAudioDevicePermissionsError";
+    case kAudioDevicePermissionsError:
+      return "kAudioDevicePermissionsError";
 
-  default:
-    return "CoreAudio unknown error";
-	}
-}
+    default:
+      return "CoreAudio unknown error";
+    }
+  }
 
-//******************** End of __MACOSX_CORE__ *********************//
+  //******************** End of __MACOSX_CORE__ *********************//
 #endif
 
 #if defined(__UNIX_JACK__)
 
-// JACK is a low-latency audio server, originally written for the
-// GNU/Linux operating system and now also ported to OS-X. It can
-// connect a number of different applications to an audio device, as
-// well as allowing them to share audio between themselves.
-//
-// When using JACK with RtAudio, "devices" refer to JACK clients that
-// have ports connected to the server.  The JACK server is typically
-// started in a terminal as follows:
-//
-// .jackd -d alsa -d hw:0
-//
-// or through an interface program such as qjackctl.  Many of the
-// parameters normally set for a stream are fixed by the JACK server
-// and can be specified when the JACK server is started.  In
-// particular,
-//
-// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
-//
-// specifies a sample rate of 44100 Hz, a buffer size of 512 sample
-// frames, and number of buffers = 4.  Once the server is running, it
-// is not possible to override these values.  If the values are not
-// specified in the command-line, the JACK server uses default values.
-//
-// The JACK server does not have to be running when an instance of
-// RtApiJack is created, though the function getDeviceCount() will
-// report 0 devices found until JACK has been started.  When no
-// devices are available (i.e., the JACK server is not running), a
-// stream cannot be opened.
+  // JACK is a low-latency audio server, originally written for the
+  // GNU/Linux operating system and now also ported to OS-X. It can
+  // connect a number of different applications to an audio device, as
+  // well as allowing them to share audio between themselves.
+  //
+  // When using JACK with RtAudio, "devices" refer to JACK clients that
+  // have ports connected to the server.  The JACK server is typically
+  // started in a terminal as follows:
+  //
+  // .jackd -d alsa -d hw:0
+  //
+  // or through an interface program such as qjackctl.  Many of the
+  // parameters normally set for a stream are fixed by the JACK server
+  // and can be specified when the JACK server is started.  In
+  // particular,
+  //
+  // .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
+  //
+  // specifies a sample rate of 44100 Hz, a buffer size of 512 sample
+  // frames, and number of buffers = 4.  Once the server is running, it
+  // is not possible to override these values.  If the values are not
+  // specified in the command-line, the JACK server uses default values.
+  //
+  // The JACK server does not have to be running when an instance of
+  // RtApiJack is created, though the function getDeviceCount() will
+  // report 0 devices found until JACK has been started.  When no
+  // devices are available (i.e., the JACK server is not running), a
+  // stream cannot be opened.
 
 #include <jack/jack.h>
 #include <unistd.h>
 
-// A structure to hold various information related to the Jack API
-// implementation.
-struct JackHandle {
-  jack_client_t *client;
-  jack_port_t **ports[2];
-  std::string deviceName[2];
-  bool xrun[2];
-  pthread_cond_t condition;
-  int drainCounter;       // Tracks callback counts when draining
-  bool internalDrain;     // Indicates if stop is initiated from callback or not.
-
-  JackHandle()
-    :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
-};
+  // A structure to hold various information related to the Jack API
+  // implementation.
+  struct JackHandle {
+    jack_client_t *client;
+    jack_port_t **ports[2];
+    std::string deviceName[2];
+    bool xrun[2];
+    pthread_cond_t condition;
+    int drainCounter;       // Tracks callback counts when draining
+    bool internalDrain;     // Indicates if stop is initiated from callback or not.
+
+    JackHandle()
+      :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
+  };
 
-RtApiJack :: RtApiJack()
-{
-  // Nothing to do here.
-}
+  RtApiJack :: RtApiJack()
+  {
+    // Nothing to do here.
+  }
 
-RtApiJack :: ~RtApiJack()
-{
-  if ( stream_.state != STREAM_CLOSED ) closeStream();
-}
+  RtApiJack :: ~RtApiJack()
+  {
+    if ( stream_.state != STREAM_CLOSED ) closeStream();
+  }
 
-unsigned int RtApiJack :: getDeviceCount( void )
-{
-  // See if we can become a jack client.
-  jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption;
-  jack_status_t *status = NULL;
-  jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
-  if ( client == 0 ) return 0;
-
-  const char **ports;
-  std::string port, previousPort;
-  unsigned int nChannels = 0, nDevices = 0;
-  ports = jack_get_ports( client, NULL, NULL, 0 );
-  if ( ports ) {
-    // Parse the port names up to the first colon (:).
-    size_t iColon = 0;
-    do {
-      port = (char *) ports[ nChannels ];
-      iColon = port.find(":");
-      if ( iColon != std::string::npos ) {
-        port = port.substr( 0, iColon + 1 );
-        if ( port != previousPort ) {
-          nDevices++;
-          previousPort = port;
+  unsigned int RtApiJack :: getDeviceCount( void )
+  {
+    // See if we can become a jack client.
+    jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption;
+    jack_status_t *status = NULL;
+    jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
+    if ( client == 0 ) return 0;
+
+    const char **ports;
+    std::string port, previousPort;
+    unsigned int nChannels = 0, nDevices = 0;
+    ports = jack_get_ports( client, NULL, NULL, 0 );
+    if ( ports ) {
+      // Parse the port names up to the first colon (:).
+      size_t iColon = 0;
+      do {
+        port = (char *) ports[ nChannels ];
+        iColon = port.find(":");
+        if ( iColon != std::string::npos ) {
+          port = port.substr( 0, iColon + 1 );
+          if ( port != previousPort ) {
+            nDevices++;
+            previousPort = port;
+          }
         }
-      }
-    } while ( ports[++nChannels] );
-    free( ports );
-  }
+      } while ( ports[++nChannels] );
+      free( ports );
+    }
 
-  jack_client_close( client );
-  return nDevices;
-}
+    jack_client_close( client );
+    return nDevices;
+  }
 
-RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
-{
-  RtAudio::DeviceInfo info;
-  info.probed = false;
+  RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
+  {
+    RtAudio::DeviceInfo info;
+    info.probed = false;
 
-  jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption
-  jack_status_t *status = NULL;
-  jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
-  if ( client == 0 ) {
-    errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
-    error( RtError::WARNING );
-    return info;
-  }
+    jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption
+    jack_status_t *status = NULL;
+    jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
+    if ( client == 0 ) {
+      errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
+      error( RtError::WARNING );
+      return info;
+    }
 
-  const char **ports;
-  std::string port, previousPort;
-  unsigned int nPorts = 0, nDevices = 0;
-  ports = jack_get_ports( client, NULL, NULL, 0 );
-  if ( ports ) {
-    // Parse the port names up to the first colon (:).
-    size_t iColon = 0;
-    do {
-      port = (char *) ports[ nPorts ];
-      iColon = port.find(":");
-      if ( iColon != std::string::npos ) {
-        port = port.substr( 0, iColon );
-        if ( port != previousPort ) {
-          if ( nDevices == device ) info.name = port;
-          nDevices++;
-          previousPort = port;
+    const char **ports;
+    std::string port, previousPort;
+    unsigned int nPorts = 0, nDevices = 0;
+    ports = jack_get_ports( client, NULL, NULL, 0 );
+    if ( ports ) {
+      // Parse the port names up to the first colon (:).
+      size_t iColon = 0;
+      do {
+        port = (char *) ports[ nPorts ];
+        iColon = port.find(":");
+        if ( iColon != std::string::npos ) {
+          port = port.substr( 0, iColon );
+          if ( port != previousPort ) {
+            if ( nDevices == device ) info.name = port;
+            nDevices++;
+            previousPort = port;
+          }
         }
-      }
-    } while ( ports[++nPorts] );
-    free( ports );
-  }
+      } while ( ports[++nPorts] );
+      free( ports );
+    }
 
-  if ( device >= nDevices ) {
-    errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
-    error( RtError::INVALID_USE );
-  }
+    if ( device >= nDevices ) {
+      errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
+      error( RtError::INVALID_USE );
+    }
 
-  // Get the current jack server sample rate.
-  info.sampleRates.clear();
-  info.sampleRates.push_back( jack_get_sample_rate( client ) );
-
-  // Count the available ports containing the client name as device
-  // channels.  Jack "input ports" equal RtAudio output channels.
-  unsigned int nChannels = 0;
-  ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
-  if ( ports ) {
-    while ( ports[ nChannels ] ) nChannels++;
-    free( ports );
-    info.outputChannels = nChannels;
-  }
+    // Get the current jack server sample rate.
+    info.sampleRates.clear();
+    info.sampleRates.push_back( jack_get_sample_rate( client ) );
 
-  // Jack "output ports" equal RtAudio input channels.
-  nChannels = 0;
-  ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
-  if ( ports ) {
-    while ( ports[ nChannels ] ) nChannels++;
-    free( ports );
-    info.inputChannels = nChannels;
-  }
+    // Count the available ports containing the client name as device
+    // channels.  Jack "input ports" equal RtAudio output channels.
+    unsigned int nChannels = 0;
+    ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
+    if ( ports ) {
+      while ( ports[ nChannels ] ) nChannels++;
+      free( ports );
+      info.outputChannels = nChannels;
+    }
 
-  if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
-    jack_client_close(client);
-    errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
-    error( RtError::WARNING );
-    return info;
-  }
+    // Jack "output ports" equal RtAudio input channels.
+    nChannels = 0;
+    ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
+    if ( ports ) {
+      while ( ports[ nChannels ] ) nChannels++;
+      free( ports );
+      info.inputChannels = nChannels;
+    }
 
-  // If device opens for both playback and capture, we determine the channels.
-  if ( info.outputChannels > 0 && info.inputChannels > 0 )
-    info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+    if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
+      jack_client_close(client);
+      errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
+      error( RtError::WARNING );
+      return info;
+    }
 
-  // Jack always uses 32-bit floats.
-  info.nativeFormats = RTAUDIO_FLOAT32;
+    // If device opens for both playback and capture, we determine the channels.
+    if ( info.outputChannels > 0 && info.inputChannels > 0 )
+      info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
 
-  // Jack doesn't provide default devices so we'll use the first available one.
-  if ( device == 0 && info.outputChannels > 0 )
-    info.isDefaultOutput = true;
-  if ( device == 0 && info.inputChannels > 0 )
-    info.isDefaultInput = true;
+    // Jack always uses 32-bit floats.
+    info.nativeFormats = RTAUDIO_FLOAT32;
 
-  jack_client_close(client);
-  info.probed = true;
-  return info;
-}
+    // Jack doesn't provide default devices so we'll use the first available one.
+    if ( device == 0 && info.outputChannels > 0 )
+      info.isDefaultOutput = true;
+    if ( device == 0 && info.inputChannels > 0 )
+      info.isDefaultInput = true;
 
-int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
-{
-  CallbackInfo *info = (CallbackInfo *) infoPointer;
+    jack_client_close(client);
+    info.probed = true;
+    return info;
+  }
 
-  RtApiJack *object = (RtApiJack *) info->object;
-  if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
+  int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
+  {
+    CallbackInfo *info = (CallbackInfo *) infoPointer;
 
-  return 0;
-}
+    RtApiJack *object = (RtApiJack *) info->object;
+    if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
 
-void jackShutdown( void *infoPointer )
-{
-  CallbackInfo *info = (CallbackInfo *) infoPointer;
-  RtApiJack *object = (RtApiJack *) info->object;
+    return 0;
+  }
 
-  // Check current stream state.  If stopped, then we'll assume this
-  // was called as a result of a call to RtApiJack::stopStream (the
-  // deactivation of a client handle causes this function to be called).
-  // If not, we'll assume the Jack server is shutting down or some
-  // other problem occurred and we should close the stream.
-  if ( object->isStreamRunning() == false ) return;
+  void jackShutdown( void *infoPointer )
+  {
+    CallbackInfo *info = (CallbackInfo *) infoPointer;
+    RtApiJack *object = (RtApiJack *) info->object;
 
-  object->closeStream();
-  std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
-}
+    // Check current stream state.  If stopped, then we'll assume this
+    // was called as a result of a call to RtApiJack::stopStream (the
+    // deactivation of a client handle causes this function to be called).
+    // If not, we'll assume the Jack server is shutting down or some
+    // other problem occurred and we should close the stream.
+    if ( object->isStreamRunning() == false ) return;
 
-int jackXrun( void *infoPointer )
-{
-  JackHandle *handle = (JackHandle *) infoPointer;
+    object->closeStream();
+    std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
+  }
 
-  if ( handle->ports[0] ) handle->xrun[0] = true;
-  if ( handle->ports[1] ) handle->xrun[1] = true;
+  int jackXrun( void *infoPointer )
+  {
+    JackHandle *handle = (JackHandle *) infoPointer;
 
-  return 0;
-}
+    if ( handle->ports[0] ) handle->xrun[0] = true;
+    if ( handle->ports[1] ) handle->xrun[1] = true;
 
-bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-                                   unsigned int firstChannel, unsigned int sampleRate,
-                                   RtAudioFormat format, unsigned int *bufferSize,
-                                   RtAudio::StreamOptions *options )
-{
-  JackHandle *handle = (JackHandle *) stream_.apiHandle;
+    return 0;
+  }
 
-  // Look for jack server and try to become a client (only do once per stream).
-  jack_client_t *client = 0;
-  if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
-    jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption;
-    jack_status_t *status = NULL;
-    if ( options && !options->streamName.empty() )
-      client = jack_client_open( options->streamName.c_str(), jackoptions, status );
-    else
-      client = jack_client_open( "RtApiJack", jackoptions, status );
-    if ( client == 0 ) {
-      errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
-      error( RtError::WARNING );
-      return FAILURE;
+  bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+                                     unsigned int firstChannel, unsigned int sampleRate,
+                                     RtAudioFormat format, unsigned int *bufferSize,
+                                     RtAudio::StreamOptions *options )
+  {
+    JackHandle *handle = (JackHandle *) stream_.apiHandle;
+
+    // Look for jack server and try to become a client (only do once per stream).
+    jack_client_t *client = 0;
+    if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
+      jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption;
+      jack_status_t *status = NULL;
+      if ( options && !options->streamName.empty() )
+        client = jack_client_open( options->streamName.c_str(), jackoptions, status );
+      else
+        client = jack_client_open( "RtApiJack", jackoptions, status );
+      if ( client == 0 ) {
+        errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
+        error( RtError::WARNING );
+        return FAILURE;
+      }
     }
-  }
-  else {
-    // The handle must have been created on an earlier pass.
-    client = handle->client;
-  }
-
-  const char **ports;
-  std::string port, previousPort, deviceName;
-  unsigned int nPorts = 0, nDevices = 0;
-  ports = jack_get_ports( client, NULL, NULL, 0 );
-  if ( ports ) {
-    // Parse the port names up to the first colon (:).
-    size_t iColon = 0;
-    do {
-      port = (char *) ports[ nPorts ];
-      iColon = port.find(":");
-      if ( iColon != std::string::npos ) {
-        port = port.substr( 0, iColon );
-        if ( port != previousPort ) {
-          if ( nDevices == device ) deviceName = port;
-          nDevices++;
-          previousPort = port;
+    else {
+      // The handle must have been created on an earlier pass.
+      client = handle->client;
+    }
+
+    const char **ports;
+    std::string port, previousPort, deviceName;
+    unsigned int nPorts = 0, nDevices = 0;
+    ports = jack_get_ports( client, NULL, NULL, 0 );
+    if ( ports ) {
+      // Parse the port names up to the first colon (:).
+      size_t iColon = 0;
+      do {
+        port = (char *) ports[ nPorts ];
+        iColon = port.find(":");
+        if ( iColon != std::string::npos ) {
+          port = port.substr( 0, iColon );
+          if ( port != previousPort ) {
+            if ( nDevices == device ) deviceName = port;
+            nDevices++;
+            previousPort = port;
+          }
         }
-      }
-    } while ( ports[++nPorts] );
-    free( ports );
-  }
-
-  if ( device >= nDevices ) {
-    errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
-    return FAILURE;
-  }
+      } while ( ports[++nPorts] );
+      free( ports );
+    }
 
-  // Count the available ports containing the client name as device
-  // channels.  Jack "input ports" equal RtAudio output channels.
-  unsigned int nChannels = 0;
-  unsigned long flag = JackPortIsInput;
-  if ( mode == INPUT ) flag = JackPortIsOutput;
-  ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
-  if ( ports ) {
-    while ( ports[ nChannels ] ) nChannels++;
-    free( ports );
-  }
+    if ( device >= nDevices ) {
+      errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
+      return FAILURE;
+    }
 
-  // Compare the jack ports for specified client to the requested number of channels.
-  if ( nChannels < (channels + firstChannel) ) {
-    errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+    // Count the available ports containing the client name as device
+    // channels.  Jack "input ports" equal RtAudio output channels.
+    unsigned int nChannels = 0;
+    unsigned long flag = JackPortIsInput;
+    if ( mode == INPUT ) flag = JackPortIsOutput;
+    ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
+    if ( ports ) {
+      while ( ports[ nChannels ] ) nChannels++;
+      free( ports );
+    }
 
-  // Check the jack server sample rate.
-  unsigned int jackRate = jack_get_sample_rate( client );
-  if ( sampleRate != jackRate ) {
-    jack_client_close( client );
-    errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-  stream_.sampleRate = jackRate;
+    // Compare the jack ports for specified client to the requested number of channels.
+    if ( nChannels < (channels + firstChannel) ) {
+      errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
-  // Get the latency of the JACK port.
-  ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
-  if ( ports[ firstChannel ] )
-    stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
-  free( ports );
+    // Check the jack server sample rate.
+    unsigned int jackRate = jack_get_sample_rate( client );
+    if ( sampleRate != jackRate ) {
+      jack_client_close( client );
+      errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+    stream_.sampleRate = jackRate;
 
-  // The jack server always uses 32-bit floating-point data.
-  stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
-  stream_.userFormat = format;
+    // Get the latency of the JACK port.
+    ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
+    if ( ports[ firstChannel ] )
+      stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
+    free( ports );
 
-  if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
-  else stream_.userInterleaved = true;
+    // The jack server always uses 32-bit floating-point data.
+    stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+    stream_.userFormat = format;
 
-  // Jack always uses non-interleaved buffers.
-  stream_.deviceInterleaved[mode] = false;
+    if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+    else stream_.userInterleaved = true;
 
-  // Jack always provides host byte-ordered data.
-  stream_.doByteSwap[mode] = false;
+    // Jack always uses non-interleaved buffers.
+    stream_.deviceInterleaved[mode] = false;
 
-  // Get the buffer size.  The buffer size and number of buffers
-  // (periods) is set when the jack server is started.
-  stream_.bufferSize = (int) jack_get_buffer_size( client );
-  *bufferSize = stream_.bufferSize;
+    // Jack always provides host byte-ordered data.
+    stream_.doByteSwap[mode] = false;
 
-  stream_.nDeviceChannels[mode] = channels;
-  stream_.nUserChannels[mode] = channels;
+    // Get the buffer size.  The buffer size and number of buffers
+    // (periods) is set when the jack server is started.
+    stream_.bufferSize = (int) jack_get_buffer_size( client );
+    *bufferSize = stream_.bufferSize;
 
-  // Set flags for buffer conversion.
-  stream_.doConvertBuffer[mode] = false;
-  if ( stream_.userFormat != stream_.deviceFormat[mode] )
-    stream_.doConvertBuffer[mode] = true;
-  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
-       stream_.nUserChannels[mode] > 1 )
-    stream_.doConvertBuffer[mode] = true;
+    stream_.nDeviceChannels[mode] = channels;
+    stream_.nUserChannels[mode] = channels;
+
+    // Set flags for buffer conversion.
+    stream_.doConvertBuffer[mode] = false;
+    if ( stream_.userFormat != stream_.deviceFormat[mode] )
+      stream_.doConvertBuffer[mode] = true;
+    if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+         stream_.nUserChannels[mode] > 1 )
+      stream_.doConvertBuffer[mode] = true;
+
+    // Allocate our JackHandle structure for the stream.
+    if ( handle == 0 ) {
+      try {
+        handle = new JackHandle;
+      }
+      catch ( std::bad_alloc& ) {
+        errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
+        goto error;
+      }
 
-  // Allocate our JackHandle structure for the stream.
-  if ( handle == 0 ) {
-    try {
-      handle = new JackHandle;
-    }
-    catch ( std::bad_alloc& ) {
-      errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
-      goto error;
+      if ( pthread_cond_init(&handle->condition, NULL) ) {
+        errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
+        goto error;
+      }
+      stream_.apiHandle = (void *) handle;
+      handle->client = client;
     }
+    handle->deviceName[mode] = deviceName;
 
-    if ( pthread_cond_init(&handle->condition, NULL) ) {
-      errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
+    // Allocate necessary internal buffers.
+    unsigned long bufferBytes;
+    bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+    stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+    if ( stream_.userBuffer[mode] == NULL ) {
+      errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
       goto error;
     }
-    stream_.apiHandle = (void *) handle;
-    handle->client = client;
-  }
-  handle->deviceName[mode] = deviceName;
 
-  // Allocate necessary internal buffers.
-  unsigned long bufferBytes;
-  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
-  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
-  if ( stream_.userBuffer[mode] == NULL ) {
-    errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
-    goto error;
-  }
+    if ( stream_.doConvertBuffer[mode] ) {
 
-  if ( stream_.doConvertBuffer[mode] ) {
-
-    bool makeBuffer = true;
-    if ( mode == OUTPUT )
-      bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
-    else { // mode == INPUT
-      bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
-      if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
-        unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
-        if ( bufferBytes < bytesOut ) makeBuffer = false;
+      bool makeBuffer = true;
+      if ( mode == OUTPUT )
+        bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+      else { // mode == INPUT
+        bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
+        if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+          unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
+          if ( bufferBytes < bytesOut ) makeBuffer = false;
+        }
       }
-    }
 
-    if ( makeBuffer ) {
-      bufferBytes *= *bufferSize;
-      if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
-      stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
-      if ( stream_.deviceBuffer == NULL ) {
-        errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
-        goto error;
+      if ( makeBuffer ) {
+        bufferBytes *= *bufferSize;
+        if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+        stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+        if ( stream_.deviceBuffer == NULL ) {
+          errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
+          goto error;
+        }
       }
     }
-  }
 
-  // Allocate memory for the Jack ports (channels) identifiers.
-  handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
-  if ( handle->ports[mode] == NULL )  {
-    errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
-    goto error;
-  }
+    // Allocate memory for the Jack ports (channels) identifiers.
+    handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
+    if ( handle->ports[mode] == NULL )  {
+      errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
+      goto error;
+    }
 
-  stream_.device[mode] = device;
-  stream_.channelOffset[mode] = firstChannel;
-  stream_.state = STREAM_STOPPED;
-  stream_.callbackInfo.object = (void *) this;
+    stream_.device[mode] = device;
+    stream_.channelOffset[mode] = firstChannel;
+    stream_.state = STREAM_STOPPED;
+    stream_.callbackInfo.object = (void *) this;
 
-  if ( stream_.mode == OUTPUT && mode == INPUT )
-    // We had already set up the stream for output.
-    stream_.mode = DUPLEX;
-  else {
-    stream_.mode = mode;
-    jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
-    jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );
-    jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
-  }
+    if ( stream_.mode == OUTPUT && mode == INPUT )
+      // We had already set up the stream for output.
+      stream_.mode = DUPLEX;
+    else {
+      stream_.mode = mode;
+      jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
+      jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );
+      jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
+    }
 
-  // Register our ports.
-  char label[64];
-  if ( mode == OUTPUT ) {
-    for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
-      snprintf( label, 64, "outport %d", i );
-      handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
-                                                JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
+    // Register our ports.
+    char label[64];
+    if ( mode == OUTPUT ) {
+      for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+        snprintf( label, 64, "outport %d", i );
+        handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
+                                                  JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
+      }
     }
-  }
-  else {
-    for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
-      snprintf( label, 64, "inport %d", i );
-      handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
-                                                JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
+    else {
+      for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+        snprintf( label, 64, "inport %d", i );
+        handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
+                                                  JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
+      }
     }
-  }
 
-  // Setup the buffer conversion information structure.  We don't use
-  // buffers to do channel offsets, so we override that parameter
-  // here.
-  if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
+    // Setup the buffer conversion information structure.  We don't use
+    // buffers to do channel offsets, so we override that parameter
+    // here.
+    if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
 
-  return SUCCESS;
+    return SUCCESS;
 
- error:
-  if ( handle ) {
-    pthread_cond_destroy( &handle->condition );
-    jack_client_close( handle->client );
+  error:
+    if ( handle ) {
+      pthread_cond_destroy( &handle->condition );
+      jack_client_close( handle->client );
 
-    if ( handle->ports[0] ) free( handle->ports[0] );
-    if ( handle->ports[1] ) free( handle->ports[1] );
+      if ( handle->ports[0] ) free( handle->ports[0] );
+      if ( handle->ports[1] ) free( handle->ports[1] );
 
-    delete handle;
-    stream_.apiHandle = 0;
-  }
-
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
+      delete handle;
+      stream_.apiHandle = 0;
     }
-  }
 
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
+    for ( int i=0; i<2; i++ ) {
+      if ( stream_.userBuffer[i] ) {
+        free( stream_.userBuffer[i] );
+        stream_.userBuffer[i] = 0;
+      }
+    }
 
-  return FAILURE;
-}
+    if ( stream_.deviceBuffer ) {
+      free( stream_.deviceBuffer );
+      stream_.deviceBuffer = 0;
+    }
 
-void RtApiJack :: closeStream( void )
-{
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiJack::closeStream(): no open stream to close!";
-    error( RtError::WARNING );
-    return;
+    return FAILURE;
   }
 
-  JackHandle *handle = (JackHandle *) stream_.apiHandle;
-  if ( handle ) {
+  void RtApiJack :: closeStream( void )
+  {
+    if ( stream_.state == STREAM_CLOSED ) {
+      errorText_ = "RtApiJack::closeStream(): no open stream to close!";
+      error( RtError::WARNING );
+      return;
+    }
 
-    if ( stream_.state == STREAM_RUNNING )
-      jack_deactivate( handle->client );
+    JackHandle *handle = (JackHandle *) stream_.apiHandle;
+    if ( handle ) {
 
-    jack_client_close( handle->client );
-  }
+      if ( stream_.state == STREAM_RUNNING )
+        jack_deactivate( handle->client );
 
-  if ( handle ) {
-    if ( handle->ports[0] ) free( handle->ports[0] );
-    if ( handle->ports[1] ) free( handle->ports[1] );
-    pthread_cond_destroy( &handle->condition );
-    delete handle;
-    stream_.apiHandle = 0;
-  }
+      jack_client_close( handle->client );
+    }
 
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
+    if ( handle ) {
+      if ( handle->ports[0] ) free( handle->ports[0] );
+      if ( handle->ports[1] ) free( handle->ports[1] );
+      pthread_cond_destroy( &handle->condition );
+      delete handle;
+      stream_.apiHandle = 0;
     }
-  }
 
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
+    for ( int i=0; i<2; i++ ) {
+      if ( stream_.userBuffer[i] ) {
+        free( stream_.userBuffer[i] );
+        stream_.userBuffer[i] = 0;
+      }
+    }
 
-  stream_.mode = UNINITIALIZED;
-  stream_.state = STREAM_CLOSED;
-}
+    if ( stream_.deviceBuffer ) {
+      free( stream_.deviceBuffer );
+      stream_.deviceBuffer = 0;
+    }
 
-void RtApiJack :: startStream( void )
-{
-  verifyStream();
-  if ( stream_.state == STREAM_RUNNING ) {
-    errorText_ = "RtApiJack::startStream(): the stream is already running!";
-    error( RtError::WARNING );
-    return;
+    stream_.mode = UNINITIALIZED;
+    stream_.state = STREAM_CLOSED;
   }
 
-  MUTEX_LOCK(&stream_.mutex);
-
-  JackHandle *handle = (JackHandle *) stream_.apiHandle;
-  int result = jack_activate( handle->client );
-  if ( result ) {
-    errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
-    goto unlock;
-  }
+  void RtApiJack :: startStream( void )
+  {
+    verifyStream();
+    if ( stream_.state == STREAM_RUNNING ) {
+      errorText_ = "RtApiJack::startStream(): the stream is already running!";
+      error( RtError::WARNING );
+      return;
+    }
 
-  const char **ports;
+    MUTEX_LOCK(&stream_.mutex);
 
-  // Get the list of available ports.
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-    result = 1;
-    ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);
-    if ( ports == NULL) {
-      errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
+    JackHandle *handle = (JackHandle *) stream_.apiHandle;
+    int result = jack_activate( handle->client );
+    if ( result ) {
+      errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
       goto unlock;
     }
 
-    // Now make the port connections.  Since RtAudio wasn't designed to
-    // allow the user to select particular channels of a device, we'll
-    // just open the first "nChannels" ports with offset.
-    for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+    const char **ports;
+
+    // Get the list of available ports.
+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
       result = 1;
-      if ( ports[ stream_.channelOffset[0] + i ] )
-        result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
-      if ( result ) {
-        free( ports );
-        errorText_ = "RtApiJack::startStream(): error connecting output ports!";
+      ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);
+      if ( ports == NULL) {
+        errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
         goto unlock;
       }
-    }
-    free(ports);
-  }
 
-  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-    result = 1;
-    ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );
-    if ( ports == NULL) {
-      errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
-      goto unlock;
+      // Now make the port connections.  Since RtAudio wasn't designed to
+      // allow the user to select particular channels of a device, we'll
+      // just open the first "nChannels" ports with offset.
+      for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+        result = 1;
+        if ( ports[ stream_.channelOffset[0] + i ] )
+          result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
+        if ( result ) {
+          free( ports );
+          errorText_ = "RtApiJack::startStream(): error connecting output ports!";
+          goto unlock;
+        }
+      }
+      free(ports);
     }
 
-    // Now make the port connections.  See note above.
-    for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+    if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
       result = 1;
-      if ( ports[ stream_.channelOffset[1] + i ] )
-        result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
-      if ( result ) {
-        free( ports );
-        errorText_ = "RtApiJack::startStream(): error connecting input ports!";
+      ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );
+      if ( ports == NULL) {
+        errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
         goto unlock;
       }
+
+      // Now make the port connections.  See note above.
+      for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+        result = 1;
+        if ( ports[ stream_.channelOffset[1] + i ] )
+          result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
+        if ( result ) {
+          free( ports );
+          errorText_ = "RtApiJack::startStream(): error connecting input ports!";
+          goto unlock;
+        }
+      }
+      free(ports);
     }
-    free(ports);
-  }
 
-  handle->drainCounter = 0;
-  handle->internalDrain = false;
-  stream_.state = STREAM_RUNNING;
+    handle->drainCounter = 0;
+    handle->internalDrain = false;
+    stream_.state = STREAM_RUNNING;
 
- unlock:
-  MUTEX_UNLOCK(&stream_.mutex);
+  unlock:
+    MUTEX_UNLOCK(&stream_.mutex);
 
-  if ( result == 0 ) return;
-  error( RtError::SYSTEM_ERROR );
-}
-
-void RtApiJack :: stopStream( void )
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
+    if ( result == 0 ) return;
+    error( RtError::SYSTEM_ERROR );
   }
 
-  MUTEX_LOCK( &stream_.mutex );
-
-  JackHandle *handle = (JackHandle *) stream_.apiHandle;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
-    if ( handle->drainCounter == 0 ) {
-      handle->drainCounter = 1;
-      pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
+  void RtApiJack :: stopStream( void )
+  {
+    verifyStream();
+    if ( stream_.state == STREAM_STOPPED ) {
+      errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
+      error( RtError::WARNING );
+      return;
     }
-  }
-
-  jack_deactivate( handle->client );
-  stream_.state = STREAM_STOPPED;
 
-  MUTEX_UNLOCK( &stream_.mutex );
-}
+    MUTEX_LOCK( &stream_.mutex );
 
-void RtApiJack :: abortStream( void )
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
-  }
+    JackHandle *handle = (JackHandle *) stream_.apiHandle;
+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
 
-  JackHandle *handle = (JackHandle *) stream_.apiHandle;
-  handle->drainCounter = 1;
+      if ( handle->drainCounter == 0 ) {
+        handle->drainCounter = 1;
+        pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
+      }
+    }
 
-  stopStream();
-}
+    jack_deactivate( handle->client );
+    stream_.state = STREAM_STOPPED;
 
-bool RtApiJack :: callbackEvent( unsigned long nframes )
-{
-  if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
-    error( RtError::WARNING );
-    return FAILURE;
-  }
-  if ( stream_.bufferSize != nframes ) {
-    errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
-    error( RtError::WARNING );
-    return FAILURE;
+    MUTEX_UNLOCK( &stream_.mutex );
   }
 
-  CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
-  JackHandle *handle = (JackHandle *) stream_.apiHandle;
+  void RtApiJack :: abortStream( void )
+  {
+    verifyStream();
+    if ( stream_.state == STREAM_STOPPED ) {
+      errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
+      error( RtError::WARNING );
+      return;
+    }
 
-  // Check if we were draining the stream and signal is finished.
-  if ( handle->drainCounter > 3 ) {
-    if ( handle->internalDrain == false )
-      pthread_cond_signal( &handle->condition );
-    else
-      stopStream();
-    return SUCCESS;
-  }
+    JackHandle *handle = (JackHandle *) stream_.apiHandle;
+    handle->drainCounter = 1;
 
-  MUTEX_LOCK( &stream_.mutex );
+    stopStream();
+  }
 
-  // Invoke user callback first, to get fresh output data.
-  if ( handle->drainCounter == 0 ) {
-    RtAudioCallback callback = (RtAudioCallback) info->callback;
-    double streamTime = getStreamTime();
-    RtAudioStreamStatus status = 0;
-    if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
-      status |= RTAUDIO_OUTPUT_UNDERFLOW;
-      handle->xrun[0] = false;
+  bool RtApiJack :: callbackEvent( unsigned long nframes )
+  {
+    if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
+    if ( stream_.state == STREAM_CLOSED ) {
+      errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
+      error( RtError::WARNING );
+      return FAILURE;
     }
-    if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
-      status |= RTAUDIO_INPUT_OVERFLOW;
-      handle->xrun[1] = false;
+    if ( stream_.bufferSize != nframes ) {
+      errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
+      error( RtError::WARNING );
+      return FAILURE;
     }
-    handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
-                                     stream_.bufferSize, streamTime, status, info->userData );
-    if ( handle->drainCounter == 2 ) {
-      MUTEX_UNLOCK( &stream_.mutex );
-      abortStream();
+
+    CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+    JackHandle *handle = (JackHandle *) stream_.apiHandle;
+
+    // Check if we were draining the stream and signal is finished.
+    if ( handle->drainCounter > 3 ) {
+      if ( handle->internalDrain == false )
+        pthread_cond_signal( &handle->condition );
+      else
+        stopStream();
       return SUCCESS;
     }
-    else if ( handle->drainCounter == 1 )
-      handle->internalDrain = true;
-  }
-
-  jack_default_audio_sample_t *jackbuffer;
-  unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
 
-    if ( handle->drainCounter > 0 ) { // write zeros to the output stream
+    MUTEX_LOCK( &stream_.mutex );
 
-      for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
-        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
-        memset( jackbuffer, 0, bufferBytes );
+    // Invoke user callback first, to get fresh output data.
+    if ( handle->drainCounter == 0 ) {
+      RtAudioCallback callback = (RtAudioCallback) info->callback;
+      double streamTime = getStreamTime();
+      RtAudioStreamStatus status = 0;
+      if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+        status |= RTAUDIO_OUTPUT_UNDERFLOW;
+        handle->xrun[0] = false;
       }
-
+      if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+        status |= RTAUDIO_INPUT_OVERFLOW;
+        handle->xrun[1] = false;
+      }
+      handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+                                       stream_.bufferSize, streamTime, status, info->userData );
+      if ( handle->drainCounter == 2 ) {
+        MUTEX_UNLOCK( &stream_.mutex );
+        abortStream();
+        return SUCCESS;
+      }
+      else if ( handle->drainCounter == 1 )
+        handle->internalDrain = true;
     }
-    else if ( stream_.doConvertBuffer[0] ) {
 
-      convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+    jack_default_audio_sample_t *jackbuffer;
+    unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+      if ( handle->drainCounter > 0 ) { // write zeros to the output stream
+
+        for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+          jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+          memset( jackbuffer, 0, bufferBytes );
+        }
 
-      for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
-        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
-        memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
       }
-    }
-    else { // no buffer conversion
-      for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
-        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
-        memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
+      else if ( stream_.doConvertBuffer[0] ) {
+
+        convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+
+        for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+          jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+          memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
+        }
+      }
+      else { // no buffer conversion
+        for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+          jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+          memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
+        }
       }
-    }
 
-    if ( handle->drainCounter ) {
-      handle->drainCounter++;
-      goto unlock;
+      if ( handle->drainCounter ) {
+        handle->drainCounter++;
+        goto unlock;
+      }
     }
-  }
 
-  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+    if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
 
-    if ( stream_.doConvertBuffer[1] ) {
-      for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
-        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
-        memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
+      if ( stream_.doConvertBuffer[1] ) {
+        for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
+          jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
+          memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
+        }
+        convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
       }
-      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
-    }
-    else { // no buffer conversion
-      for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
-        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
-        memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
+      else { // no buffer conversion
+        for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+          jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
+          memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
+        }
       }
     }
-  }
 
- unlock:
-  MUTEX_UNLOCK(&stream_.mutex);
+  unlock:
+    MUTEX_UNLOCK(&stream_.mutex);
 
-  RtApi::tickStreamTime();
-  return SUCCESS;
-}
-//******************** End of __UNIX_JACK__ *********************//
+    RtApi::tickStreamTime();
+    return SUCCESS;
+  }
+  //******************** End of __UNIX_JACK__ *********************//
 #endif
 
 #if defined(__WINDOWS_ASIO__) // ASIO API on Windows
 
-// The ASIO API is designed around a callback scheme, so this
-// implementation is similar to that used for OS-X CoreAudio and Linux
-// Jack.  The primary constraint with ASIO is that it only allows
-// access to a single driver at a time.  Thus, it is not possible to
-// have more than one simultaneous RtAudio stream.
-//
-// This implementation also requires a number of external ASIO files
-// and a few global variables.  The ASIO callback scheme does not
-// allow for the passing of user data, so we must create a global
-// pointer to our callbackInfo structure.
-//
-// On unix systems, we make use of a pthread condition variable.
-// Since there is no equivalent in Windows, I hacked something based
-// on information found in
-// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
+  // The ASIO API is designed around a callback scheme, so this
+  // implementation is similar to that used for OS-X CoreAudio and Linux
+  // Jack.  The primary constraint with ASIO is that it only allows
+  // access to a single driver at a time.  Thus, it is not possible to
+  // have more than one simultaneous RtAudio stream.
+  //
+  // This implementation also requires a number of external ASIO files
+  // and a few global variables.  The ASIO callback scheme does not
+  // allow for the passing of user data, so we must create a global
+  // pointer to our callbackInfo structure.
+  //
+  // On unix systems, we make use of a pthread condition variable.
+  // Since there is no equivalent in Windows, I hacked something based
+  // on information found in
+  // http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
 
 #include "asiosys.h"
 #include "asio.h"
@@ -2328,947 +2429,947 @@ bool RtApiJack :: callbackEvent( unsigned long nframes )
 #include "asiodrivers.h"
 #include <cmath>
 
-AsioDrivers drivers;
-ASIOCallbacks asioCallbacks;
-ASIODriverInfo driverInfo;
-CallbackInfo *asioCallbackInfo;
-bool asioXRun;
+  AsioDrivers drivers;
+  ASIOCallbacks asioCallbacks;
+  ASIODriverInfo driverInfo;
+  CallbackInfo *asioCallbackInfo;
+  bool asioXRun;
 
-struct AsioHandle {
-  int drainCounter;       // Tracks callback counts when draining
-  bool internalDrain;     // Indicates if stop is initiated from callback or not.
-  ASIOBufferInfo *bufferInfos;
-  HANDLE condition;
+  struct AsioHandle {
+    int drainCounter;       // Tracks callback counts when draining
+    bool internalDrain;     // Indicates if stop is initiated from callback or not.
+    ASIOBufferInfo *bufferInfos;
+    HANDLE condition;
 
-  AsioHandle()
-    :drainCounter(0), internalDrain(false), bufferInfos(0) {}
-};
+    AsioHandle()
+      :drainCounter(0), internalDrain(false), bufferInfos(0) {}
+  };
 
-// Function declarations (definitions at end of section)
-static const char* getAsioErrorString( ASIOError result );
-void sampleRateChanged( ASIOSampleRate sRate );
-long asioMessages( long selector, long value, void* message, double* opt );
+  // Function declarations (definitions at end of section)
+  static const char* getAsioErrorString( ASIOError result );
+  void sampleRateChanged( ASIOSampleRate sRate );
+  long asioMessages( long selector, long value, void* message, double* opt );
 
-RtApiAsio :: RtApiAsio()
-{
-  // ASIO cannot run on a multi-threaded appartment. You can call
-  // CoInitialize beforehand, but it must be for appartment threading
-  // (in which case, CoInitilialize will return S_FALSE here).
-  coInitialized_ = false;
-  HRESULT hr = CoInitialize( NULL ); 
-  if ( FAILED(hr) ) {
-    errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
-    error( RtError::WARNING );
+  RtApiAsio :: RtApiAsio()
+  {
+    // ASIO cannot run on a multi-threaded appartment. You can call
+    // CoInitialize beforehand, but it must be for appartment threading
+    // (in which case, CoInitilialize will return S_FALSE here).
+    coInitialized_ = false;
+    HRESULT hr = CoInitialize( NULL ); 
+    if ( FAILED(hr) ) {
+      errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
+      error( RtError::WARNING );
+    }
+    coInitialized_ = true;
+
+    drivers.removeCurrentDriver();
+    driverInfo.asioVersion = 2;
+
+    // See note in DirectSound implementation about GetDesktopWindow().
+    driverInfo.sysRef = GetForegroundWindow();
   }
-  coInitialized_ = true;
 
-  drivers.removeCurrentDriver();
-  driverInfo.asioVersion = 2;
+  RtApiAsio :: ~RtApiAsio()
+  {
+    if ( stream_.state != STREAM_CLOSED ) closeStream();
+    if ( coInitialized_ ) CoUninitialize();
+  }
 
-  // See note in DirectSound implementation about GetDesktopWindow().
-  driverInfo.sysRef = GetForegroundWindow();
-}
+  unsigned int RtApiAsio :: getDeviceCount( void )
+  {
+    return (unsigned int) drivers.asioGetNumDev();
+  }
 
-RtApiAsio :: ~RtApiAsio()
-{
-  if ( stream_.state != STREAM_CLOSED ) closeStream();
-  if ( coInitialized_ ) CoUninitialize();
-}
+  RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
+  {
+    RtAudio::DeviceInfo info;
+    info.probed = false;
 
-unsigned int RtApiAsio :: getDeviceCount( void )
-{
-  return (unsigned int) drivers.asioGetNumDev();
-}
+    // Get device ID
+    unsigned int nDevices = getDeviceCount();
+    if ( nDevices == 0 ) {
+      errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
+      error( RtError::INVALID_USE );
+    }
 
-RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
-{
-  RtAudio::DeviceInfo info;
-  info.probed = false;
+    if ( device >= nDevices ) {
+      errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
+      error( RtError::INVALID_USE );
+    }
 
-  // Get device ID
-  unsigned int nDevices = getDeviceCount();
-  if ( nDevices == 0 ) {
-    errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
-    error( RtError::INVALID_USE );
-  }
+    // If a stream is already open, we cannot probe other devices.  Thus, use the saved results.
+    if ( stream_.state != STREAM_CLOSED ) {
+      if ( device >= devices_.size() ) {
+        errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
+        error( RtError::WARNING );
+        return info;
+      }
+      return devices_[ device ];
+    }
 
-  if ( device >= nDevices ) {
-    errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
-    error( RtError::INVALID_USE );
-  }
+    char driverName[32];
+    ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
+    if ( result != ASE_OK ) {
+      errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      return info;
+    }
 
-  // If a stream is already open, we cannot probe other devices.  Thus, use the saved results.
-  if ( stream_.state != STREAM_CLOSED ) {
-    if ( device >= devices_.size() ) {
-      errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
+    info.name = driverName;
+
+    if ( !drivers.loadDriver( driverName ) ) {
+      errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
+      errorText_ = errorStream_.str();
       error( RtError::WARNING );
       return info;
     }
-    return devices_[ device ];
-  }
 
-  char driverName[32];
-  ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
-  if ( result != ASE_OK ) {
-    errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
+    result = ASIOInit( &driverInfo );
+    if ( result != ASE_OK ) {
+      errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      return info;
+    }
 
-  info.name = driverName;
+    // Determine the device channel information.
+    long inputChannels, outputChannels;
+    result = ASIOGetChannels( &inputChannels, &outputChannels );
+    if ( result != ASE_OK ) {
+      drivers.removeCurrentDriver();
+      errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      return info;
+    }
 
-  if ( !drivers.loadDriver( driverName ) ) {
-    errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
+    info.outputChannels = outputChannels;
+    info.inputChannels = inputChannels;
+    if ( info.outputChannels > 0 && info.inputChannels > 0 )
+      info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
 
-  result = ASIOInit( &driverInfo );
-  if ( result != ASE_OK ) {
-    errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
+    // Determine the supported sample rates.
+    info.sampleRates.clear();
+    for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
+      result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
+      if ( result == ASE_OK )
+        info.sampleRates.push_back( SAMPLE_RATES[i] );
+    }
 
-  // Determine the device channel information.
-  long inputChannels, outputChannels;
-  result = ASIOGetChannels( &inputChannels, &outputChannels );
-  if ( result != ASE_OK ) {
-    drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
+    // Determine supported data types ... just check first channel and assume rest are the same.
+    ASIOChannelInfo channelInfo;
+    channelInfo.channel = 0;
+    channelInfo.isInput = true;
+    if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
+    result = ASIOGetChannelInfo( &channelInfo );
+    if ( result != ASE_OK ) {
+      drivers.removeCurrentDriver();
+      errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      return info;
+    }
 
-  info.outputChannels = outputChannels;
-  info.inputChannels = inputChannels;
-  if ( info.outputChannels > 0 && info.inputChannels > 0 )
-    info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+    info.nativeFormats = 0;
+    if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
+      info.nativeFormats |= RTAUDIO_SINT16;
+    else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
+      info.nativeFormats |= RTAUDIO_SINT32;
+    else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
+      info.nativeFormats |= RTAUDIO_FLOAT32;
+    else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
+      info.nativeFormats |= RTAUDIO_FLOAT64;
 
-  // Determine the supported sample rates.
-  info.sampleRates.clear();
-  for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
-    result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
-    if ( result == ASE_OK )
-      info.sampleRates.push_back( SAMPLE_RATES[i] );
-  }
-
-  // Determine supported data types ... just check first channel and assume rest are the same.
-  ASIOChannelInfo channelInfo;
-  channelInfo.channel = 0;
-  channelInfo.isInput = true;
-  if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
-  result = ASIOGetChannelInfo( &channelInfo );
-  if ( result != ASE_OK ) {
+    if ( getDefaultOutputDevice() == device )
+      info.isDefaultOutput = true;
+    if ( getDefaultInputDevice() == device )
+      info.isDefaultInput = true;
+
+    info.probed = true;
     drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
     return info;
   }
 
-  info.nativeFormats = 0;
-  if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
-    info.nativeFormats |= RTAUDIO_SINT16;
-  else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
-    info.nativeFormats |= RTAUDIO_SINT32;
-  else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
-    info.nativeFormats |= RTAUDIO_FLOAT32;
-  else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
-    info.nativeFormats |= RTAUDIO_FLOAT64;
-
-  if ( getDefaultOutputDevice() == device )
-    info.isDefaultOutput = true;
-  if ( getDefaultInputDevice() == device )
-    info.isDefaultInput = true;
-
-  info.probed = true;
-  drivers.removeCurrentDriver();
-  return info;
-}
-
-void bufferSwitch( long index, ASIOBool processNow )
-{
-  RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
-  object->callbackEvent( index );
-}
-
-void RtApiAsio :: saveDeviceInfo( void )
-{
-  devices_.clear();
-
-  unsigned int nDevices = getDeviceCount();
-  devices_.resize( nDevices );
-  for ( unsigned int i=0; i<nDevices; i++ )
-    devices_[i] = getDeviceInfo( i );
-}
-
-bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-                                   unsigned int firstChannel, unsigned int sampleRate,
-                                   RtAudioFormat format, unsigned int *bufferSize,
-                                   RtAudio::StreamOptions *options )
-{
-  // For ASIO, a duplex stream MUST use the same driver.
-  if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) {
-    errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
-    return FAILURE;
+  void bufferSwitch( long index, ASIOBool processNow )
+  {
+    RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
+    object->callbackEvent( index );
   }
 
-  char driverName[32];
-  ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
-  if ( result != ASE_OK ) {
-    errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+  void RtApiAsio :: saveDeviceInfo( void )
+  {
+    devices_.clear();
 
-  // The getDeviceInfo() function will not work when a stream is open
-  // because ASIO does not allow multiple devices to run at the same
-  // time.  Thus, we'll probe the system before opening a stream and
-  // save the results for use by getDeviceInfo().
-  this->saveDeviceInfo();
+    unsigned int nDevices = getDeviceCount();
+    devices_.resize( nDevices );
+    for ( unsigned int i=0; i<nDevices; i++ )
+      devices_[i] = getDeviceInfo( i );
+  }
 
-  // Only load the driver once for duplex stream.
-  if ( mode != INPUT || stream_.mode != OUTPUT ) {
-    if ( !drivers.loadDriver( driverName ) ) {
-      errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
-      errorText_ = errorStream_.str();
+  bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+                                     unsigned int firstChannel, unsigned int sampleRate,
+                                     RtAudioFormat format, unsigned int *bufferSize,
+                                     RtAudio::StreamOptions *options )
+  {
+    // For ASIO, a duplex stream MUST use the same driver.
+    if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) {
+      errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
       return FAILURE;
     }
 
-    result = ASIOInit( &driverInfo );
+    char driverName[32];
+    ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
     if ( result != ASE_OK ) {
-      errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
+      errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
       errorText_ = errorStream_.str();
       return FAILURE;
     }
-  }
 
-  // Check the device channel count.
-  long inputChannels, outputChannels;
-  result = ASIOGetChannels( &inputChannels, &outputChannels );
-  if ( result != ASE_OK ) {
-    drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
-       ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
-    drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-  stream_.nDeviceChannels[mode] = channels;
-  stream_.nUserChannels[mode] = channels;
-  stream_.channelOffset[mode] = firstChannel;
+    // The getDeviceInfo() function will not work when a stream is open
+    // because ASIO does not allow multiple devices to run at the same
+    // time.  Thus, we'll probe the system before opening a stream and
+    // save the results for use by getDeviceInfo().
+    this->saveDeviceInfo();
 
-  // Verify the sample rate is supported.
-  result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
-  if ( result != ASE_OK ) {
-    drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+    // Only load the driver once for duplex stream.
+    if ( mode != INPUT || stream_.mode != OUTPUT ) {
+      if ( !drivers.loadDriver( driverName ) ) {
+        errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
 
-  // Get the current sample rate
-  ASIOSampleRate currentRate;
-  result = ASIOGetSampleRate( &currentRate );
-  if ( result != ASE_OK ) {
-    drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+      result = ASIOInit( &driverInfo );
+      if ( result != ASE_OK ) {
+        errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
+    }
 
-  // Set the sample rate only if necessary
-  if ( currentRate != sampleRate ) {
-    result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
+    // Check the device channel count.
+    long inputChannels, outputChannels;
+    result = ASIOGetChannels( &inputChannels, &outputChannels );
     if ( result != ASE_OK ) {
       drivers.removeCurrentDriver();
-      errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
+      errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
       errorText_ = errorStream_.str();
       return FAILURE;
     }
-  }
 
-  // Determine the driver data type.
-  ASIOChannelInfo channelInfo;
-  channelInfo.channel = 0;
-  if ( mode == OUTPUT ) channelInfo.isInput = false;
-  else channelInfo.isInput = true;
-  result = ASIOGetChannelInfo( &channelInfo );
-  if ( result != ASE_OK ) {
-    drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+    if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
+         ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
+      drivers.removeCurrentDriver();
+      errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+    stream_.nDeviceChannels[mode] = channels;
+    stream_.nUserChannels[mode] = channels;
+    stream_.channelOffset[mode] = firstChannel;
 
-  // Assuming WINDOWS host is always little-endian.
-  stream_.doByteSwap[mode] = false;
-  stream_.userFormat = format;
-  stream_.deviceFormat[mode] = 0;
-  if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
-    stream_.deviceFormat[mode] = RTAUDIO_SINT16;
-    if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
-  }
-  else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
-    stream_.deviceFormat[mode] = RTAUDIO_SINT32;
-    if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
-  }
-  else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
-    stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
-    if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
-  }
-  else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
-    stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
-    if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
-  }
+    // Verify the sample rate is supported.
+    result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
+    if ( result != ASE_OK ) {
+      drivers.removeCurrentDriver();
+      errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
-  if ( stream_.deviceFormat[mode] == 0 ) {
-    drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+    // Get the current sample rate
+    ASIOSampleRate currentRate;
+    result = ASIOGetSampleRate( &currentRate );
+    if ( result != ASE_OK ) {
+      drivers.removeCurrentDriver();
+      errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
-  // Set the buffer size.  For a duplex stream, this will end up
-  // setting the buffer size based on the input constraints, which
-  // should be ok.
-  long minSize, maxSize, preferSize, granularity;
-  result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
-  if ( result != ASE_OK ) {
-    drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+    // Set the sample rate only if necessary
+    if ( currentRate != sampleRate ) {
+      result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
+      if ( result != ASE_OK ) {
+        drivers.removeCurrentDriver();
+        errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
+    }
 
-  if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
-  else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
-  else if ( granularity == -1 ) {
-    // Make sure bufferSize is a power of two.
-	  int log2_of_min_size = 0;
-	  int log2_of_max_size = 0;
+    // Determine the driver data type.
+    ASIOChannelInfo channelInfo;
+    channelInfo.channel = 0;
+    if ( mode == OUTPUT ) channelInfo.isInput = false;
+    else channelInfo.isInput = true;
+    result = ASIOGetChannelInfo( &channelInfo );
+    if ( result != ASE_OK ) {
+      drivers.removeCurrentDriver();
+      errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
-	  for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
-		  if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
-		  if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
-	  }
+    // Assuming WINDOWS host is always little-endian.
+    stream_.doByteSwap[mode] = false;
+    stream_.userFormat = format;
+    stream_.deviceFormat[mode] = 0;
+    if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
+      stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+      if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
+    }
+    else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
+      stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+      if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
+    }
+    else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
+      stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+      if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
+    }
+    else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
+      stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
+      if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
+    }
 
-	  long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
-	  int min_delta_num = log2_of_min_size;
+    if ( stream_.deviceFormat[mode] == 0 ) {
+      drivers.removeCurrentDriver();
+      errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
-	  for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
-		  long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
-		  if (current_delta < min_delta) {
-			  min_delta = current_delta;
-			  min_delta_num = i;
-		  }
-	  }
+    // Set the buffer size.  For a duplex stream, this will end up
+    // setting the buffer size based on the input constraints, which
+    // should be ok.
+    long minSize, maxSize, preferSize, granularity;
+    result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
+    if ( result != ASE_OK ) {
+      drivers.removeCurrentDriver();
+      errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
-	  *bufferSize = ( (unsigned int)1 << min_delta_num );
-	  if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
-	  else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
-  }
-  else if ( granularity != 0 ) {
-    // Set to an even multiple of granularity, rounding up.
-    *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
-  }
+    if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
+    else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
+    else if ( granularity == -1 ) {
+      // Make sure bufferSize is a power of two.
+      int log2_of_min_size = 0;
+      int log2_of_max_size = 0;
 
-  if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) {
-    drivers.removeCurrentDriver();
-    errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
-    return FAILURE;
-  }
+      for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
+        if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
+        if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
+      }
 
-  stream_.bufferSize = *bufferSize;
-  stream_.nBuffers = 2;
+      long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
+      int min_delta_num = log2_of_min_size;
 
-  if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
-  else stream_.userInterleaved = true;
-
-  // ASIO always uses non-interleaved buffers.
-  stream_.deviceInterleaved[mode] = false;
+      for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
+        long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
+        if (current_delta < min_delta) {
+          min_delta = current_delta;
+          min_delta_num = i;
+        }
+      }
 
-  // Allocate, if necessary, our AsioHandle structure for the stream.
-  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-  if ( handle == 0 ) {
-    try {
-      handle = new AsioHandle;
+      *bufferSize = ( (unsigned int)1 << min_delta_num );
+      if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
+      else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
     }
-    catch ( std::bad_alloc& ) {
-      //if ( handle == NULL ) {    
+    else if ( granularity != 0 ) {
+      // Set to an even multiple of granularity, rounding up.
+      *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
+    }
+
+    if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) {
       drivers.removeCurrentDriver();
-      errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
+      errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
       return FAILURE;
     }
-    handle->bufferInfos = 0;
 
-    // Create a manual-reset event.
-    handle->condition = CreateEvent( NULL,   // no security
-                                     TRUE,   // manual-reset
-                                     FALSE,  // non-signaled initially
-                                     NULL ); // unnamed
-    stream_.apiHandle = (void *) handle;
-  }
+    stream_.bufferSize = *bufferSize;
+    stream_.nBuffers = 2;
 
-  // Create the ASIO internal buffers.  Since RtAudio sets up input
-  // and output separately, we'll have to dispose of previously
-  // created output buffers for a duplex stream.
-  long inputLatency, outputLatency;
-  if ( mode == INPUT && stream_.mode == OUTPUT ) {
-    ASIODisposeBuffers();
-    if ( handle->bufferInfos ) free( handle->bufferInfos );
-  }
+    if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+    else stream_.userInterleaved = true;
 
-  // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
-  bool buffersAllocated = false;
-  unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
-  handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
-  if ( handle->bufferInfos == NULL ) {
-    errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
-    errorText_ = errorStream_.str();
-    goto error;
-  }
+    // ASIO always uses non-interleaved buffers.
+    stream_.deviceInterleaved[mode] = false;
 
-  ASIOBufferInfo *infos;
-  infos = handle->bufferInfos;
-  for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
-    infos->isInput = ASIOFalse;
-    infos->channelNum = i + stream_.channelOffset[0];
-    infos->buffers[0] = infos->buffers[1] = 0;
-  }
-  for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
-    infos->isInput = ASIOTrue;
-    infos->channelNum = i + stream_.channelOffset[1];
-    infos->buffers[0] = infos->buffers[1] = 0;
-  }
-
-  // Set up the ASIO callback structure and create the ASIO data buffers.
-  asioCallbacks.bufferSwitch = &bufferSwitch;
-  asioCallbacks.sampleRateDidChange = &sampleRateChanged;
-  asioCallbacks.asioMessage = &asioMessages;
-  asioCallbacks.bufferSwitchTimeInfo = NULL;
-  result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
-  if ( result != ASE_OK ) {
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
-    errorText_ = errorStream_.str();
-    goto error;
-  }
-  buffersAllocated = true;
+    // Allocate, if necessary, our AsioHandle structure for the stream.
+    AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+    if ( handle == 0 ) {
+      try {
+        handle = new AsioHandle;
+      }
+      catch ( std::bad_alloc& ) {
+        //if ( handle == NULL ) {    
+        drivers.removeCurrentDriver();
+        errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
+        return FAILURE;
+      }
+      handle->bufferInfos = 0;
+
+      // Create a manual-reset event.
+      handle->condition = CreateEvent( NULL,   // no security
+                                       TRUE,   // manual-reset
+                                       FALSE,  // non-signaled initially
+                                       NULL ); // unnamed
+      stream_.apiHandle = (void *) handle;
+    }
+
+    // Create the ASIO internal buffers.  Since RtAudio sets up input
+    // and output separately, we'll have to dispose of previously
+    // created output buffers for a duplex stream.
+    long inputLatency, outputLatency;
+    if ( mode == INPUT && stream_.mode == OUTPUT ) {
+      ASIODisposeBuffers();
+      if ( handle->bufferInfos ) free( handle->bufferInfos );
+    }
+
+    // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
+    bool buffersAllocated = false;
+    unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+    handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
+    if ( handle->bufferInfos == NULL ) {
+      errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
+      errorText_ = errorStream_.str();
+      goto error;
+    }
 
-  // Set flags for buffer conversion.
-  stream_.doConvertBuffer[mode] = false;
-  if ( stream_.userFormat != stream_.deviceFormat[mode] )
-    stream_.doConvertBuffer[mode] = true;
-  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
-       stream_.nUserChannels[mode] > 1 )
-    stream_.doConvertBuffer[mode] = true;
+    ASIOBufferInfo *infos;
+    infos = handle->bufferInfos;
+    for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
+      infos->isInput = ASIOFalse;
+      infos->channelNum = i + stream_.channelOffset[0];
+      infos->buffers[0] = infos->buffers[1] = 0;
+    }
+    for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
+      infos->isInput = ASIOTrue;
+      infos->channelNum = i + stream_.channelOffset[1];
+      infos->buffers[0] = infos->buffers[1] = 0;
+    }
 
-  // Allocate necessary internal buffers
-  unsigned long bufferBytes;
-  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
-  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
-  if ( stream_.userBuffer[mode] == NULL ) {
-    errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
-    goto error;
-  }
+    // Set up the ASIO callback structure and create the ASIO data buffers.
+    asioCallbacks.bufferSwitch = &bufferSwitch;
+    asioCallbacks.sampleRateDidChange = &sampleRateChanged;
+    asioCallbacks.asioMessage = &asioMessages;
+    asioCallbacks.bufferSwitchTimeInfo = NULL;
+    result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
+    if ( result != ASE_OK ) {
+      errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
+      errorText_ = errorStream_.str();
+      goto error;
+    }
+    buffersAllocated = true;
+
+    // Set flags for buffer conversion.
+    stream_.doConvertBuffer[mode] = false;
+    if ( stream_.userFormat != stream_.deviceFormat[mode] )
+      stream_.doConvertBuffer[mode] = true;
+    if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+         stream_.nUserChannels[mode] > 1 )
+      stream_.doConvertBuffer[mode] = true;
+
+    // Allocate necessary internal buffers
+    unsigned long bufferBytes;
+    bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+    stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+    if ( stream_.userBuffer[mode] == NULL ) {
+      errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
+      goto error;
+    }
 
-  if ( stream_.doConvertBuffer[mode] ) {
+    if ( stream_.doConvertBuffer[mode] ) {
 
-    bool makeBuffer = true;
-    bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
-    if ( mode == INPUT ) {
-      if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
-        unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
-        if ( bufferBytes <= bytesOut ) makeBuffer = false;
+      bool makeBuffer = true;
+      bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+      if ( mode == INPUT ) {
+        if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+          unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+          if ( bufferBytes <= bytesOut ) makeBuffer = false;
+        }
       }
-    }
 
-    if ( makeBuffer ) {
-      bufferBytes *= *bufferSize;
-      if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
-      stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
-      if ( stream_.deviceBuffer == NULL ) {
-        errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
-        goto error;
+      if ( makeBuffer ) {
+        bufferBytes *= *bufferSize;
+        if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+        stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+        if ( stream_.deviceBuffer == NULL ) {
+          errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
+          goto error;
+        }
       }
     }
-  }
 
-  stream_.sampleRate = sampleRate;
-  stream_.device[mode] = device;
-  stream_.state = STREAM_STOPPED;
-  asioCallbackInfo = &stream_.callbackInfo;
-  stream_.callbackInfo.object = (void *) this;
-  if ( stream_.mode == OUTPUT && mode == INPUT )
-    // We had already set up an output stream.
-    stream_.mode = DUPLEX;
-  else
-    stream_.mode = mode;
-
-  // Determine device latencies
-  result = ASIOGetLatencies( &inputLatency, &outputLatency );
-  if ( result != ASE_OK ) {
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING); // warn but don't fail
-  }
-  else {
-    stream_.latency[0] = outputLatency;
-    stream_.latency[1] = inputLatency;
-  }
+    stream_.sampleRate = sampleRate;
+    stream_.device[mode] = device;
+    stream_.state = STREAM_STOPPED;
+    asioCallbackInfo = &stream_.callbackInfo;
+    stream_.callbackInfo.object = (void *) this;
+    if ( stream_.mode == OUTPUT && mode == INPUT )
+      // We had already set up an output stream.
+      stream_.mode = DUPLEX;
+    else
+      stream_.mode = mode;
 
-  // Setup the buffer conversion information structure.  We don't use
-  // buffers to do channel offsets, so we override that parameter
-  // here.
-  if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
+    // Determine device latencies
+    result = ASIOGetLatencies( &inputLatency, &outputLatency );
+    if ( result != ASE_OK ) {
+      errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING); // warn but don't fail
+    }
+    else {
+      stream_.latency[0] = outputLatency;
+      stream_.latency[1] = inputLatency;
+    }
 
-  return SUCCESS;
+    // Setup the buffer conversion information structure.  We don't use
+    // buffers to do channel offsets, so we override that parameter
+    // here.
+    if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
 
- error:
-  if ( buffersAllocated )
-    ASIODisposeBuffers();
-  drivers.removeCurrentDriver();
+    return SUCCESS;
 
-  if ( handle ) {
-    CloseHandle( handle->condition );
-    if ( handle->bufferInfos )
-      free( handle->bufferInfos );
-    delete handle;
-    stream_.apiHandle = 0;
-  }
+  error:
+    if ( buffersAllocated )
+      ASIODisposeBuffers();
+    drivers.removeCurrentDriver();
 
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
+    if ( handle ) {
+      CloseHandle( handle->condition );
+      if ( handle->bufferInfos )
+        free( handle->bufferInfos );
+      delete handle;
+      stream_.apiHandle = 0;
     }
-  }
 
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
+    for ( int i=0; i<2; i++ ) {
+      if ( stream_.userBuffer[i] ) {
+        free( stream_.userBuffer[i] );
+        stream_.userBuffer[i] = 0;
+      }
+    }
 
-  return FAILURE;
-}
+    if ( stream_.deviceBuffer ) {
+      free( stream_.deviceBuffer );
+      stream_.deviceBuffer = 0;
+    }
 
-void RtApiAsio :: closeStream()
-{
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
-    error( RtError::WARNING );
-    return;
+    return FAILURE;
   }
 
-  if ( stream_.state == STREAM_RUNNING ) {
-    stream_.state = STREAM_STOPPED;
-    ASIOStop();
-  }
-  ASIODisposeBuffers();
-  drivers.removeCurrentDriver();
+  void RtApiAsio :: closeStream()
+  {
+    if ( stream_.state == STREAM_CLOSED ) {
+      errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
+      error( RtError::WARNING );
+      return;
+    }
 
-  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-  if ( handle ) {
-    CloseHandle( handle->condition );
-    if ( handle->bufferInfos )
-      free( handle->bufferInfos );
-    delete handle;
-    stream_.apiHandle = 0;
-  }
+    if ( stream_.state == STREAM_RUNNING ) {
+      stream_.state = STREAM_STOPPED;
+      ASIOStop();
+    }
+    ASIODisposeBuffers();
+    drivers.removeCurrentDriver();
 
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
+    AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+    if ( handle ) {
+      CloseHandle( handle->condition );
+      if ( handle->bufferInfos )
+        free( handle->bufferInfos );
+      delete handle;
+      stream_.apiHandle = 0;
     }
-  }
 
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
+    for ( int i=0; i<2; i++ ) {
+      if ( stream_.userBuffer[i] ) {
+        free( stream_.userBuffer[i] );
+        stream_.userBuffer[i] = 0;
+      }
+    }
 
-  stream_.mode = UNINITIALIZED;
-  stream_.state = STREAM_CLOSED;
-}
+    if ( stream_.deviceBuffer ) {
+      free( stream_.deviceBuffer );
+      stream_.deviceBuffer = 0;
+    }
 
-void RtApiAsio :: startStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_RUNNING ) {
-    errorText_ = "RtApiAsio::startStream(): the stream is already running!";
-    error( RtError::WARNING );
-    return;
+    stream_.mode = UNINITIALIZED;
+    stream_.state = STREAM_CLOSED;
   }
 
-  MUTEX_LOCK( &stream_.mutex );
+  void RtApiAsio :: startStream()
+  {
+    verifyStream();
+    if ( stream_.state == STREAM_RUNNING ) {
+      errorText_ = "RtApiAsio::startStream(): the stream is already running!";
+      error( RtError::WARNING );
+      return;
+    }
 
-  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-  ASIOError result = ASIOStart();
-  if ( result != ASE_OK ) {
-    errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
-    errorText_ = errorStream_.str();
-    goto unlock;
-  }
+    MUTEX_LOCK( &stream_.mutex );
 
-  handle->drainCounter = 0;
-  handle->internalDrain = false;
-  stream_.state = STREAM_RUNNING;
-  asioXRun = false;
+    AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+    ASIOError result = ASIOStart();
+    if ( result != ASE_OK ) {
+      errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
+      errorText_ = errorStream_.str();
+      goto unlock;
+    }
 
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
+    handle->drainCounter = 0;
+    handle->internalDrain = false;
+    stream_.state = STREAM_RUNNING;
+    asioXRun = false;
 
-  if ( result == ASE_OK ) return;
-  error( RtError::SYSTEM_ERROR );
-}
+  unlock:
+    MUTEX_UNLOCK( &stream_.mutex );
 
-void RtApiAsio :: stopStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
+    if ( result == ASE_OK ) return;
+    error( RtError::SYSTEM_ERROR );
   }
 
-  MUTEX_LOCK( &stream_.mutex );
-
-  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-    if ( handle->drainCounter == 0 ) {
-      handle->drainCounter = 1;
-      MUTEX_UNLOCK( &stream_.mutex );
-      WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE );  // block until signaled
-      ResetEvent( handle->condition );
-      MUTEX_LOCK( &stream_.mutex );
+  void RtApiAsio :: stopStream()
+  {
+    verifyStream();
+    if ( stream_.state == STREAM_STOPPED ) {
+      errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
+      error( RtError::WARNING );
+      return;
     }
-  }
 
-  ASIOError result = ASIOStop();
-  if ( result != ASE_OK ) {
-    errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
-    errorText_ = errorStream_.str();
-  }
-
-  stream_.state = STREAM_STOPPED;
-  MUTEX_UNLOCK( &stream_.mutex );
+    MUTEX_LOCK( &stream_.mutex );
 
-  if ( result == ASE_OK ) return;
-  error( RtError::SYSTEM_ERROR );
-}
+    AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+      if ( handle->drainCounter == 0 ) {
+        handle->drainCounter = 1;
+        MUTEX_UNLOCK( &stream_.mutex );
+        WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE );  // block until signaled
+        ResetEvent( handle->condition );
+        MUTEX_LOCK( &stream_.mutex );
+      }
+    }
 
-void RtApiAsio :: abortStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
-  }
+    ASIOError result = ASIOStop();
+    if ( result != ASE_OK ) {
+      errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
+      errorText_ = errorStream_.str();
+    }
 
-  // The following lines were commented-out because some behavior was
-  // noted where the device buffers need to be zeroed to avoid
-  // continuing sound, even when the device buffers are completed
-  // disposed.  So now, calling abort is the same as calling stop.
-  //AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-  //handle->drainCounter = 1;
-  stopStream();
-}
+    stream_.state = STREAM_STOPPED;
+    MUTEX_UNLOCK( &stream_.mutex );
 
-bool RtApiAsio :: callbackEvent( long bufferIndex )
-{
-  if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
-    error( RtError::WARNING );
-    return FAILURE;
+    if ( result == ASE_OK ) return;
+    error( RtError::SYSTEM_ERROR );
   }
 
-  CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
-  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+  void RtApiAsio :: abortStream()
+  {
+    verifyStream();
+    if ( stream_.state == STREAM_STOPPED ) {
+      errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
+      error( RtError::WARNING );
+      return;
+    }
 
-  // Check if we were draining the stream and signal is finished.
-  if ( handle->drainCounter > 3 ) {
-    if ( handle->internalDrain == false )
-      SetEvent( handle->condition );
-    else
-      stopStream();
-    return SUCCESS;
+    // The following lines were commented-out because some behavior was
+    // noted where the device buffers need to be zeroed to avoid
+    // continuing sound, even when the device buffers are completed
+    // disposed.  So now, calling abort is the same as calling stop.
+    //AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+    //handle->drainCounter = 1;
+    stopStream();
   }
 
-  MUTEX_LOCK( &stream_.mutex );
+  bool RtApiAsio :: callbackEvent( long bufferIndex )
+  {
+    if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
+    if ( stream_.state == STREAM_CLOSED ) {
+      errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
+      error( RtError::WARNING );
+      return FAILURE;
+    }
 
-  // The state might change while waiting on a mutex.
-  if ( stream_.state == STREAM_STOPPED ) goto unlock;
+    CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+    AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
 
-  // Invoke user callback to get fresh output data UNLESS we are
-  // draining stream.
-  if ( handle->drainCounter == 0 ) {
-    RtAudioCallback callback = (RtAudioCallback) info->callback;
-    double streamTime = getStreamTime();
-    RtAudioStreamStatus status = 0;
-    if ( stream_.mode != INPUT && asioXRun == true ) {
-      status |= RTAUDIO_OUTPUT_UNDERFLOW;
-      asioXRun = false;
-    }
-    if ( stream_.mode != OUTPUT && asioXRun == true ) {
-      status |= RTAUDIO_INPUT_OVERFLOW;
-      asioXRun = false;
-    }
-    handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
-                                     stream_.bufferSize, streamTime, status, info->userData );
-    if ( handle->drainCounter == 2 ) {
-      MUTEX_UNLOCK( &stream_.mutex );
-      abortStream();
+    // Check if we were draining the stream and signal is finished.
+    if ( handle->drainCounter > 3 ) {
+      if ( handle->internalDrain == false )
+        SetEvent( handle->condition );
+      else
+        stopStream();
       return SUCCESS;
     }
-    else if ( handle->drainCounter == 1 )
-      handle->internalDrain = true;
-  }
-
-  unsigned int nChannels, bufferBytes, i, j;
-  nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
 
-    bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
+    MUTEX_LOCK( &stream_.mutex );
 
-    if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+    // The state might change while waiting on a mutex.
+    if ( stream_.state == STREAM_STOPPED ) goto unlock;
 
-      for ( i=0, j=0; i<nChannels; i++ ) {
-        if ( handle->bufferInfos[i].isInput != ASIOTrue )
-          memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
+    // Invoke user callback to get fresh output data UNLESS we are
+    // draining stream.
+    if ( handle->drainCounter == 0 ) {
+      RtAudioCallback callback = (RtAudioCallback) info->callback;
+      double streamTime = getStreamTime();
+      RtAudioStreamStatus status = 0;
+      if ( stream_.mode != INPUT && asioXRun == true ) {
+        status |= RTAUDIO_OUTPUT_UNDERFLOW;
+        asioXRun = false;
       }
-
+      if ( stream_.mode != OUTPUT && asioXRun == true ) {
+        status |= RTAUDIO_INPUT_OVERFLOW;
+        asioXRun = false;
+      }
+      handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+                                       stream_.bufferSize, streamTime, status, info->userData );
+      if ( handle->drainCounter == 2 ) {
+        MUTEX_UNLOCK( &stream_.mutex );
+        abortStream();
+        return SUCCESS;
+      }
+      else if ( handle->drainCounter == 1 )
+        handle->internalDrain = true;
     }
-    else if ( stream_.doConvertBuffer[0] ) {
 
-      convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
-      if ( stream_.doByteSwap[0] )
-        byteSwapBuffer( stream_.deviceBuffer,
-                        stream_.bufferSize * stream_.nDeviceChannels[0],
-                        stream_.deviceFormat[0] );
+    unsigned int nChannels, bufferBytes, i, j;
+    nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
 
-      for ( i=0, j=0; i<nChannels; i++ ) {
-        if ( handle->bufferInfos[i].isInput != ASIOTrue )
-          memcpy( handle->bufferInfos[i].buffers[bufferIndex],
-                  &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
-      }
+      bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
 
-    }
-    else {
+      if ( handle->drainCounter > 1 ) { // write zeros to the output stream
 
-      if ( stream_.doByteSwap[0] )
-        byteSwapBuffer( stream_.userBuffer[0],
-                        stream_.bufferSize * stream_.nUserChannels[0],
-                        stream_.userFormat );
+        for ( i=0, j=0; i<nChannels; i++ ) {
+          if ( handle->bufferInfos[i].isInput != ASIOTrue )
+            memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
+        }
 
-      for ( i=0, j=0; i<nChannels; i++ ) {
-        if ( handle->bufferInfos[i].isInput != ASIOTrue )
-          memcpy( handle->bufferInfos[i].buffers[bufferIndex],
-                  &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
       }
+      else if ( stream_.doConvertBuffer[0] ) {
+
+        convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+        if ( stream_.doByteSwap[0] )
+          byteSwapBuffer( stream_.deviceBuffer,
+                          stream_.bufferSize * stream_.nDeviceChannels[0],
+                          stream_.deviceFormat[0] );
+
+        for ( i=0, j=0; i<nChannels; i++ ) {
+          if ( handle->bufferInfos[i].isInput != ASIOTrue )
+            memcpy( handle->bufferInfos[i].buffers[bufferIndex],
+                    &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
+        }
 
-    }
-
-    if ( handle->drainCounter ) {
-      handle->drainCounter++;
-      goto unlock;
-    }
-  }
+      }
+      else {
 
-  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+        if ( stream_.doByteSwap[0] )
+          byteSwapBuffer( stream_.userBuffer[0],
+                          stream_.bufferSize * stream_.nUserChannels[0],
+                          stream_.userFormat );
 
-    bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
+        for ( i=0, j=0; i<nChannels; i++ ) {
+          if ( handle->bufferInfos[i].isInput != ASIOTrue )
+            memcpy( handle->bufferInfos[i].buffers[bufferIndex],
+                    &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
+        }
 
-    if (stream_.doConvertBuffer[1]) {
+      }
 
-      // Always interleave ASIO input data.
-      for ( i=0, j=0; i<nChannels; i++ ) {
-        if ( handle->bufferInfos[i].isInput == ASIOTrue )
-          memcpy( &stream_.deviceBuffer[j++*bufferBytes],
-                  handle->bufferInfos[i].buffers[bufferIndex],
-                  bufferBytes );
+      if ( handle->drainCounter ) {
+        handle->drainCounter++;
+        goto unlock;
       }
+    }
 
-      if ( stream_.doByteSwap[1] )
-        byteSwapBuffer( stream_.deviceBuffer,
-                        stream_.bufferSize * stream_.nDeviceChannels[1],
-                        stream_.deviceFormat[1] );
-      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+    if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
 
-    }
-    else {
-      for ( i=0, j=0; i<nChannels; i++ ) {
-        if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
-          memcpy( &stream_.userBuffer[1][bufferBytes*j++],
-                  handle->bufferInfos[i].buffers[bufferIndex],
-                  bufferBytes );
+      bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
+
+      if (stream_.doConvertBuffer[1]) {
+
+        // Always interleave ASIO input data.
+        for ( i=0, j=0; i<nChannels; i++ ) {
+          if ( handle->bufferInfos[i].isInput == ASIOTrue )
+            memcpy( &stream_.deviceBuffer[j++*bufferBytes],
+                    handle->bufferInfos[i].buffers[bufferIndex],
+                    bufferBytes );
         }
+
+        if ( stream_.doByteSwap[1] )
+          byteSwapBuffer( stream_.deviceBuffer,
+                          stream_.bufferSize * stream_.nDeviceChannels[1],
+                          stream_.deviceFormat[1] );
+        convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+
       }
+      else {
+        for ( i=0, j=0; i<nChannels; i++ ) {
+          if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
+            memcpy( &stream_.userBuffer[1][bufferBytes*j++],
+                    handle->bufferInfos[i].buffers[bufferIndex],
+                    bufferBytes );
+          }
+        }
 
-      if ( stream_.doByteSwap[1] )
-        byteSwapBuffer( stream_.userBuffer[1],
-                        stream_.bufferSize * stream_.nUserChannels[1],
-                        stream_.userFormat );
+        if ( stream_.doByteSwap[1] )
+          byteSwapBuffer( stream_.userBuffer[1],
+                          stream_.bufferSize * stream_.nUserChannels[1],
+                          stream_.userFormat );
+      }
     }
-  }
-
- unlock:
-  // The following call was suggested by Malte Clasen.  While the API
-  // documentation indicates it should not be required, some device
-  // drivers apparently do not function correctly without it.
-  ASIOOutputReady();
 
-  MUTEX_UNLOCK( &stream_.mutex );
+  unlock:
+    // The following call was suggested by Malte Clasen.  While the API
+    // documentation indicates it should not be required, some device
+    // drivers apparently do not function correctly without it.
+    ASIOOutputReady();
 
-  RtApi::tickStreamTime();
-  return SUCCESS;
-}
+    MUTEX_UNLOCK( &stream_.mutex );
 
-void sampleRateChanged( ASIOSampleRate sRate )
-{
-  // The ASIO documentation says that this usually only happens during
-  // external sync.  Audio processing is not stopped by the driver,
-  // actual sample rate might not have even changed, maybe only the
-  // sample rate status of an AES/EBU or S/PDIF digital input at the
-  // audio device.
-
-  RtApi *object = (RtApi *) asioCallbackInfo->object;
-  try {
-    object->stopStream();
-  }
-  catch ( RtError &exception ) {
-    std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
-    return;
+    RtApi::tickStreamTime();
+    return SUCCESS;
   }
 
-  std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
-}
+  void sampleRateChanged( ASIOSampleRate sRate )
+  {
+    // The ASIO documentation says that this usually only happens during
+    // external sync.  Audio processing is not stopped by the driver,
+    // actual sample rate might not have even changed, maybe only the
+    // sample rate status of an AES/EBU or S/PDIF digital input at the
+    // audio device.
 
-long asioMessages( long selector, long value, void* message, double* opt )
-{
-  long ret = 0;
-
-  switch( selector ) {
-  case kAsioSelectorSupported:
-    if ( value == kAsioResetRequest
-         || value == kAsioEngineVersion
-         || value == kAsioResyncRequest
-         || value == kAsioLatenciesChanged
-         // The following three were added for ASIO 2.0, you don't
-         // necessarily have to support them.
-         || value == kAsioSupportsTimeInfo
-         || value == kAsioSupportsTimeCode
-         || value == kAsioSupportsInputMonitor)
-      ret = 1L;
-    break;
-  case kAsioResetRequest:
-    // Defer the task and perform the reset of the driver during the
-    // next "safe" situation.  You cannot reset the driver right now,
-    // as this code is called from the driver.  Reset the driver is
-    // done by completely destruct is. I.e. ASIOStop(),
-    // ASIODisposeBuffers(), Destruction Afterwards you initialize the
-    // driver again.
-    std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
-    ret = 1L;
-    break;
-  case kAsioResyncRequest:
-    // This informs the application that the driver encountered some
-    // non-fatal data loss.  It is used for synchronization purposes
-    // of different media.  Added mainly to work around the Win16Mutex
-    // problems in Windows 95/98 with the Windows Multimedia system,
-    // which could lose data because the Mutex was held too long by
-    // another thread.  However a driver can issue it in other
-    // situations, too.
-    // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
-    asioXRun = true;
-    ret = 1L;
-    break;
-  case kAsioLatenciesChanged:
-    // This will inform the host application that the drivers were
-    // latencies changed.  Beware, it this does not mean that the
-    // buffer sizes have changed!  You might need to update internal
-    // delay data.
-    std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
-    ret = 1L;
-    break;
-  case kAsioEngineVersion:
-    // Return the supported ASIO version of the host application.  If
-    // a host application does not implement this selector, ASIO 1.0
-    // is assumed by the driver.
-    ret = 2L;
-    break;
-  case kAsioSupportsTimeInfo:
-    // Informs the driver whether the
-    // asioCallbacks.bufferSwitchTimeInfo() callback is supported.
-    // For compatibility with ASIO 1.0 drivers the host application
-    // should always support the "old" bufferSwitch method, too.
-    ret = 0;
-    break;
-  case kAsioSupportsTimeCode:
-    // Informs the driver whether application is interested in time
-    // code info.  If an application does not need to know about time
-    // code, the driver has less work to do.
-    ret = 0;
-    break;
-  }
-  return ret;
-}
+    RtApi *object = (RtApi *) asioCallbackInfo->object;
+    try {
+      object->stopStream();
+    }
+    catch ( RtError &exception ) {
+      std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
+      return;
+    }
 
-static const char* getAsioErrorString( ASIOError result )
-{
-  struct Messages 
-  {
-    ASIOError value;
-    const char*message;
-  };
+    std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
+  }
 
-  static Messages m[] = 
+  long asioMessages( long selector, long value, void* message, double* opt )
   {
-    {   ASE_NotPresent,    "Hardware input or output is not present or available." },
-    {   ASE_HWMalfunction,  "Hardware is malfunctioning." },
-    {   ASE_InvalidParameter, "Invalid input parameter." },
-    {   ASE_InvalidMode,      "Invalid mode." },
-    {   ASE_SPNotAdvancing,     "Sample position not advancing." },
-    {   ASE_NoClock,            "Sample clock or rate cannot be determined or is not present." },
-    {   ASE_NoMemory,           "Not enough memory to complete the request." }
-  };
-
-  for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
-    if ( m[i].value == result ) return m[i].message;
+    long ret = 0;
+
+    switch( selector ) {
+    case kAsioSelectorSupported:
+      if ( value == kAsioResetRequest
+           || value == kAsioEngineVersion
+           || value == kAsioResyncRequest
+           || value == kAsioLatenciesChanged
+           // The following three were added for ASIO 2.0, you don't
+           // necessarily have to support them.
+           || value == kAsioSupportsTimeInfo
+           || value == kAsioSupportsTimeCode
+           || value == kAsioSupportsInputMonitor)
+        ret = 1L;
+      break;
+    case kAsioResetRequest:
+      // Defer the task and perform the reset of the driver during the
+      // next "safe" situation.  You cannot reset the driver right now,
+      // as this code is called from the driver.  Reset the driver is
+      // done by completely destruct is. I.e. ASIOStop(),
+      // ASIODisposeBuffers(), Destruction Afterwards you initialize the
+      // driver again.
+      std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
+      ret = 1L;
+      break;
+    case kAsioResyncRequest:
+      // This informs the application that the driver encountered some
+      // non-fatal data loss.  It is used for synchronization purposes
+      // of different media.  Added mainly to work around the Win16Mutex
+      // problems in Windows 95/98 with the Windows Multimedia system,
+      // which could lose data because the Mutex was held too long by
+      // another thread.  However a driver can issue it in other
+      // situations, too.
+      // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
+      asioXRun = true;
+      ret = 1L;
+      break;
+    case kAsioLatenciesChanged:
+      // This will inform the host application that the drivers were
+      // latencies changed.  Beware, it this does not mean that the
+      // buffer sizes have changed!  You might need to update internal
+      // delay data.
+      std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
+      ret = 1L;
+      break;
+    case kAsioEngineVersion:
+      // Return the supported ASIO version of the host application.  If
+      // a host application does not implement this selector, ASIO 1.0
+      // is assumed by the driver.
+      ret = 2L;
+      break;
+    case kAsioSupportsTimeInfo:
+      // Informs the driver whether the
+      // asioCallbacks.bufferSwitchTimeInfo() callback is supported.
+      // For compatibility with ASIO 1.0 drivers the host application
+      // should always support the "old" bufferSwitch method, too.
+      ret = 0;
+      break;
+    case kAsioSupportsTimeCode:
+      // Informs the driver whether application is interested in time
+      // code info.  If an application does not need to know about time
+      // code, the driver has less work to do.
+      ret = 0;
+      break;
+    }
+    return ret;
+  }
 
-  return "Unknown error.";
-}
-//******************** End of __WINDOWS_ASIO__ *********************//
+  static const char* getAsioErrorString( ASIOError result )
+  {
+    struct Messages 
+    {
+      ASIOError value;
+      const char*message;
+    };
+
+    static Messages m[] = 
+      {
+        {   ASE_NotPresent,    "Hardware input or output is not present or available." },
+        {   ASE_HWMalfunction,  "Hardware is malfunctioning." },
+        {   ASE_InvalidParameter, "Invalid input parameter." },
+        {   ASE_InvalidMode,      "Invalid mode." },
+        {   ASE_SPNotAdvancing,     "Sample position not advancing." },
+        {   ASE_NoClock,            "Sample clock or rate cannot be determined or is not present." },
+        {   ASE_NoMemory,           "Not enough memory to complete the request." }
+      };
+
+    for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
+      if ( m[i].value == result ) return m[i].message;
+
+    return "Unknown error.";
+  }
+  //******************** End of __WINDOWS_ASIO__ *********************//
 #endif
 
 
 #if defined(__WINDOWS_DS__) // Windows DirectSound API
 
-// Modified by Robin Davies, October 2005
-// - Improvements to DirectX pointer chasing. 
-// - Backdoor RtDsStatistics hook provides DirectX performance information.
-// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
-// - Auto-call CoInitialize for DSOUND and ASIO platforms.
-// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
+  // Modified by Robin Davies, October 2005
+  // - Improvements to DirectX pointer chasing. 
+  // - Backdoor RtDsStatistics hook provides DirectX performance information.
+  // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
+  // - Auto-call CoInitialize for DSOUND and ASIO platforms.
+  // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
 
 #include <dsound.h>
 #include <assert.h>
 
 #if defined(__MINGW32__)
-// missing from latest mingw winapi
+  // missing from latest mingw winapi
 #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
 #define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
 #define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
@@ -3281,1587 +3382,1587 @@ static const char* getAsioErrorString( ASIOError result )
 #pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
 #endif
 
-static inline DWORD dsPointerDifference( DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
-{
-  if ( laterPointer > earlierPointer )
-    return laterPointer - earlierPointer;
-  else
-    return laterPointer - earlierPointer + bufferSize;
-}
-
-static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
-{
-  if ( pointer > bufferSize ) pointer -= bufferSize;
-  if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
-  if ( pointer < earlierPointer ) pointer += bufferSize;
-  return pointer >= earlierPointer && pointer < laterPointer;
-}
-
-// A structure to hold various information related to the DirectSound
-// API implementation.
-struct DsHandle {
-  unsigned int drainCounter; // Tracks callback counts when draining
-  bool internalDrain;        // Indicates if stop is initiated from callback or not.
-  void *id[2];
-  void *buffer[2];
-  bool xrun[2];
-  UINT bufferPointer[2];  
-  DWORD dsBufferSize[2];
-  DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
-  HANDLE condition;
+  static inline DWORD dsPointerDifference( DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
+  {
+    if ( laterPointer > earlierPointer )
+      return laterPointer - earlierPointer;
+    else
+      return laterPointer - earlierPointer + bufferSize;
+  }
 
-  DsHandle()
-    :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; }
-};
+  static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
+  {
+    if ( pointer > bufferSize ) pointer -= bufferSize;
+    if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
+    if ( pointer < earlierPointer ) pointer += bufferSize;
+    return pointer >= earlierPointer && pointer < laterPointer;
+  }
+
+  // A structure to hold various information related to the DirectSound
+  // API implementation.
+  struct DsHandle {
+    unsigned int drainCounter; // Tracks callback counts when draining
+    bool internalDrain;        // Indicates if stop is initiated from callback or not.
+    void *id[2];
+    void *buffer[2];
+    bool xrun[2];
+    UINT bufferPointer[2];  
+    DWORD dsBufferSize[2];
+    DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
+    HANDLE condition;
+
+    DsHandle()
+      :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; }
+  };
 
-/*
-RtApiDs::RtDsStatistics RtApiDs::statistics;
+  /*
+    RtApiDs::RtDsStatistics RtApiDs::statistics;
 
-// Provides a backdoor hook to monitor for DirectSound read overruns and write underruns.
-RtApiDs::RtDsStatistics RtApiDs::getDsStatistics()
-{
-  RtDsStatistics s = statistics;
+    // Provides a backdoor hook to monitor for DirectSound read overruns and write underruns.
+    RtApiDs::RtDsStatistics RtApiDs::getDsStatistics()
+    {
+    RtDsStatistics s = statistics;
 
-  // update the calculated fields.
-  if ( s.inputFrameSize != 0 )
+    // update the calculated fields.
+    if ( s.inputFrameSize != 0 )
     s.latency += s.readDeviceSafeLeadBytes * 1.0 / s.inputFrameSize / s.sampleRate;
 
-  if ( s.outputFrameSize != 0 )
+    if ( s.outputFrameSize != 0 )
     s.latency += (s.writeDeviceSafeLeadBytes + s.writeDeviceBufferLeadBytes) * 1.0 / s.outputFrameSize / s.sampleRate;
 
-  return s;
-}
-*/
-
-// Declarations for utility functions, callbacks, and structures
-// specific to the DirectSound implementation.
-static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
-                                          LPCTSTR description,
-                                          LPCTSTR module,
-                                          LPVOID lpContext );
-
-static char* getErrorString( int code );
-
-extern "C" unsigned __stdcall callbackHandler( void *ptr );
-
-struct EnumInfo {
-  bool isInput;
-  bool getDefault;
-  bool findIndex;
-  unsigned int counter;
-  unsigned int index;
-  LPGUID id;
-  std::string name;
-
-  EnumInfo()
-    : isInput(false), getDefault(false), findIndex(false), counter(0), index(0) {}
-};
-
-RtApiDs :: RtApiDs()
-{
-  // Dsound will run both-threaded. If CoInitialize fails, then just
-  // accept whatever the mainline chose for a threading model.
-  coInitialized_ = false;
-  HRESULT hr = CoInitialize( NULL );
-  if ( !FAILED( hr ) ) coInitialized_ = true;
-}
-
-RtApiDs :: ~RtApiDs()
-{
-  if ( coInitialized_ ) CoUninitialize(); // balanced call.
-  if ( stream_.state != STREAM_CLOSED ) closeStream();
-}
-
-unsigned int RtApiDs :: getDefaultInputDevice( void )
-{
-  // Count output devices.
-  EnumInfo info;
-  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
-  if ( FAILED( result ) ) {
-    errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") counting output devices!";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return 0;
-  }
-
-  // Now enumerate input devices until we find the id = NULL.
-  info.isInput = true;
-  info.getDefault = true;
-  result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
-  if ( FAILED( result ) ) {
-    errorStream_ << "RtApiDs::getDefaultInputDevice: error (" << getErrorString( result ) << ") enumerating input devices!";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return 0;
-  }
-
-  if ( info.counter > 0 ) return info.counter - 1;
-  return 0;
-}
-
-unsigned int RtApiDs :: getDefaultOutputDevice( void )
-{
-  // Enumerate output devices until we find the id = NULL.
-  EnumInfo info;
-  info.getDefault = true;
-  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
-  if ( FAILED( result ) ) {
-    errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") enumerating output devices!";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return 0;
-  }
-
-  if ( info.counter > 0 ) return info.counter - 1;
-  return 0;
-}
-
-unsigned int RtApiDs :: getDeviceCount( void )
-{
-  // Count DirectSound devices.
-  EnumInfo info;
-  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
-  if ( FAILED( result ) ) {
-    errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-  }
+    return s;
+    }
+  */
 
-  // Count DirectSoundCapture devices.
-  info.isInput = true;
-  result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
-  if ( FAILED( result ) ) {
-    errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-  }
+  // Declarations for utility functions, callbacks, and structures
+  // specific to the DirectSound implementation.
+  static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
+                                            LPCTSTR description,
+                                            LPCTSTR module,
+                                            LPVOID lpContext );
 
-  return info.counter;
-}
+  static char* getErrorString( int code );
 
-RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
-{
-  // Because DirectSound always enumerates input and output devices
-  // separately (and because we don't attempt to combine devices
-  // internally), none of our "devices" will ever be duplex.
+  extern "C" unsigned __stdcall callbackHandler( void *ptr );
 
-  RtAudio::DeviceInfo info;
-  info.probed = false;
-
-  // Enumerate through devices to find the id (if it exists).  Note
-  // that we have to do the output enumeration first, even if this is
-  // an input device, in order for the device counter to be correct.
-  EnumInfo dsinfo;
-  dsinfo.findIndex = true;
-  dsinfo.index = device;
-  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
-  if ( FAILED( result ) ) {
-    errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating output devices!";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-  }
+  struct EnumInfo {
+    bool isInput;
+    bool getDefault;
+    bool findIndex;
+    unsigned int counter;
+    unsigned int index;
+    LPGUID id;
+    std::string name;
 
-  if ( dsinfo.name.empty() ) goto probeInput;
-
-  LPDIRECTSOUND output;
-  DSCAPS outCaps;
-  result = DirectSoundCreate( dsinfo.id, &output, NULL );
-  if ( FAILED( result ) ) {
-    errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
-
-  outCaps.dwSize = sizeof( outCaps );
-  result = output->GetCaps( &outCaps );
-  if ( FAILED( result ) ) {
-    output->Release();
-    errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
-
-  // Get output channel information.
-  info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
+    EnumInfo()
+      : isInput(false), getDefault(false), findIndex(false), counter(0), index(0) {}
+  };
 
-  // Get sample rate information.
-  info.sampleRates.clear();
-  for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
-    if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
-         SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate )
-      info.sampleRates.push_back( SAMPLE_RATES[k] );
+  RtApiDs :: RtApiDs()
+  {
+    // Dsound will run both-threaded. If CoInitialize fails, then just
+    // accept whatever the mainline chose for a threading model.
+    coInitialized_ = false;
+    HRESULT hr = CoInitialize( NULL );
+    if ( !FAILED( hr ) ) coInitialized_ = true;
   }
 
-  // Get format information.
-  if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
-  if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
-
-  output->Release();
-
-  if ( getDefaultOutputDevice() == device )
-    info.isDefaultOutput = true;
-
-  // Copy name and return.
-  info.name = dsinfo.name;
-
-  info.probed = true;
-  return info;
-
- probeInput:
-
-  dsinfo.isInput = true;
-  result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
-  if ( FAILED( result ) ) {
-    errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating input devices!";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
+  RtApiDs :: ~RtApiDs()
+  {
+    if ( coInitialized_ ) CoUninitialize(); // balanced call.
+    if ( stream_.state != STREAM_CLOSED ) closeStream();
   }
 
-  if ( dsinfo.name.empty() ) return info;
+  unsigned int RtApiDs :: getDefaultInputDevice( void )
+  {
+    // Count output devices.
+    EnumInfo info;
+    HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
+    if ( FAILED( result ) ) {
+      errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") counting output devices!";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      return 0;
+    }
 
-  LPDIRECTSOUNDCAPTURE input;
-  result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL );
-  if ( FAILED( result ) ) {
-    errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
+    // Now enumerate input devices until we find the id = NULL.
+    info.isInput = true;
+    info.getDefault = true;
+    result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
+    if ( FAILED( result ) ) {
+      errorStream_ << "RtApiDs::getDefaultInputDevice: error (" << getErrorString( result ) << ") enumerating input devices!";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      return 0;
+    }
 
-  DSCCAPS inCaps;
-  inCaps.dwSize = sizeof( inCaps );
-  result = input->GetCaps( &inCaps );
-  if ( FAILED( result ) ) {
-    input->Release();
-    errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsinfo.name << ")!";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
+    if ( info.counter > 0 ) return info.counter - 1;
+    return 0;
   }
 
-  // Get input channel information.
-  info.inputChannels = inCaps.dwChannels;
-
-  // Get sample rate and format information.
-  if ( inCaps.dwChannels == 2 ) {
-    if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
-    if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
-    if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
-    if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
-    if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
-    if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
-    if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
-    if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
-
-    if ( info.nativeFormats & RTAUDIO_SINT16 ) {
-      if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.sampleRates.push_back( 11025 );
-      if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.sampleRates.push_back( 22050 );
-      if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.sampleRates.push_back( 44100 );
-      if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.sampleRates.push_back( 96000 );
-    }
-    else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
-      if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.sampleRates.push_back( 11025 );
-      if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.sampleRates.push_back( 22050 );
-      if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.sampleRates.push_back( 44100 );
-      if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.sampleRates.push_back( 44100 );
-    }
-  }
-  else if ( inCaps.dwChannels == 1 ) {
-    if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
-    if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
-    if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
-    if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
-    if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
-    if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
-    if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
-    if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
-
-    if ( info.nativeFormats & RTAUDIO_SINT16 ) {
-      if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.sampleRates.push_back( 11025 );
-      if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.sampleRates.push_back( 22050 );
-      if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.sampleRates.push_back( 44100 );
-      if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.sampleRates.push_back( 96000 );
-    }
-    else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
-      if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.sampleRates.push_back( 11025 );
-      if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.sampleRates.push_back( 22050 );
-      if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.sampleRates.push_back( 44100 );
-      if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.sampleRates.push_back( 96000 );
-    }
-  }
-  else info.inputChannels = 0; // technically, this would be an error
-
-  input->Release();
-
-  if ( info.inputChannels == 0 ) return info;
-
-  if ( getDefaultInputDevice() == device )
-    info.isDefaultInput = true;
-
-  // Copy name and return.
-  info.name = dsinfo.name;
-  info.probed = true;
-  return info;
-}
-
-bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-                                 unsigned int firstChannel, unsigned int sampleRate,
-                                 RtAudioFormat format, unsigned int *bufferSize,
-                                 RtAudio::StreamOptions *options )
-{
-  if ( channels + firstChannel > 2 ) {
-    errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
-    return FAILURE;
-  }
+  unsigned int RtApiDs :: getDefaultOutputDevice( void )
+  {
+    // Enumerate output devices until we find the id = NULL.
+    EnumInfo info;
+    info.getDefault = true;
+    HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
+    if ( FAILED( result ) ) {
+      errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") enumerating output devices!";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      return 0;
+    }
 
-  // Enumerate through devices to find the id (if it exists).  Note
-  // that we have to do the output enumeration first, even if this is
-  // an input device, in order for the device counter to be correct.
-  EnumInfo dsinfo;
-  dsinfo.findIndex = true;
-  dsinfo.index = device;
-  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
-  if ( FAILED( result ) ) {
-    errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating output devices!";
-    errorText_ = errorStream_.str();
-    return FAILURE;
+    if ( info.counter > 0 ) return info.counter - 1;
+    return 0;
   }
 
-  if ( mode == OUTPUT ) {
-    if ( dsinfo.name.empty() ) {
-      errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
+  unsigned int RtApiDs :: getDeviceCount( void )
+  {
+    // Count DirectSound devices.
+    EnumInfo info;
+    HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
+    if ( FAILED( result ) ) {
+      errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
       errorText_ = errorStream_.str();
-      return FAILURE;
+      error( RtError::WARNING );
     }
-  }
-  else { // mode == INPUT
-    dsinfo.isInput = true;
-    HRESULT result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
+
+    // Count DirectSoundCapture devices.
+    info.isInput = true;
+    result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
     if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating input devices!";
+      errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
       errorText_ = errorStream_.str();
-      return FAILURE;
+      error( RtError::WARNING );
     }
-    if ( dsinfo.name.empty() ) {
-      errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
+
+    return info.counter;
+  }
+
+  RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
+  {
+    // Because DirectSound always enumerates input and output devices
+    // separately (and because we don't attempt to combine devices
+    // internally), none of our "devices" will ever be duplex.
+
+    RtAudio::DeviceInfo info;
+    info.probed = false;
+
+    // Enumerate through devices to find the id (if it exists).  Note
+    // that we have to do the output enumeration first, even if this is
+    // an input device, in order for the device counter to be correct.
+    EnumInfo dsinfo;
+    dsinfo.findIndex = true;
+    dsinfo.index = device;
+    HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
+    if ( FAILED( result ) ) {
+      errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating output devices!";
       errorText_ = errorStream_.str();
-      return FAILURE;
+      error( RtError::WARNING );
     }
-  }
 
-  // According to a note in PortAudio, using GetDesktopWindow()
-  // instead of GetForegroundWindow() is supposed to avoid problems
-  // that occur when the application's window is not the foreground
-  // window.  Also, if the application window closes before the
-  // DirectSound buffer, DirectSound can crash.  However, for console
-  // applications, no sound was produced when using GetDesktopWindow().
-  HWND hWnd = GetForegroundWindow();
-
-  // Check the numberOfBuffers parameter and limit the lowest value to
-  // two.  This is a judgement call and a value of two is probably too
-  // low for capture, but it should work for playback.
-  int nBuffers = 0;
-  if ( options ) nBuffers = options->numberOfBuffers;
-  if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
-  if ( nBuffers < 2 ) nBuffers = 3;
-
-  // Create the wave format structure.  The data format setting will
-  // be determined later.
-  WAVEFORMATEX waveFormat;
-  ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
-  waveFormat.wFormatTag = WAVE_FORMAT_PCM;
-  waveFormat.nChannels = channels + firstChannel;
-  waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
-
-  // Determine the device buffer size. By default, 32k, but we will
-  // grow it to make allowances for very large software buffer sizes.
-  DWORD dsBufferSize = 0;
-  DWORD dsPointerLeadTime = 0;
-  long bufferBytes = MINIMUM_DEVICE_BUFFER_SIZE; // sound cards will always *knock wood* support this
-
-  void *ohandle = 0, *bhandle = 0;
-  if ( mode == OUTPUT ) {
+    if ( dsinfo.name.empty() ) goto probeInput;
 
     LPDIRECTSOUND output;
+    DSCAPS outCaps;
     result = DirectSoundCreate( dsinfo.id, &output, NULL );
     if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!";
+      errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!";
       errorText_ = errorStream_.str();
-      return FAILURE;
+      error( RtError::WARNING );
+      return info;
     }
 
-    DSCAPS outCaps;
     outCaps.dwSize = sizeof( outCaps );
     result = output->GetCaps( &outCaps );
     if ( FAILED( result ) ) {
       output->Release();
-      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsinfo.name << ")!";
+      errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
       errorText_ = errorStream_.str();
-      return FAILURE;
+      error( RtError::WARNING );
+      return info;
     }
 
-    // Check channel information.
-    if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
-      errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsinfo.name << ") does not support stereo playback.";
-      errorText_ = errorStream_.str();
-      return FAILURE;
-    }
+    // Get output channel information.
+    info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
 
-    // Check format information.  Use 16-bit format unless not
-    // supported or user requests 8-bit.
-    if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
-         !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
-      waveFormat.wBitsPerSample = 16;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT16;
-    }
-    else {
-      waveFormat.wBitsPerSample = 8;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+    // Get sample rate information.
+    info.sampleRates.clear();
+    for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+      if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
+           SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate )
+        info.sampleRates.push_back( SAMPLE_RATES[k] );
     }
-    stream_.userFormat = format;
 
-    // Update wave format structure and buffer information.
-    waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
-    waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
-    dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
+    // Get format information.
+    if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
+    if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
+
+    output->Release();
 
-    // If the user wants an even bigger buffer, increase the device buffer size accordingly.
-    while ( dsPointerLeadTime * 2U > (DWORD) bufferBytes )
-      bufferBytes *= 2;
+    if ( getDefaultOutputDevice() == device )
+      info.isDefaultOutput = true;
 
-    // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
-    //result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
-    // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
-    result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
+    // Copy name and return.
+    info.name = dsinfo.name;
+
+    info.probed = true;
+    return info;
+
+  probeInput:
+
+    dsinfo.isInput = true;
+    result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
     if ( FAILED( result ) ) {
-      output->Release();
-      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsinfo.name << ")!";
+      errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating input devices!";
       errorText_ = errorStream_.str();
-      return FAILURE;
+      error( RtError::WARNING );
     }
 
-    // Even though we will write to the secondary buffer, we need to
-    // access the primary buffer to set the correct output format
-    // (since the default is 8-bit, 22 kHz!).  Setup the DS primary
-    // buffer description.
-    DSBUFFERDESC bufferDescription;
-    ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
-    bufferDescription.dwSize = sizeof( DSBUFFERDESC );
-    bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
+    if ( dsinfo.name.empty() ) return info;
 
-    // Obtain the primary buffer
-    LPDIRECTSOUNDBUFFER buffer;
-    result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+    LPDIRECTSOUNDCAPTURE input;
+    result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL );
     if ( FAILED( result ) ) {
-      output->Release();
-      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsinfo.name << ")!";
+      errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!";
       errorText_ = errorStream_.str();
-      return FAILURE;
+      error( RtError::WARNING );
+      return info;
     }
 
-    // Set the primary DS buffer sound format.
-    result = buffer->SetFormat( &waveFormat );
+    DSCCAPS inCaps;
+    inCaps.dwSize = sizeof( inCaps );
+    result = input->GetCaps( &inCaps );
     if ( FAILED( result ) ) {
-      output->Release();
-      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsinfo.name << ")!";
+      input->Release();
+      errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsinfo.name << ")!";
       errorText_ = errorStream_.str();
-      return FAILURE;
+      error( RtError::WARNING );
+      return info;
     }
 
-    // Setup the secondary DS buffer description.
-    dsBufferSize = (DWORD) bufferBytes;
-    ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
-    bufferDescription.dwSize = sizeof( DSBUFFERDESC );
-    bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
-                                  DSBCAPS_GLOBALFOCUS |
-                                  DSBCAPS_GETCURRENTPOSITION2 |
-                                  DSBCAPS_LOCHARDWARE );  // Force hardware mixing
-    bufferDescription.dwBufferBytes = bufferBytes;
-    bufferDescription.lpwfxFormat = &waveFormat;
-
-    // Try to create the secondary DS buffer.  If that doesn't work,
-    // try to use software mixing.  Otherwise, there's a problem.
-    result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
-    if ( FAILED( result ) ) {
-      bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
-                                    DSBCAPS_GLOBALFOCUS |
-                                    DSBCAPS_GETCURRENTPOSITION2 |
-                                    DSBCAPS_LOCSOFTWARE );  // Force software mixing
-      result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
-      if ( FAILED( result ) ) {
-        output->Release();
-        errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsinfo.name << ")!";
-        errorText_ = errorStream_.str();
-        return FAILURE;
+    // Get input channel information.
+    info.inputChannels = inCaps.dwChannels;
+
+    // Get sample rate and format information.
+    if ( inCaps.dwChannels == 2 ) {
+      if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+      if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+      if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+      if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+      if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+      if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+      if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+      if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+
+      if ( info.nativeFormats & RTAUDIO_SINT16 ) {
+        if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.sampleRates.push_back( 11025 );
+        if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.sampleRates.push_back( 22050 );
+        if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.sampleRates.push_back( 44100 );
+        if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.sampleRates.push_back( 96000 );
+      }
+      else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
+        if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.sampleRates.push_back( 11025 );
+        if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.sampleRates.push_back( 22050 );
+        if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.sampleRates.push_back( 44100 );
+        if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.sampleRates.push_back( 44100 );
       }
     }
-
-    // Get the buffer size ... might be different from what we specified.
-    DSBCAPS dsbcaps;
-    dsbcaps.dwSize = sizeof( DSBCAPS );
-    result = buffer->GetCaps( &dsbcaps );
-    if ( FAILED( result ) ) {
-      output->Release();
-      buffer->Release();
-      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsinfo.name << ")!";
-      errorText_ = errorStream_.str();
-      return FAILURE;
+    else if ( inCaps.dwChannels == 1 ) {
+      if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+      if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+      if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+      if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+      if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+      if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+      if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+      if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+
+      if ( info.nativeFormats & RTAUDIO_SINT16 ) {
+        if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.sampleRates.push_back( 11025 );
+        if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.sampleRates.push_back( 22050 );
+        if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.sampleRates.push_back( 44100 );
+        if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.sampleRates.push_back( 96000 );
+      }
+      else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
+        if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.sampleRates.push_back( 11025 );
+        if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.sampleRates.push_back( 22050 );
+        if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.sampleRates.push_back( 44100 );
+        if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.sampleRates.push_back( 96000 );
+      }
     }
+    else info.inputChannels = 0; // technically, this would be an error
 
-    bufferBytes = dsbcaps.dwBufferBytes;
-
-    // Lock the DS buffer
-    LPVOID audioPtr;
-    DWORD dataLen;
-    result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 );
-    if ( FAILED( result ) ) {
-      output->Release();
-      buffer->Release();
-      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsinfo.name << ")!";
-      errorText_ = errorStream_.str();
-      return FAILURE;
-    }
+    input->Release();
 
-    // Zero the DS buffer
-    ZeroMemory( audioPtr, dataLen );
+    if ( info.inputChannels == 0 ) return info;
 
-    // Unlock the DS buffer
-    result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
-    if ( FAILED( result ) ) {
-      output->Release();
-      buffer->Release();
-      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsinfo.name << ")!";
-      errorText_ = errorStream_.str();
-      return FAILURE;
-    }
+    if ( getDefaultInputDevice() == device )
+      info.isDefaultInput = true;
 
-    dsBufferSize = bufferBytes;
-    ohandle = (void *) output;
-    bhandle = (void *) buffer;
+    // Copy name and return.
+    info.name = dsinfo.name;
+    info.probed = true;
+    return info;
   }
 
-  if ( mode == INPUT ) {
-
-    LPDIRECTSOUNDCAPTURE input;
-    result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!";
-      errorText_ = errorStream_.str();
+  bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+                                   unsigned int firstChannel, unsigned int sampleRate,
+                                   RtAudioFormat format, unsigned int *bufferSize,
+                                   RtAudio::StreamOptions *options )
+  {
+    if ( channels + firstChannel > 2 ) {
+      errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
       return FAILURE;
     }
 
-    DSCCAPS inCaps;
-    inCaps.dwSize = sizeof( inCaps );
-    result = input->GetCaps( &inCaps );
+    // Enumerate through devices to find the id (if it exists).  Note
+    // that we have to do the output enumeration first, even if this is
+    // an input device, in order for the device counter to be correct.
+    EnumInfo dsinfo;
+    dsinfo.findIndex = true;
+    dsinfo.index = device;
+    HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
     if ( FAILED( result ) ) {
-      input->Release();
-      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsinfo.name << ")!";
+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating output devices!";
       errorText_ = errorStream_.str();
       return FAILURE;
     }
 
-    // Check channel information.
-    if ( inCaps.dwChannels < channels + firstChannel ) {
-      errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
-      return FAILURE;
+    if ( mode == OUTPUT ) {
+      if ( dsinfo.name.empty() ) {
+        errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
+    }
+    else { // mode == INPUT
+      dsinfo.isInput = true;
+      HRESULT result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating input devices!";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
+      if ( dsinfo.name.empty() ) {
+        errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
     }
 
-    // Check format information.  Use 16-bit format unless user
-    // requests 8-bit.
-    DWORD deviceFormats;
-    if ( channels + firstChannel == 2 ) {
-      deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
-      if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
-        waveFormat.wBitsPerSample = 8;
-        stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+    // According to a note in PortAudio, using GetDesktopWindow()
+    // instead of GetForegroundWindow() is supposed to avoid problems
+    // that occur when the application's window is not the foreground
+    // window.  Also, if the application window closes before the
+    // DirectSound buffer, DirectSound can crash.  However, for console
+    // applications, no sound was produced when using GetDesktopWindow().
+    HWND hWnd = GetForegroundWindow();
+
+    // Check the numberOfBuffers parameter and limit the lowest value to
+    // two.  This is a judgement call and a value of two is probably too
+    // low for capture, but it should work for playback.
+    int nBuffers = 0;
+    if ( options ) nBuffers = options->numberOfBuffers;
+    if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
+    if ( nBuffers < 2 ) nBuffers = 3;
+
+    // Create the wave format structure.  The data format setting will
+    // be determined later.
+    WAVEFORMATEX waveFormat;
+    ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
+    waveFormat.wFormatTag = WAVE_FORMAT_PCM;
+    waveFormat.nChannels = channels + firstChannel;
+    waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
+
+    // Determine the device buffer size. By default, 32k, but we will
+    // grow it to make allowances for very large software buffer sizes.
+    DWORD dsBufferSize = 0;
+    DWORD dsPointerLeadTime = 0;
+    long bufferBytes = MINIMUM_DEVICE_BUFFER_SIZE; // sound cards will always *knock wood* support this
+
+    void *ohandle = 0, *bhandle = 0;
+    if ( mode == OUTPUT ) {
+
+      LPDIRECTSOUND output;
+      result = DirectSoundCreate( dsinfo.id, &output, NULL );
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!";
+        errorText_ = errorStream_.str();
+        return FAILURE;
       }
-      else { // assume 16-bit is supported
+
+      DSCAPS outCaps;
+      outCaps.dwSize = sizeof( outCaps );
+      result = output->GetCaps( &outCaps );
+      if ( FAILED( result ) ) {
+        output->Release();
+        errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsinfo.name << ")!";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
+
+      // Check channel information.
+      if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
+        errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsinfo.name << ") does not support stereo playback.";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
+
+      // Check format information.  Use 16-bit format unless not
+      // supported or user requests 8-bit.
+      if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
+           !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
         waveFormat.wBitsPerSample = 16;
         stream_.deviceFormat[mode] = RTAUDIO_SINT16;
       }
-    }
-    else { // channel == 1
-      deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
-      if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
+      else {
         waveFormat.wBitsPerSample = 8;
         stream_.deviceFormat[mode] = RTAUDIO_SINT8;
       }
-      else { // assume 16-bit is supported
-        waveFormat.wBitsPerSample = 16;
-        stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+      stream_.userFormat = format;
+
+      // Update wave format structure and buffer information.
+      waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+      waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+      dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
+
+      // If the user wants an even bigger buffer, increase the device buffer size accordingly.
+      while ( dsPointerLeadTime * 2U > (DWORD) bufferBytes )
+        bufferBytes *= 2;
+
+      // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
+      //result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
+      // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
+      result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
+      if ( FAILED( result ) ) {
+        output->Release();
+        errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsinfo.name << ")!";
+        errorText_ = errorStream_.str();
+        return FAILURE;
       }
-    }
-    stream_.userFormat = format;
 
-    // Update wave format structure and buffer information.
-    waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
-    waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
-
-    // Setup the secondary DS buffer description.
-    dsBufferSize = bufferBytes;
-    DSCBUFFERDESC bufferDescription;
-    ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
-    bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
-    bufferDescription.dwFlags = 0;
-    bufferDescription.dwReserved = 0;
-    bufferDescription.dwBufferBytes = bufferBytes;
-    bufferDescription.lpwfxFormat = &waveFormat;
-
-    // Create the capture buffer.
-    LPDIRECTSOUNDCAPTUREBUFFER buffer;
-    result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
-    if ( FAILED( result ) ) {
-      input->Release();
-      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsinfo.name << ")!";
-      errorText_ = errorStream_.str();
-      return FAILURE;
-    }
+      // Even though we will write to the secondary buffer, we need to
+      // access the primary buffer to set the correct output format
+      // (since the default is 8-bit, 22 kHz!).  Setup the DS primary
+      // buffer description.
+      DSBUFFERDESC bufferDescription;
+      ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
+      bufferDescription.dwSize = sizeof( DSBUFFERDESC );
+      bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
+
+      // Obtain the primary buffer
+      LPDIRECTSOUNDBUFFER buffer;
+      result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+      if ( FAILED( result ) ) {
+        output->Release();
+        errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsinfo.name << ")!";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
 
-    // Lock the capture buffer
-    LPVOID audioPtr;
-    DWORD dataLen;
-    result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 );
-    if ( FAILED( result ) ) {
-      input->Release();
-      buffer->Release();
-      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsinfo.name << ")!";
-      errorText_ = errorStream_.str();
-      return FAILURE;
-    }
+      // Set the primary DS buffer sound format.
+      result = buffer->SetFormat( &waveFormat );
+      if ( FAILED( result ) ) {
+        output->Release();
+        errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsinfo.name << ")!";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
 
-    // Zero the buffer
-    ZeroMemory( audioPtr, dataLen );
+      // Setup the secondary DS buffer description.
+      dsBufferSize = (DWORD) bufferBytes;
+      ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
+      bufferDescription.dwSize = sizeof( DSBUFFERDESC );
+      bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
+                                    DSBCAPS_GLOBALFOCUS |
+                                    DSBCAPS_GETCURRENTPOSITION2 |
+                                    DSBCAPS_LOCHARDWARE );  // Force hardware mixing
+      bufferDescription.dwBufferBytes = bufferBytes;
+      bufferDescription.lpwfxFormat = &waveFormat;
 
-    // Unlock the buffer
-    result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
-    if ( FAILED( result ) ) {
-      input->Release();
-      buffer->Release();
-      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsinfo.name << ")!";
-      errorText_ = errorStream_.str();
-      return FAILURE;
-    }
+      // Try to create the secondary DS buffer.  If that doesn't work,
+      // try to use software mixing.  Otherwise, there's a problem.
+      result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+      if ( FAILED( result ) ) {
+        bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
+                                      DSBCAPS_GLOBALFOCUS |
+                                      DSBCAPS_GETCURRENTPOSITION2 |
+                                      DSBCAPS_LOCSOFTWARE );  // Force software mixing
+        result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+        if ( FAILED( result ) ) {
+          output->Release();
+          errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsinfo.name << ")!";
+          errorText_ = errorStream_.str();
+          return FAILURE;
+        }
+      }
 
-    dsBufferSize = bufferBytes;
-    ohandle = (void *) input;
-    bhandle = (void *) buffer;
-  }
+      // Get the buffer size ... might be different from what we specified.
+      DSBCAPS dsbcaps;
+      dsbcaps.dwSize = sizeof( DSBCAPS );
+      result = buffer->GetCaps( &dsbcaps );
+      if ( FAILED( result ) ) {
+        output->Release();
+        buffer->Release();
+        errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsinfo.name << ")!";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
 
-  // Set various stream parameters
-  DsHandle *handle = 0;
-  stream_.nDeviceChannels[mode] = channels + firstChannel;
-  stream_.nUserChannels[mode] = channels;
-  stream_.bufferSize = *bufferSize;
-  stream_.channelOffset[mode] = firstChannel;
-  stream_.deviceInterleaved[mode] = true;
-  if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
-  else stream_.userInterleaved = true;
+      bufferBytes = dsbcaps.dwBufferBytes;
 
-  // Set flag for buffer conversion
-  stream_.doConvertBuffer[mode] = false;
-  if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
-    stream_.doConvertBuffer[mode] = true;
-  if (stream_.userFormat != stream_.deviceFormat[mode])
-    stream_.doConvertBuffer[mode] = true;
-  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
-       stream_.nUserChannels[mode] > 1 )
-    stream_.doConvertBuffer[mode] = true;
+      // Lock the DS buffer
+      LPVOID audioPtr;
+      DWORD dataLen;
+      result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 );
+      if ( FAILED( result ) ) {
+        output->Release();
+        buffer->Release();
+        errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsinfo.name << ")!";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
 
-  // Allocate necessary internal buffers
-  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
-  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
-  if ( stream_.userBuffer[mode] == NULL ) {
-    errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
-    goto error;
-  }
+      // Zero the DS buffer
+      ZeroMemory( audioPtr, dataLen );
 
-  if ( stream_.doConvertBuffer[mode] ) {
+      // Unlock the DS buffer
+      result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+      if ( FAILED( result ) ) {
+        output->Release();
+        buffer->Release();
+        errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsinfo.name << ")!";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
+
+      dsBufferSize = bufferBytes;
+      ohandle = (void *) output;
+      bhandle = (void *) buffer;
+    }
 
-    bool makeBuffer = true;
-    bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
     if ( mode == INPUT ) {
-      if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
-        unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
-        if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
+
+      LPDIRECTSOUNDCAPTURE input;
+      result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL );
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!";
+        errorText_ = errorStream_.str();
+        return FAILURE;
       }
-    }
 
-    if ( makeBuffer ) {
-      bufferBytes *= *bufferSize;
-      if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
-      stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
-      if ( stream_.deviceBuffer == NULL ) {
-        errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
-        goto error;
+      DSCCAPS inCaps;
+      inCaps.dwSize = sizeof( inCaps );
+      result = input->GetCaps( &inCaps );
+      if ( FAILED( result ) ) {
+        input->Release();
+        errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsinfo.name << ")!";
+        errorText_ = errorStream_.str();
+        return FAILURE;
       }
-    }
-  }
 
-  // Allocate our DsHandle structures for the stream.
-  if ( stream_.apiHandle == 0 ) {
-    try {
-      handle = new DsHandle;
-    }
-    catch ( std::bad_alloc& ) {
-      errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
+      // Check channel information.
+      if ( inCaps.dwChannels < channels + firstChannel ) {
+        errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
+        return FAILURE;
+      }
+
+      // Check format information.  Use 16-bit format unless user
+      // requests 8-bit.
+      DWORD deviceFormats;
+      if ( channels + firstChannel == 2 ) {
+        deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
+        if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
+          waveFormat.wBitsPerSample = 8;
+          stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+        }
+        else { // assume 16-bit is supported
+          waveFormat.wBitsPerSample = 16;
+          stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+        }
+      }
+      else { // channel == 1
+        deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
+        if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
+          waveFormat.wBitsPerSample = 8;
+          stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+        }
+        else { // assume 16-bit is supported
+          waveFormat.wBitsPerSample = 16;
+          stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+        }
+      }
+      stream_.userFormat = format;
+
+      // Update wave format structure and buffer information.
+      waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+      waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+
+      // Setup the secondary DS buffer description.
+      dsBufferSize = bufferBytes;
+      DSCBUFFERDESC bufferDescription;
+      ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
+      bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
+      bufferDescription.dwFlags = 0;
+      bufferDescription.dwReserved = 0;
+      bufferDescription.dwBufferBytes = bufferBytes;
+      bufferDescription.lpwfxFormat = &waveFormat;
+
+      // Create the capture buffer.
+      LPDIRECTSOUNDCAPTUREBUFFER buffer;
+      result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
+      if ( FAILED( result ) ) {
+        input->Release();
+        errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsinfo.name << ")!";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
+
+      // Lock the capture buffer
+      LPVOID audioPtr;
+      DWORD dataLen;
+      result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 );
+      if ( FAILED( result ) ) {
+        input->Release();
+        buffer->Release();
+        errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsinfo.name << ")!";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
+
+      // Zero the buffer
+      ZeroMemory( audioPtr, dataLen );
+
+      // Unlock the buffer
+      result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+      if ( FAILED( result ) ) {
+        input->Release();
+        buffer->Release();
+        errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsinfo.name << ")!";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
+
+      dsBufferSize = bufferBytes;
+      ohandle = (void *) input;
+      bhandle = (void *) buffer;
+    }
+
+    // Set various stream parameters
+    DsHandle *handle = 0;
+    stream_.nDeviceChannels[mode] = channels + firstChannel;
+    stream_.nUserChannels[mode] = channels;
+    stream_.bufferSize = *bufferSize;
+    stream_.channelOffset[mode] = firstChannel;
+    stream_.deviceInterleaved[mode] = true;
+    if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+    else stream_.userInterleaved = true;
+
+    // Set flag for buffer conversion
+    stream_.doConvertBuffer[mode] = false;
+    if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
+      stream_.doConvertBuffer[mode] = true;
+    if (stream_.userFormat != stream_.deviceFormat[mode])
+      stream_.doConvertBuffer[mode] = true;
+    if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+         stream_.nUserChannels[mode] > 1 )
+      stream_.doConvertBuffer[mode] = true;
+
+    // Allocate necessary internal buffers
+    bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+    stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+    if ( stream_.userBuffer[mode] == NULL ) {
+      errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
       goto error;
     }
 
-    // Create a manual-reset event.
-    handle->condition = CreateEvent( NULL,   // no security
-                                     TRUE,   // manual-reset
-                                     FALSE,  // non-signaled initially
-                                     NULL ); // unnamed
-    stream_.apiHandle = (void *) handle;
-  }
-  else
-    handle = (DsHandle *) stream_.apiHandle;
-  handle->id[mode] = ohandle;
-  handle->buffer[mode] = bhandle;
-  handle->dsBufferSize[mode] = dsBufferSize;
-  handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
-
-  stream_.device[mode] = device;
-  stream_.state = STREAM_STOPPED;
-  if ( stream_.mode == OUTPUT && mode == INPUT )
-    // We had already set up an output stream.
-    stream_.mode = DUPLEX;
-  else
-    stream_.mode = mode;
-  stream_.nBuffers = nBuffers;
-  stream_.sampleRate = sampleRate;
+    if ( stream_.doConvertBuffer[mode] ) {
 
-  // Setup the buffer conversion information structure.
-  if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+      bool makeBuffer = true;
+      bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+      if ( mode == INPUT ) {
+        if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+          unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+          if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
+        }
+      }
 
-  // Setup the callback thread.
-  unsigned threadId;
-  stream_.callbackInfo.object = (void *) this;
-  stream_.callbackInfo.isRunning = true;
-  stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
-                                                &stream_.callbackInfo, 0, &threadId );
-  if ( stream_.callbackInfo.thread == 0 ) {
-    errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
-    goto error;
-  }
+      if ( makeBuffer ) {
+        bufferBytes *= *bufferSize;
+        if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+        stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+        if ( stream_.deviceBuffer == NULL ) {
+          errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
+          goto error;
+        }
+      }
+    }
 
-  // Boost DS thread priority
-  SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
-  return SUCCESS;
+    // Allocate our DsHandle structures for the stream.
+    if ( stream_.apiHandle == 0 ) {
+      try {
+        handle = new DsHandle;
+      }
+      catch ( std::bad_alloc& ) {
+        errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
+        goto error;
+      }
 
- error:
-  if ( handle ) {
-    if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
-      LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
-      LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
-      if ( buffer ) buffer->Release();
-      object->Release();
+      // Create a manual-reset event.
+      handle->condition = CreateEvent( NULL,   // no security
+                                       TRUE,   // manual-reset
+                                       FALSE,  // non-signaled initially
+                                       NULL ); // unnamed
+      stream_.apiHandle = (void *) handle;
     }
-    if ( handle->buffer[1] ) {
-      LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
-      LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
-      if ( buffer ) buffer->Release();
-      object->Release();
+    else
+      handle = (DsHandle *) stream_.apiHandle;
+    handle->id[mode] = ohandle;
+    handle->buffer[mode] = bhandle;
+    handle->dsBufferSize[mode] = dsBufferSize;
+    handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
+
+    stream_.device[mode] = device;
+    stream_.state = STREAM_STOPPED;
+    if ( stream_.mode == OUTPUT && mode == INPUT )
+      // We had already set up an output stream.
+      stream_.mode = DUPLEX;
+    else
+      stream_.mode = mode;
+    stream_.nBuffers = nBuffers;
+    stream_.sampleRate = sampleRate;
+
+    // Setup the buffer conversion information structure.
+    if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+
+    // Setup the callback thread.
+    unsigned threadId;
+    stream_.callbackInfo.object = (void *) this;
+    stream_.callbackInfo.isRunning = true;
+    stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
+                                                  &stream_.callbackInfo, 0, &threadId );
+    if ( stream_.callbackInfo.thread == 0 ) {
+      errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
+      goto error;
     }
-    CloseHandle( handle->condition );
-    delete handle;
-    stream_.apiHandle = 0;
-  }
 
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
+    // Boost DS thread priority
+    SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
+    return SUCCESS;
+
+  error:
+    if ( handle ) {
+      if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
+        LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
+        LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+        if ( buffer ) buffer->Release();
+        object->Release();
+      }
+      if ( handle->buffer[1] ) {
+        LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
+        LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+        if ( buffer ) buffer->Release();
+        object->Release();
+      }
+      CloseHandle( handle->condition );
+      delete handle;
+      stream_.apiHandle = 0;
     }
-  }
 
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
+    for ( int i=0; i<2; i++ ) {
+      if ( stream_.userBuffer[i] ) {
+        free( stream_.userBuffer[i] );
+        stream_.userBuffer[i] = 0;
+      }
+    }
 
-  return FAILURE;
-}
+    if ( stream_.deviceBuffer ) {
+      free( stream_.deviceBuffer );
+      stream_.deviceBuffer = 0;
+    }
 
-void RtApiDs :: closeStream()
-{
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiDs::closeStream(): no open stream to close!";
-    error( RtError::WARNING );
-    return;
+    return FAILURE;
   }
 
-  // Stop the callback thread.
-  stream_.callbackInfo.isRunning = false;
-  WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
-  CloseHandle( (HANDLE) stream_.callbackInfo.thread );
+  void RtApiDs :: closeStream()
+  {
+    if ( stream_.state == STREAM_CLOSED ) {
+      errorText_ = "RtApiDs::closeStream(): no open stream to close!";
+      error( RtError::WARNING );
+      return;
+    }
 
-  DsHandle *handle = (DsHandle *) stream_.apiHandle;
-  if ( handle ) {
-    if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
-      LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
-      LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
-      if ( buffer ) {
-        buffer->Stop();
-        buffer->Release();
+    // Stop the callback thread.
+    stream_.callbackInfo.isRunning = false;
+    WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
+    CloseHandle( (HANDLE) stream_.callbackInfo.thread );
+
+    DsHandle *handle = (DsHandle *) stream_.apiHandle;
+    if ( handle ) {
+      if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
+        LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
+        LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+        if ( buffer ) {
+          buffer->Stop();
+          buffer->Release();
+        }
+        object->Release();
       }
-      object->Release();
+      if ( handle->buffer[1] ) {
+        LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
+        LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+        if ( buffer ) {
+          buffer->Stop();
+          buffer->Release();
+        }
+        object->Release();
+      }
+      CloseHandle( handle->condition );
+      delete handle;
+      stream_.apiHandle = 0;
     }
-    if ( handle->buffer[1] ) {
-      LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
-      LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
-      if ( buffer ) {
-        buffer->Stop();
-        buffer->Release();
+
+    for ( int i=0; i<2; i++ ) {
+      if ( stream_.userBuffer[i] ) {
+        free( stream_.userBuffer[i] );
+        stream_.userBuffer[i] = 0;
       }
-      object->Release();
     }
-    CloseHandle( handle->condition );
-    delete handle;
-    stream_.apiHandle = 0;
-  }
 
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
+    if ( stream_.deviceBuffer ) {
+      free( stream_.deviceBuffer );
+      stream_.deviceBuffer = 0;
     }
-  }
 
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
+    stream_.mode = UNINITIALIZED;
+    stream_.state = STREAM_CLOSED;
   }
 
-  stream_.mode = UNINITIALIZED;
-  stream_.state = STREAM_CLOSED;
-}
-
-void RtApiDs :: startStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_RUNNING ) {
-    errorText_ = "RtApiDs::startStream(): the stream is already running!";
-    error( RtError::WARNING );
-    return;
-  }
+  void RtApiDs :: startStream()
+  {
+    verifyStream();
+    if ( stream_.state == STREAM_RUNNING ) {
+      errorText_ = "RtApiDs::startStream(): the stream is already running!";
+      error( RtError::WARNING );
+      return;
+    }
 
-  // Increase scheduler frequency on lesser windows (a side-effect of
-  // increasing timer accuracy).  On greater windows (Win2K or later),
-  // this is already in effect.
+    // Increase scheduler frequency on lesser windows (a side-effect of
+    // increasing timer accuracy).  On greater windows (Win2K or later),
+    // this is already in effect.
 
-  MUTEX_LOCK( &stream_.mutex );
+    MUTEX_LOCK( &stream_.mutex );
   
-  DsHandle *handle = (DsHandle *) stream_.apiHandle;
+    DsHandle *handle = (DsHandle *) stream_.apiHandle;
 
-  timeBeginPeriod( 1 ); 
+    timeBeginPeriod( 1 ); 
 
-  /*
-  memset( &statistics, 0, sizeof( statistics ) );
-  statistics.sampleRate = stream_.sampleRate;
-  statistics.writeDeviceBufferLeadBytes = handle->dsPointerLeadTime[0];
-  */
+    /*
+      memset( &statistics, 0, sizeof( statistics ) );
+      statistics.sampleRate = stream_.sampleRate;
+      statistics.writeDeviceBufferLeadBytes = handle->dsPointerLeadTime[0];
+    */
 
-  buffersRolling = false;
-  duplexPrerollBytes = 0;
+    buffersRolling = false;
+    duplexPrerollBytes = 0;
 
-  if ( stream_.mode == DUPLEX ) {
-    // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
-    duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
-  }
+    if ( stream_.mode == DUPLEX ) {
+      // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
+      duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
+    }
 
-  HRESULT result = 0;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-    //statistics.outputFrameSize = formatBytes( stream_.deviceFormat[0] ) * stream_.nDeviceChannels[0];
+    HRESULT result = 0;
+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+      //statistics.outputFrameSize = formatBytes( stream_.deviceFormat[0] ) * stream_.nDeviceChannels[0];
 
-    LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
-    result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
+      LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+      result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
+        errorText_ = errorStream_.str();
+        goto unlock;
+      }
     }
-  }
 
-  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-    //statistics.inputFrameSize = formatBytes( stream_.deviceFormat[1]) * stream_.nDeviceChannels[1];
+    if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+      //statistics.inputFrameSize = formatBytes( stream_.deviceFormat[1]) * stream_.nDeviceChannels[1];
 
-    LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
-    result = buffer->Start( DSCBSTART_LOOPING );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
+      LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+      result = buffer->Start( DSCBSTART_LOOPING );
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
+        errorText_ = errorStream_.str();
+        goto unlock;
+      }
     }
-  }
 
-  handle->drainCounter = 0;
-  handle->internalDrain = false;
-  stream_.state = STREAM_RUNNING;
+    handle->drainCounter = 0;
+    handle->internalDrain = false;
+    stream_.state = STREAM_RUNNING;
 
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
-}
+  unlock:
+    MUTEX_UNLOCK( &stream_.mutex );
 
-void RtApiDs :: stopStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
+    if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
   }
 
-  MUTEX_LOCK( &stream_.mutex );
-
-  HRESULT result = 0;
-  LPVOID audioPtr;
-  DWORD dataLen;
-  DsHandle *handle = (DsHandle *) stream_.apiHandle;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-    if ( handle->drainCounter == 0 ) {
-      handle->drainCounter = 1;
-      MUTEX_UNLOCK( &stream_.mutex );
-      WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE );  // block until signaled
-      ResetEvent( handle->condition );
-      MUTEX_LOCK( &stream_.mutex );
-    }
-
-    // Stop the buffer and clear memory
-    LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
-    result = buffer->Stop();
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping output buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
+  void RtApiDs :: stopStream()
+  {
+    verifyStream();
+    if ( stream_.state == STREAM_STOPPED ) {
+      errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
+      error( RtError::WARNING );
+      return;
     }
 
-    // Lock the buffer and clear it so that if we start to play again,
-    // we won't have old data playing.
-    result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking output buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
+    MUTEX_LOCK( &stream_.mutex );
 
-    // Zero the DS buffer
-    ZeroMemory( audioPtr, dataLen );
+    HRESULT result = 0;
+    LPVOID audioPtr;
+    DWORD dataLen;
+    DsHandle *handle = (DsHandle *) stream_.apiHandle;
+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+      if ( handle->drainCounter == 0 ) {
+        handle->drainCounter = 1;
+        MUTEX_UNLOCK( &stream_.mutex );
+        WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE );  // block until signaled
+        ResetEvent( handle->condition );
+        MUTEX_LOCK( &stream_.mutex );
+      }
 
-    // Unlock the DS buffer
-    result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
+      // Stop the buffer and clear memory
+      LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+      result = buffer->Stop();
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping output buffer!";
+        errorText_ = errorStream_.str();
+        goto unlock;
+      }
 
-    // If we start playing again, we must begin at beginning of buffer.
-    handle->bufferPointer[0] = 0;
-  }
+      // Lock the buffer and clear it so that if we start to play again,
+      // we won't have old data playing.
+      result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking output buffer!";
+        errorText_ = errorStream_.str();
+        goto unlock;
+      }
 
-  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-    LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
-    audioPtr = NULL;
-    dataLen = 0;
+      // Zero the DS buffer
+      ZeroMemory( audioPtr, dataLen );
 
-    result = buffer->Stop();
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping input buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
+      // Unlock the DS buffer
+      result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
+        errorText_ = errorStream_.str();
+        goto unlock;
+      }
 
-    // Lock the buffer and clear it so that if we start to play again,
-    // we won't have old data playing.
-    result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking input buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
+      // If we start playing again, we must begin at beginning of buffer.
+      handle->bufferPointer[0] = 0;
     }
 
-    // Zero the DS buffer
-    ZeroMemory( audioPtr, dataLen );
+    if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+      LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+      audioPtr = NULL;
+      dataLen = 0;
 
-    // Unlock the DS buffer
-    result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
+      result = buffer->Stop();
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping input buffer!";
+        errorText_ = errorStream_.str();
+        goto unlock;
+      }
 
-    // If we start recording again, we must begin at beginning of buffer.
-    handle->bufferPointer[1] = 0;
-  }
+      // Lock the buffer and clear it so that if we start to play again,
+      // we won't have old data playing.
+      result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking input buffer!";
+        errorText_ = errorStream_.str();
+        goto unlock;
+      }
 
- unlock:
-  timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
-  stream_.state = STREAM_STOPPED;
-  MUTEX_UNLOCK( &stream_.mutex );
-  if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
-}
+      // Zero the DS buffer
+      ZeroMemory( audioPtr, dataLen );
 
-void RtApiDs :: abortStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  DsHandle *handle = (DsHandle *) stream_.apiHandle;
-  handle->drainCounter = 1;
+      // Unlock the DS buffer
+      result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
+        errorText_ = errorStream_.str();
+        goto unlock;
+      }
 
-  stopStream();
-}
+      // If we start recording again, we must begin at beginning of buffer.
+      handle->bufferPointer[1] = 0;
+    }
 
-void RtApiDs :: callbackEvent()
-{
-  if ( stream_.state == STREAM_STOPPED ) {
-    Sleep(50); // sleep 50 milliseconds
-    return;
+  unlock:
+    timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
+    stream_.state = STREAM_STOPPED;
+    MUTEX_UNLOCK( &stream_.mutex );
+    if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
   }
 
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
-    error( RtError::WARNING );
-    return;
-  }
+  void RtApiDs :: abortStream()
+  {
+    verifyStream();
+    if ( stream_.state == STREAM_STOPPED ) {
+      errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
+      error( RtError::WARNING );
+      return;
+    }
 
-  CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
-  DsHandle *handle = (DsHandle *) stream_.apiHandle;
+    DsHandle *handle = (DsHandle *) stream_.apiHandle;
+    handle->drainCounter = 1;
 
-  // Check if we were draining the stream and signal is finished.
-  if ( handle->drainCounter > stream_.nBuffers + 2 ) {
-    if ( handle->internalDrain == false )
-      SetEvent( handle->condition );
-    else
-      stopStream();
-    return;
+    stopStream();
   }
 
-  MUTEX_LOCK( &stream_.mutex );
-
-  // Invoke user callback to get fresh output data UNLESS we are
-  // draining stream.
-  if ( handle->drainCounter == 0 ) {
-    RtAudioCallback callback = (RtAudioCallback) info->callback;
-    double streamTime = getStreamTime();
-    RtAudioStreamStatus status = 0;
-    if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
-      status |= RTAUDIO_OUTPUT_UNDERFLOW;
-      handle->xrun[0] = false;
+  void RtApiDs :: callbackEvent()
+  {
+    if ( stream_.state == STREAM_STOPPED ) {
+      Sleep(50); // sleep 50 milliseconds
+      return;
     }
-    if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
-      status |= RTAUDIO_INPUT_OVERFLOW;
-      handle->xrun[1] = false;
+
+    if ( stream_.state == STREAM_CLOSED ) {
+      errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
+      error( RtError::WARNING );
+      return;
     }
-    handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
-                                     stream_.bufferSize, streamTime, status, info->userData );
-    if ( handle->drainCounter == 2 ) {
-      MUTEX_UNLOCK( &stream_.mutex );
-      abortStream();
+
+    CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+    DsHandle *handle = (DsHandle *) stream_.apiHandle;
+
+    // Check if we were draining the stream and signal is finished.
+    if ( handle->drainCounter > stream_.nBuffers + 2 ) {
+      if ( handle->internalDrain == false )
+        SetEvent( handle->condition );
+      else
+        stopStream();
       return;
     }
-    else if ( handle->drainCounter == 1 )
-      handle->internalDrain = true;
-  }
 
-  HRESULT result;
-  DWORD currentWritePos, safeWritePos;
-  DWORD currentReadPos, safeReadPos;
-  DWORD leadPos;
-  UINT nextWritePos;
+    MUTEX_LOCK( &stream_.mutex );
+
+    // Invoke user callback to get fresh output data UNLESS we are
+    // draining stream.
+    if ( handle->drainCounter == 0 ) {
+      RtAudioCallback callback = (RtAudioCallback) info->callback;
+      double streamTime = getStreamTime();
+      RtAudioStreamStatus status = 0;
+      if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+        status |= RTAUDIO_OUTPUT_UNDERFLOW;
+        handle->xrun[0] = false;
+      }
+      if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+        status |= RTAUDIO_INPUT_OVERFLOW;
+        handle->xrun[1] = false;
+      }
+      handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+                                       stream_.bufferSize, streamTime, status, info->userData );
+      if ( handle->drainCounter == 2 ) {
+        MUTEX_UNLOCK( &stream_.mutex );
+        abortStream();
+        return;
+      }
+      else if ( handle->drainCounter == 1 )
+        handle->internalDrain = true;
+    }
+
+    HRESULT result;
+    DWORD currentWritePos, safeWritePos;
+    DWORD currentReadPos, safeReadPos;
+    DWORD leadPos;
+    UINT nextWritePos;
 
 #ifdef GENERATE_DEBUG_LOG
-  DWORD writeTime, readTime;
+    DWORD writeTime, readTime;
 #endif
 
-  LPVOID buffer1 = NULL;
-  LPVOID buffer2 = NULL;
-  DWORD bufferSize1 = 0;
-  DWORD bufferSize2 = 0;
+    LPVOID buffer1 = NULL;
+    LPVOID buffer2 = NULL;
+    DWORD bufferSize1 = 0;
+    DWORD bufferSize2 = 0;
 
-  char *buffer;
-  long bufferBytes;
+    char *buffer;
+    long bufferBytes;
 
-  if ( stream_.mode == DUPLEX && !buffersRolling ) {
-    assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
+    if ( stream_.mode == DUPLEX && !buffersRolling ) {
+      assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
 
-    // It takes a while for the devices to get rolling. As a result,
-    // there's no guarantee that the capture and write device pointers
-    // will move in lockstep.  Wait here for both devices to start
-    // rolling, and then set our buffer pointers accordingly.
-    // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
-    // bytes later than the write buffer.
+      // It takes a while for the devices to get rolling. As a result,
+      // there's no guarantee that the capture and write device pointers
+      // will move in lockstep.  Wait here for both devices to start
+      // rolling, and then set our buffer pointers accordingly.
+      // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
+      // bytes later than the write buffer.
 
-    // Stub: a serious risk of having a pre-emptive scheduling round
-    // take place between the two GetCurrentPosition calls... but I'm
-    // really not sure how to solve the problem.  Temporarily boost to
-    // Realtime priority, maybe; but I'm not sure what priority the
-    // DirectSound service threads run at. We *should* be roughly
-    // within a ms or so of correct.
+      // Stub: a serious risk of having a pre-emptive scheduling round
+      // take place between the two GetCurrentPosition calls... but I'm
+      // really not sure how to solve the problem.  Temporarily boost to
+      // Realtime priority, maybe; but I'm not sure what priority the
+      // DirectSound service threads run at. We *should* be roughly
+      // within a ms or so of correct.
 
-    LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
-    LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+      LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+      LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
 
-    DWORD initialWritePos, initialSafeWritePos;
-    DWORD initialReadPos, initialSafeReadPos;
+      DWORD initialWritePos, initialSafeWritePos;
+      DWORD initialReadPos, initialSafeReadPos;
 
-    result = dsWriteBuffer->GetCurrentPosition( &initialWritePos, &initialSafeWritePos );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
-      errorText_ = errorStream_.str();
-      error( RtError::SYSTEM_ERROR );
-    }
-    result = dsCaptureBuffer->GetCurrentPosition( &initialReadPos, &initialSafeReadPos );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
-      errorText_ = errorStream_.str();
-      error( RtError::SYSTEM_ERROR );
-    }
-    while ( true ) {
-      result = dsWriteBuffer->GetCurrentPosition( &currentWritePos, &safeWritePos );
+      result = dsWriteBuffer->GetCurrentPosition( &initialWritePos, &initialSafeWritePos );
       if ( FAILED( result ) ) {
         errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
         errorText_ = errorStream_.str();
         error( RtError::SYSTEM_ERROR );
       }
-      result = dsCaptureBuffer->GetCurrentPosition( &currentReadPos, &safeReadPos );
+      result = dsCaptureBuffer->GetCurrentPosition( &initialReadPos, &initialSafeReadPos );
       if ( FAILED( result ) ) {
         errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
         errorText_ = errorStream_.str();
         error( RtError::SYSTEM_ERROR );
       }
-      if ( safeWritePos != initialSafeWritePos && safeReadPos != initialSafeReadPos ) break;
-      Sleep( 1 );
-    }
+      while ( true ) {
+        result = dsWriteBuffer->GetCurrentPosition( &currentWritePos, &safeWritePos );
+        if ( FAILED( result ) ) {
+          errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+          errorText_ = errorStream_.str();
+          error( RtError::SYSTEM_ERROR );
+        }
+        result = dsCaptureBuffer->GetCurrentPosition( &currentReadPos, &safeReadPos );
+        if ( FAILED( result ) ) {
+          errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+          errorText_ = errorStream_.str();
+          error( RtError::SYSTEM_ERROR );
+        }
+        if ( safeWritePos != initialSafeWritePos && safeReadPos != initialSafeReadPos ) break;
+        Sleep( 1 );
+      }
 
-    assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
+      assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
 
-    buffersRolling = true;
-    handle->bufferPointer[0] = ( safeWritePos + handle->dsPointerLeadTime[0] );
-    handle->bufferPointer[1] = safeReadPos;
-  }
+      buffersRolling = true;
+      handle->bufferPointer[0] = ( safeWritePos + handle->dsPointerLeadTime[0] );
+      handle->bufferPointer[1] = safeReadPos;
+    }
 
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
     
-    LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+      LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
 
-    if ( handle->drainCounter > 1 ) { // write zeros to the output stream
-      bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
-      bufferBytes *= formatBytes( stream_.userFormat );
-      memset( stream_.userBuffer[0], 0, bufferBytes );
-    }
+      if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+        bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
+        bufferBytes *= formatBytes( stream_.userFormat );
+        memset( stream_.userBuffer[0], 0, bufferBytes );
+      }
 
-    // Setup parameters and do buffer conversion if necessary.
-    if ( stream_.doConvertBuffer[0] ) {
-      buffer = stream_.deviceBuffer;
-      convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
-      bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
-      bufferBytes *= formatBytes( stream_.deviceFormat[0] );
-    }
-    else {
-      buffer = stream_.userBuffer[0];
-      bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
-      bufferBytes *= formatBytes( stream_.userFormat );
-    }
+      // Setup parameters and do buffer conversion if necessary.
+      if ( stream_.doConvertBuffer[0] ) {
+        buffer = stream_.deviceBuffer;
+        convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+        bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
+        bufferBytes *= formatBytes( stream_.deviceFormat[0] );
+      }
+      else {
+        buffer = stream_.userBuffer[0];
+        bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
+        bufferBytes *= formatBytes( stream_.userFormat );
+      }
+
+      // No byte swapping necessary in DirectSound implementation.
 
-    // No byte swapping necessary in DirectSound implementation.
+      // Ahhh ... windoze.  16-bit data is signed but 8-bit data is
+      // unsigned.  So, we need to convert our signed 8-bit data here to
+      // unsigned.
+      if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
+        for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
 
-    // Ahhh ... windoze.  16-bit data is signed but 8-bit data is
-    // unsigned.  So, we need to convert our signed 8-bit data here to
-    // unsigned.
-    if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
-      for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
+      DWORD dsBufferSize = handle->dsBufferSize[0];
+      nextWritePos = handle->bufferPointer[0];
 
-    DWORD dsBufferSize = handle->dsBufferSize[0];
-    nextWritePos = handle->bufferPointer[0];
+      DWORD endWrite;
+      while ( true ) {
+        // Find out where the read and "safe write" pointers are.
+        result = dsBuffer->GetCurrentPosition( &currentWritePos, &safeWritePos );
+        if ( FAILED( result ) ) {
+          errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+          errorText_ = errorStream_.str();
+          error( RtError::SYSTEM_ERROR );
+        }
+
+        leadPos = safeWritePos + handle->dsPointerLeadTime[0];
+        if ( leadPos > dsBufferSize ) leadPos -= dsBufferSize;
+        if ( leadPos < nextWritePos ) leadPos += dsBufferSize; // unwrap offset
+        endWrite = nextWritePos + bufferBytes;
+
+        // Check whether the entire write region is behind the play pointer.
+        if ( leadPos >= endWrite ) break;
+
+        // If we are here, then we must wait until the play pointer gets
+        // beyond the write region.  The approach here is to use the
+        // Sleep() function to suspend operation until safePos catches
+        // up. Calculate number of milliseconds to wait as:
+        //   time = distance * (milliseconds/second) * fudgefactor /
+        //          ((bytes/sample) * (samples/second))
+        // A "fudgefactor" less than 1 is used because it was found
+        // that sleeping too long was MUCH worse than sleeping for
+        // several shorter periods.
+        double millis = ( endWrite - leadPos ) * 900.0;
+        millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
+        if ( millis < 1.0 ) millis = 1.0;
+        if ( millis > 50.0 ) {
+          static int nOverruns = 0;
+          ++nOverruns;
+        }
+        Sleep( (DWORD) millis );
+      }
 
-    DWORD endWrite;
-    while ( true ) {
-      // Find out where the read and "safe write" pointers are.
-      result = dsBuffer->GetCurrentPosition( &currentWritePos, &safeWritePos );
+      //if ( statistics.writeDeviceSafeLeadBytes < dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ) ) {
+      //  statistics.writeDeviceSafeLeadBytes = dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] );
+      //}
+
+      if ( dsPointerBetween( nextWritePos, safeWritePos, currentWritePos, dsBufferSize )
+           || dsPointerBetween( endWrite, safeWritePos, currentWritePos, dsBufferSize ) ) { 
+        // We've strayed into the forbidden zone ... resync the read pointer.
+        //++statistics.numberOfWriteUnderruns;
+        handle->xrun[0] = true;
+        nextWritePos = safeWritePos + handle->dsPointerLeadTime[0] - bufferBytes + dsBufferSize;
+        while ( nextWritePos >= dsBufferSize ) nextWritePos -= dsBufferSize;
+        handle->bufferPointer[0] = nextWritePos;
+        endWrite = nextWritePos + bufferBytes;
+      }
+
+      // Lock free space in the buffer
+      result = dsBuffer->Lock( nextWritePos, bufferBytes, &buffer1,
+                               &bufferSize1, &buffer2, &bufferSize2, 0 );
       if ( FAILED( result ) ) {
-        errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+        errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
         errorText_ = errorStream_.str();
         error( RtError::SYSTEM_ERROR );
       }
 
-      leadPos = safeWritePos + handle->dsPointerLeadTime[0];
-      if ( leadPos > dsBufferSize ) leadPos -= dsBufferSize;
-      if ( leadPos < nextWritePos ) leadPos += dsBufferSize; // unwrap offset
-      endWrite = nextWritePos + bufferBytes;
-
-      // Check whether the entire write region is behind the play pointer.
-      if ( leadPos >= endWrite ) break;
-
-      // If we are here, then we must wait until the play pointer gets
-      // beyond the write region.  The approach here is to use the
-      // Sleep() function to suspend operation until safePos catches
-      // up. Calculate number of milliseconds to wait as:
-      //   time = distance * (milliseconds/second) * fudgefactor /
-      //          ((bytes/sample) * (samples/second))
-      // A "fudgefactor" less than 1 is used because it was found
-      // that sleeping too long was MUCH worse than sleeping for
-      // several shorter periods.
-      double millis = ( endWrite - leadPos ) * 900.0;
-      millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
-      if ( millis < 1.0 ) millis = 1.0;
-      if ( millis > 50.0 ) {
-        static int nOverruns = 0;
-        ++nOverruns;
-      }
-      Sleep( (DWORD) millis );
-    }
-
-    //if ( statistics.writeDeviceSafeLeadBytes < dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ) ) {
-    //  statistics.writeDeviceSafeLeadBytes = dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] );
-    //}
-
-    if ( dsPointerBetween( nextWritePos, safeWritePos, currentWritePos, dsBufferSize )
-         || dsPointerBetween( endWrite, safeWritePos, currentWritePos, dsBufferSize ) ) { 
-      // We've strayed into the forbidden zone ... resync the read pointer.
-      //++statistics.numberOfWriteUnderruns;
-      handle->xrun[0] = true;
-      nextWritePos = safeWritePos + handle->dsPointerLeadTime[0] - bufferBytes + dsBufferSize;
-      while ( nextWritePos >= dsBufferSize ) nextWritePos -= dsBufferSize;
-      handle->bufferPointer[0] = nextWritePos;
-      endWrite = nextWritePos + bufferBytes;
-    }
-
-    // Lock free space in the buffer
-    result = dsBuffer->Lock( nextWritePos, bufferBytes, &buffer1,
-                             &bufferSize1, &buffer2, &bufferSize2, 0 );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
-      errorText_ = errorStream_.str();
-      error( RtError::SYSTEM_ERROR );
-    }
-
-    // Copy our buffer into the DS buffer
-    CopyMemory( buffer1, buffer, bufferSize1 );
-    if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
-
-    // Update our buffer offset and unlock sound buffer
-    dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
-      errorText_ = errorStream_.str();
-      error( RtError::SYSTEM_ERROR );
-    }
-    nextWritePos = ( nextWritePos + bufferSize1 + bufferSize2 ) % dsBufferSize;
-    handle->bufferPointer[0] = nextWritePos;
-
-    if ( handle->drainCounter ) {
-      handle->drainCounter++;
-      goto unlock;
-    }
-  }
+      // Copy our buffer into the DS buffer
+      CopyMemory( buffer1, buffer, bufferSize1 );
+      if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
 
-  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+      // Update our buffer offset and unlock sound buffer
+      dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
+        errorText_ = errorStream_.str();
+        error( RtError::SYSTEM_ERROR );
+      }
+      nextWritePos = ( nextWritePos + bufferSize1 + bufferSize2 ) % dsBufferSize;
+      handle->bufferPointer[0] = nextWritePos;
 
-    // Setup parameters.
-    if ( stream_.doConvertBuffer[1] ) {
-      buffer = stream_.deviceBuffer;
-      bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
-      bufferBytes *= formatBytes( stream_.deviceFormat[1] );
-    }
-    else {
-      buffer = stream_.userBuffer[1];
-      bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
-      bufferBytes *= formatBytes( stream_.userFormat );
+      if ( handle->drainCounter ) {
+        handle->drainCounter++;
+        goto unlock;
+      }
     }
 
-    LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
-    long nextReadPos = handle->bufferPointer[1];
-    DWORD dsBufferSize = handle->dsBufferSize[1];
+    if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
 
-    // Find out where the write and "safe read" pointers are.
-    result = dsBuffer->GetCurrentPosition( &currentReadPos, &safeReadPos );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
-      errorText_ = errorStream_.str();
-      error( RtError::SYSTEM_ERROR );
-    }
+      // Setup parameters.
+      if ( stream_.doConvertBuffer[1] ) {
+        buffer = stream_.deviceBuffer;
+        bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
+        bufferBytes *= formatBytes( stream_.deviceFormat[1] );
+      }
+      else {
+        buffer = stream_.userBuffer[1];
+        bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
+        bufferBytes *= formatBytes( stream_.userFormat );
+      }
 
-    if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset
-    DWORD endRead = nextReadPos + bufferBytes;
+      LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+      long nextReadPos = handle->bufferPointer[1];
+      DWORD dsBufferSize = handle->dsBufferSize[1];
 
-    // Handling depends on whether we are INPUT or DUPLEX. 
-    // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
-    // then a wait here will drag the write pointers into the forbidden zone.
-    // 
-    // In DUPLEX mode, rather than wait, we will back off the read pointer until 
-    // it's in a safe position. This causes dropouts, but it seems to be the only 
-    // practical way to sync up the read and write pointers reliably, given the 
-    // the very complex relationship between phase and increment of the read and write 
-    // pointers.
-    //
-    // In order to minimize audible dropouts in DUPLEX mode, we will
-    // provide a pre-roll period of 0.5 seconds in which we return
-    // zeros from the read buffer while the pointers sync up.
+      // Find out where the write and "safe read" pointers are.
+      result = dsBuffer->GetCurrentPosition( &currentReadPos, &safeReadPos );
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+        errorText_ = errorStream_.str();
+        error( RtError::SYSTEM_ERROR );
+      }
 
-    if ( stream_.mode == DUPLEX ) {
-      if ( safeReadPos < endRead ) {
-        if ( duplexPrerollBytes <= 0 ) {
-          // Pre-roll time over. Be more agressive.
-          int adjustment = endRead-safeReadPos;
-
-          handle->xrun[1] = true;
-          //++statistics.numberOfReadOverruns;
-          // Two cases:
-          //   - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
-          //     and perform fine adjustments later.
-          //   - small adjustments: back off by twice as much.
-          if ( adjustment >= 2*bufferBytes )
-            nextReadPos = safeReadPos-2*bufferBytes;
-          else
-            nextReadPos = safeReadPos-bufferBytes-adjustment;
-
-          //statistics.readDeviceSafeLeadBytes = currentReadPos-nextReadPos;
-          //if ( statistics.readDeviceSafeLeadBytes < 0) statistics.readDeviceSafeLeadBytes += dsBufferSize;
-          if ( nextReadPos < 0 ) nextReadPos += dsBufferSize;
+      if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset
+      DWORD endRead = nextReadPos + bufferBytes;
+
+      // Handling depends on whether we are INPUT or DUPLEX. 
+      // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
+      // then a wait here will drag the write pointers into the forbidden zone.
+      // 
+      // In DUPLEX mode, rather than wait, we will back off the read pointer until 
+      // it's in a safe position. This causes dropouts, but it seems to be the only 
+      // practical way to sync up the read and write pointers reliably, given the 
+      // the very complex relationship between phase and increment of the read and write 
+      // pointers.
+      //
+      // In order to minimize audible dropouts in DUPLEX mode, we will
+      // provide a pre-roll period of 0.5 seconds in which we return
+      // zeros from the read buffer while the pointers sync up.
+
+      if ( stream_.mode == DUPLEX ) {
+        if ( safeReadPos < endRead ) {
+          if ( duplexPrerollBytes <= 0 ) {
+            // Pre-roll time over. Be more agressive.
+            int adjustment = endRead-safeReadPos;
+
+            handle->xrun[1] = true;
+            //++statistics.numberOfReadOverruns;
+            // Two cases:
+            //   - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
+            //     and perform fine adjustments later.
+            //   - small adjustments: back off by twice as much.
+            if ( adjustment >= 2*bufferBytes )
+              nextReadPos = safeReadPos-2*bufferBytes;
+            else
+              nextReadPos = safeReadPos-bufferBytes-adjustment;
+
+            //statistics.readDeviceSafeLeadBytes = currentReadPos-nextReadPos;
+            //if ( statistics.readDeviceSafeLeadBytes < 0) statistics.readDeviceSafeLeadBytes += dsBufferSize;
+            if ( nextReadPos < 0 ) nextReadPos += dsBufferSize;
 
+          }
+          else {
+            // In pre=roll time. Just do it.
+            nextReadPos = safeReadPos-bufferBytes;
+            while ( nextReadPos < 0 ) nextReadPos += dsBufferSize;
+          }
+          endRead = nextReadPos + bufferBytes;
         }
-        else {
-          // In pre=roll time. Just do it.
-          nextReadPos = safeReadPos-bufferBytes;
-          while ( nextReadPos < 0 ) nextReadPos += dsBufferSize;
-        }
-        endRead = nextReadPos + bufferBytes;
       }
-    }
-    else { // mode == INPUT
-      while ( safeReadPos < endRead ) {
-        // See comments for playback.
-        double millis = (endRead - safeReadPos) * 900.0;
-        millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
-        if ( millis < 1.0 ) millis = 1.0;
-        Sleep( (DWORD) millis );
-
-        // Wake up, find out where we are now
-        result = dsBuffer->GetCurrentPosition( &currentReadPos, &safeReadPos );
-        if ( FAILED( result ) ) {
-          errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
-          errorText_ = errorStream_.str();
-          error( RtError::SYSTEM_ERROR );
-        }
+      else { // mode == INPUT
+        while ( safeReadPos < endRead ) {
+          // See comments for playback.
+          double millis = (endRead - safeReadPos) * 900.0;
+          millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
+          if ( millis < 1.0 ) millis = 1.0;
+          Sleep( (DWORD) millis );
+
+          // Wake up, find out where we are now
+          result = dsBuffer->GetCurrentPosition( &currentReadPos, &safeReadPos );
+          if ( FAILED( result ) ) {
+            errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+            errorText_ = errorStream_.str();
+            error( RtError::SYSTEM_ERROR );
+          }
       
-        if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset
+          if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset
+        }
       }
-    }
 
-    //if (statistics.readDeviceSafeLeadBytes < dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ) )
-    //  statistics.readDeviceSafeLeadBytes = dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize );
+      //if (statistics.readDeviceSafeLeadBytes < dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ) )
+      //  statistics.readDeviceSafeLeadBytes = dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize );
 
-    // Lock free space in the buffer
-    result = dsBuffer->Lock( nextReadPos, bufferBytes, &buffer1,
-                             &bufferSize1, &buffer2, &bufferSize2, 0 );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
-      errorText_ = errorStream_.str();
-      error( RtError::SYSTEM_ERROR );
-    }
+      // Lock free space in the buffer
+      result = dsBuffer->Lock( nextReadPos, bufferBytes, &buffer1,
+                               &bufferSize1, &buffer2, &bufferSize2, 0 );
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
+        errorText_ = errorStream_.str();
+        error( RtError::SYSTEM_ERROR );
+      }
 
-    if ( duplexPrerollBytes <= 0 ) {
-      // Copy our buffer into the DS buffer
-      CopyMemory( buffer, buffer1, bufferSize1 );
-      if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
-    }
-    else {
-      memset( buffer, 0, bufferSize1 );
-      if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
-      duplexPrerollBytes -= bufferSize1 + bufferSize2;
-    }
+      if ( duplexPrerollBytes <= 0 ) {
+        // Copy our buffer into the DS buffer
+        CopyMemory( buffer, buffer1, bufferSize1 );
+        if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
+      }
+      else {
+        memset( buffer, 0, bufferSize1 );
+        if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
+        duplexPrerollBytes -= bufferSize1 + bufferSize2;
+      }
 
-    // Update our buffer offset and unlock sound buffer
-    nextReadPos = ( nextReadPos + bufferSize1 + bufferSize2 ) % dsBufferSize;
-    dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
-      errorText_ = errorStream_.str();
-      error( RtError::SYSTEM_ERROR );
-    }
-    handle->bufferPointer[1] = nextReadPos;
+      // Update our buffer offset and unlock sound buffer
+      nextReadPos = ( nextReadPos + bufferSize1 + bufferSize2 ) % dsBufferSize;
+      dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
+        errorText_ = errorStream_.str();
+        error( RtError::SYSTEM_ERROR );
+      }
+      handle->bufferPointer[1] = nextReadPos;
 
-    // No byte swapping necessary in DirectSound implementation.
+      // No byte swapping necessary in DirectSound implementation.
 
-    // If necessary, convert 8-bit data from unsigned to signed.
-    if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
-      for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
+      // If necessary, convert 8-bit data from unsigned to signed.
+      if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
+        for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
 
-    // Do buffer conversion if necessary.
-    if ( stream_.doConvertBuffer[1] )
-      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
-  }
+      // Do buffer conversion if necessary.
+      if ( stream_.doConvertBuffer[1] )
+        convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+    }
 #ifdef GENERATE_DEBUG_LOG
-  if ( currentDebugLogEntry < debugLog.size() )
-  {
-    TTickRecord &r = debugLog[currentDebugLogEntry++];
-    r.currentReadPointer = currentReadPos;
-    r.safeReadPointer = safeReadPos;
-    r.currentWritePointer = currentWritePos;
-    r.safeWritePointer = safeWritePos;
-    r.readTime = readTime;
-    r.writeTime = writeTime;
-    r.nextReadPointer = handles[1].bufferPointer;
-    r.nextWritePointer = handles[0].bufferPointer;
-  }
+    if ( currentDebugLogEntry < debugLog.size() )
+      {
+        TTickRecord &r = debugLog[currentDebugLogEntry++];
+        r.currentReadPointer = currentReadPos;
+        r.safeReadPointer = safeReadPos;
+        r.currentWritePointer = currentWritePos;
+        r.safeWritePointer = safeWritePos;
+        r.readTime = readTime;
+        r.writeTime = writeTime;
+        r.nextReadPointer = handles[1].bufferPointer;
+        r.nextWritePointer = handles[0].bufferPointer;
+      }
 #endif
 
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
+  unlock:
+    MUTEX_UNLOCK( &stream_.mutex );
 
-  RtApi::tickStreamTime();
-}
+    RtApi::tickStreamTime();
+  }
 
-// Definitions for utility functions and callbacks
-// specific to the DirectSound implementation.
+  // Definitions for utility functions and callbacks
+  // specific to the DirectSound implementation.
 
-extern "C" unsigned __stdcall callbackHandler( void *ptr )
-{
-  CallbackInfo *info = (CallbackInfo *) ptr;
-  RtApiDs *object = (RtApiDs *) info->object;
-  bool* isRunning = &info->isRunning;
+  extern "C" unsigned __stdcall callbackHandler( void *ptr )
+  {
+    CallbackInfo *info = (CallbackInfo *) ptr;
+    RtApiDs *object = (RtApiDs *) info->object;
+    bool* isRunning = &info->isRunning;
 
-  while ( *isRunning == true ) {
-    object->callbackEvent();
-  }
+    while ( *isRunning == true ) {
+      object->callbackEvent();
+    }
 
-  _endthreadex( 0 );
-  return 0;
-}
+    _endthreadex( 0 );
+    return 0;
+  }
 
 #include "tchar.h"
 
-std::string convertTChar( LPCTSTR name )
-{
-  std::string s;
+  std::string convertTChar( LPCTSTR name )
+  {
+    std::string s;
 
 #if defined( UNICODE ) || defined( _UNICODE )
-  // Yes, this conversion doesn't make sense for two-byte characters
-  // but RtAudio is currently written to return an std::string of
-  // one-byte chars for the device name.
-  for ( unsigned int i=0; i<wcslen( name ); i++ )
-    s.push_back( name[i] );
+    // Yes, this conversion doesn't make sense for two-byte characters
+    // but RtAudio is currently written to return an std::string of
+    // one-byte chars for the device name.
+    for ( unsigned int i=0; i<wcslen( name ); i++ )
+      s.push_back( name[i] );
 #else
-  s.append( std::string( name ) );
+    s.append( std::string( name ) );
 #endif
 
-  return s;
-}
+    return s;
+  }
 
-static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
-                                          LPCTSTR description,
-                                          LPCTSTR module,
-                                          LPVOID lpContext )
-{
-  EnumInfo *info = (EnumInfo *) lpContext;
+  static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
+                                            LPCTSTR description,
+                                            LPCTSTR module,
+                                            LPVOID lpContext )
+  {
+    EnumInfo *info = (EnumInfo *) lpContext;
 
-  HRESULT hr;
-  if ( info->isInput == true ) {
-    DSCCAPS caps;
-    LPDIRECTSOUNDCAPTURE object;
+    HRESULT hr;
+    if ( info->isInput == true ) {
+      DSCCAPS caps;
+      LPDIRECTSOUNDCAPTURE object;
 
-    hr = DirectSoundCaptureCreate(  lpguid, &object,   NULL );
-    if ( hr != DS_OK ) return TRUE;
+      hr = DirectSoundCaptureCreate(  lpguid, &object,   NULL );
+      if ( hr != DS_OK ) return TRUE;
 
-    caps.dwSize = sizeof(caps);
-    hr = object->GetCaps( &caps );
-    if ( hr == DS_OK ) {
-      if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
-        info->counter++;
+      caps.dwSize = sizeof(caps);
+      hr = object->GetCaps( &caps );
+      if ( hr == DS_OK ) {
+        if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
+          info->counter++;
+      }
+      object->Release();
     }
-    object->Release();
-  }
-  else {
-    DSCAPS caps;
-    LPDIRECTSOUND object;
-    hr = DirectSoundCreate(  lpguid, &object,   NULL );
-    if ( hr != DS_OK ) return TRUE;
-
-    caps.dwSize = sizeof(caps);
-    hr = object->GetCaps( &caps );
-    if ( hr == DS_OK ) {
-      if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
-        info->counter++;
+    else {
+      DSCAPS caps;
+      LPDIRECTSOUND object;
+      hr = DirectSoundCreate(  lpguid, &object,   NULL );
+      if ( hr != DS_OK ) return TRUE;
+
+      caps.dwSize = sizeof(caps);
+      hr = object->GetCaps( &caps );
+      if ( hr == DS_OK ) {
+        if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
+          info->counter++;
+      }
+      object->Release();
     }
-    object->Release();
-  }
 
-  if ( info->getDefault && lpguid == NULL ) return FALSE;
+    if ( info->getDefault && lpguid == NULL ) return FALSE;
 
-  if ( info->findIndex && info->counter > info->index ) {
-    info->id = lpguid;
-    info->name = convertTChar( description );
-    return FALSE;
-  }
+    if ( info->findIndex && info->counter > info->index ) {
+      info->id = lpguid;
+      info->name = convertTChar( description );
+      return FALSE;
+    }
 
-  return TRUE;
-}
+    return TRUE;
+  }
 
-static char* getErrorString( int code )
-{
-	switch ( code ) {
+  static char* getErrorString( int code )
+  {
+    switch ( code ) {
 
-  case DSERR_ALLOCATED:
-    return "Already allocated";
+    case DSERR_ALLOCATED:
+      return "Already allocated";
 
-  case DSERR_CONTROLUNAVAIL:
-    return "Control unavailable";
+    case DSERR_CONTROLUNAVAIL:
+      return "Control unavailable";
 
-  case DSERR_INVALIDPARAM:
-    return "Invalid parameter";
+    case DSERR_INVALIDPARAM:
+      return "Invalid parameter";
 
-  case DSERR_INVALIDCALL:
-    return "Invalid call";
+    case DSERR_INVALIDCALL:
+      return "Invalid call";
 
-  case DSERR_GENERIC:
-    return "Generic error";
+    case DSERR_GENERIC:
+      return "Generic error";
 
-  case DSERR_PRIOLEVELNEEDED:
-    return "Priority level needed";
+    case DSERR_PRIOLEVELNEEDED:
+      return "Priority level needed";
 
-  case DSERR_OUTOFMEMORY:
-    return "Out of memory";
+    case DSERR_OUTOFMEMORY:
+      return "Out of memory";
 
-  case DSERR_BADFORMAT:
-    return "The sample rate or the channel format is not supported";
+    case DSERR_BADFORMAT:
+      return "The sample rate or the channel format is not supported";
 
-  case DSERR_UNSUPPORTED:
-    return "Not supported";
+    case DSERR_UNSUPPORTED:
+      return "Not supported";
 
-  case DSERR_NODRIVER:
-    return "No driver";
+    case DSERR_NODRIVER:
+      return "No driver";
 
-  case DSERR_ALREADYINITIALIZED:
-    return "Already initialized";
+    case DSERR_ALREADYINITIALIZED:
+      return "Already initialized";
 
-  case DSERR_NOAGGREGATION:
-    return "No aggregation";
+    case DSERR_NOAGGREGATION:
+      return "No aggregation";
 
-  case DSERR_BUFFERLOST:
-    return "Buffer lost";
+    case DSERR_BUFFERLOST:
+      return "Buffer lost";
 
-  case DSERR_OTHERAPPHASPRIO:
-    return "Another application already has priority";
+    case DSERR_OTHERAPPHASPRIO:
+      return "Another application already has priority";
 
-  case DSERR_UNINITIALIZED:
-    return "Uninitialized";
+    case DSERR_UNINITIALIZED:
+      return "Uninitialized";
 
-  default:
-    return "DirectSound unknown error";
-	}
-}
-//******************** End of __WINDOWS_DS__ *********************//
+    default:
+      return "DirectSound unknown error";
+    }
+  }
+  //******************** End of __WINDOWS_DS__ *********************//
 #endif
 
 
@@ -4870,1222 +4971,1222 @@ static char* getErrorString( int code )
 #include <alsa/asoundlib.h>
 #include <unistd.h>
 
-// A structure to hold various information related to the ALSA API
-// implementation.
-struct AlsaHandle {
-  snd_pcm_t *handles[2];
-  bool synchronized;
-  bool xrun[2];
-  pthread_cond_t runnable;
+  // A structure to hold various information related to the ALSA API
+  // implementation.
+  struct AlsaHandle {
+    snd_pcm_t *handles[2];
+    bool synchronized;
+    bool xrun[2];
+    pthread_cond_t runnable;
 
-  AlsaHandle()
-    :synchronized(false) { xrun[0] = false; xrun[1] = false; }
-};
+    AlsaHandle()
+      :synchronized(false) { xrun[0] = false; xrun[1] = false; }
+  };
 
-extern "C" void *alsaCallbackHandler( void * ptr );
+  extern "C" void *alsaCallbackHandler( void * ptr );
 
-RtApiAlsa :: RtApiAlsa()
-{
-  // Nothing to do here.
-}
+  RtApiAlsa :: RtApiAlsa()
+  {
+    // Nothing to do here.
+  }
 
-RtApiAlsa :: ~RtApiAlsa()
-{
-  if ( stream_.state != STREAM_CLOSED ) closeStream();
-}
+  RtApiAlsa :: ~RtApiAlsa()
+  {
+    if ( stream_.state != STREAM_CLOSED ) closeStream();
+  }
 
-unsigned int RtApiAlsa :: getDeviceCount( void )
-{
-  unsigned nDevices = 0;
-  int result, subdevice, card;
-  char name[64];
-  snd_ctl_t *handle;
-
-  // Count cards and devices
-  card = -1;
-  snd_card_next( &card );
-  while ( card >= 0 ) {
-    sprintf( name, "hw:%d", card );
-    result = snd_ctl_open( &handle, name, 0 );
-    if ( result < 0 ) {
-      errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
-      errorText_ = errorStream_.str();
-      error( RtError::WARNING );
-      goto nextcard;
-		}
-		subdevice = -1;
-		while( 1 ) {
-      result = snd_ctl_pcm_next_device( handle, &subdevice );
-			if ( result < 0 ) {
-        errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
+  unsigned int RtApiAlsa :: getDeviceCount( void )
+  {
+    unsigned nDevices = 0;
+    int result, subdevice, card;
+    char name[64];
+    snd_ctl_t *handle;
+
+    // Count cards and devices
+    card = -1;
+    snd_card_next( &card );
+    while ( card >= 0 ) {
+      sprintf( name, "hw:%d", card );
+      result = snd_ctl_open( &handle, name, 0 );
+      if ( result < 0 ) {
+        errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
         errorText_ = errorStream_.str();
         error( RtError::WARNING );
-        break;
+        goto nextcard;
       }
-			if ( subdevice < 0 )
-        break;
-      nDevices++;
+      subdevice = -1;
+      while( 1 ) {
+        result = snd_ctl_pcm_next_device( handle, &subdevice );
+        if ( result < 0 ) {
+          errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
+          errorText_ = errorStream_.str();
+          error( RtError::WARNING );
+          break;
+        }
+        if ( subdevice < 0 )
+          break;
+        nDevices++;
+      }
+    nextcard:
+      snd_ctl_close( handle );
+      snd_card_next( &card );
     }
-  nextcard:
-    snd_ctl_close( handle );
-    snd_card_next( &card );
+
+    return nDevices;
   }
 
-  return nDevices;
-}
+  RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
+  {
+    RtAudio::DeviceInfo info;
+    info.probed = false;
 
-RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
-{
-  RtAudio::DeviceInfo info;
-  info.probed = false;
+    unsigned nDevices = 0;
+    int result, subdevice, card;
+    char name[64];
+    snd_ctl_t *chandle;
 
-  unsigned nDevices = 0;
-  int result, subdevice, card;
-  char name[64];
-  snd_ctl_t *chandle;
-
-  // Count cards and devices
-  card = -1;
-  snd_card_next( &card );
-  while ( card >= 0 ) {
-    sprintf( name, "hw:%d", card );
-    result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
-    if ( result < 0 ) {
-      errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
-      errorText_ = errorStream_.str();
-      error( RtError::WARNING );
-      goto nextcard;
-		}
-		subdevice = -1;
-		while( 1 ) {
-      result = snd_ctl_pcm_next_device( chandle, &subdevice );
-			if ( result < 0 ) {
-        errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
+    // Count cards and devices
+    card = -1;
+    snd_card_next( &card );
+    while ( card >= 0 ) {
+      sprintf( name, "hw:%d", card );
+      result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
+      if ( result < 0 ) {
+        errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
         errorText_ = errorStream_.str();
         error( RtError::WARNING );
-        break;
+        goto nextcard;
       }
-			if ( subdevice < 0 ) break;
-      if ( nDevices == device ) {
-        sprintf( name, "hw:%d,%d", card, subdevice );
-        goto foundDevice;
+      subdevice = -1;
+      while( 1 ) {
+        result = snd_ctl_pcm_next_device( chandle, &subdevice );
+        if ( result < 0 ) {
+          errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
+          errorText_ = errorStream_.str();
+          error( RtError::WARNING );
+          break;
+        }
+        if ( subdevice < 0 ) break;
+        if ( nDevices == device ) {
+          sprintf( name, "hw:%d,%d", card, subdevice );
+          goto foundDevice;
+        }
+        nDevices++;
       }
-      nDevices++;
+    nextcard:
+      snd_ctl_close( chandle );
+      snd_card_next( &card );
     }
-  nextcard:
-    snd_ctl_close( chandle );
-    snd_card_next( &card );
-  }
 
-  if ( nDevices == 0 ) {
-    errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
-    error( RtError::INVALID_USE );
-  }
+    if ( nDevices == 0 ) {
+      errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
+      error( RtError::INVALID_USE );
+    }
 
-  if ( device >= nDevices ) {
-    errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
-    error( RtError::INVALID_USE );
-  }
+    if ( device >= nDevices ) {
+      errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
+      error( RtError::INVALID_USE );
+    }
 
- foundDevice:
+  foundDevice:
 
-  // If a stream is already open, we cannot probe the stream devices.
-  // Thus, use the saved results.
-  if ( stream_.state != STREAM_CLOSED &&
-       ( stream_.device[0] == device || stream_.device[1] == device ) ) {
-    if ( device >= devices_.size() ) {
-      errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
-      error( RtError::WARNING );
-      return info;
+    // If a stream is already open, we cannot probe the stream devices.
+    // Thus, use the saved results.
+    if ( stream_.state != STREAM_CLOSED &&
+         ( stream_.device[0] == device || stream_.device[1] == device ) ) {
+      if ( device >= devices_.size() ) {
+        errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
+        error( RtError::WARNING );
+        return info;
+      }
+      return devices_[ device ];
     }
-    return devices_[ device ];
-  }
 
-  int openMode = SND_PCM_ASYNC;
-  snd_pcm_stream_t stream;
-	snd_pcm_info_t *pcminfo;
-	snd_pcm_info_alloca( &pcminfo );
-  snd_pcm_t *phandle;
-  snd_pcm_hw_params_t *params;
-  snd_pcm_hw_params_alloca( &params );
+    int openMode = SND_PCM_ASYNC;
+    snd_pcm_stream_t stream;
+    snd_pcm_info_t *pcminfo;
+    snd_pcm_info_alloca( &pcminfo );
+    snd_pcm_t *phandle;
+    snd_pcm_hw_params_t *params;
+    snd_pcm_hw_params_alloca( &params );
 
-  // First try for playback
-  stream = SND_PCM_STREAM_PLAYBACK;
-  snd_pcm_info_set_device( pcminfo, subdevice );
-  snd_pcm_info_set_subdevice( pcminfo, 0 );
-  snd_pcm_info_set_stream( pcminfo, stream );
+    // First try for playback
+    stream = SND_PCM_STREAM_PLAYBACK;
+    snd_pcm_info_set_device( pcminfo, subdevice );
+    snd_pcm_info_set_subdevice( pcminfo, 0 );
+    snd_pcm_info_set_stream( pcminfo, stream );
 
-  result = snd_ctl_pcm_info( chandle, pcminfo );
-  if ( result < 0 ) {
-    // Device probably doesn't support playback.
-    goto captureProbe;
-  }
+    result = snd_ctl_pcm_info( chandle, pcminfo );
+    if ( result < 0 ) {
+      // Device probably doesn't support playback.
+      goto captureProbe;
+    }
 
-  result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
-  if ( result < 0 ) {
-    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    goto captureProbe;
-  }
+    result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
+    if ( result < 0 ) {
+      errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      goto captureProbe;
+    }
 
-  // The device is open ... fill the parameter structure.
-  result = snd_pcm_hw_params_any( phandle, params );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    goto captureProbe;
-  }
+    // The device is open ... fill the parameter structure.
+    result = snd_pcm_hw_params_any( phandle, params );
+    if ( result < 0 ) {
+      snd_pcm_close( phandle );
+      errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      goto captureProbe;
+    }
 
-  // Get output channel information.
-  unsigned int value;
-  result = snd_pcm_hw_params_get_channels_max( params, &value );
-  if ( result < 0 ) {
+    // Get output channel information.
+    unsigned int value;
+    result = snd_pcm_hw_params_get_channels_max( params, &value );
+    if ( result < 0 ) {
+      snd_pcm_close( phandle );
+      errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      goto captureProbe;
+    }
+    info.outputChannels = value;
     snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    goto captureProbe;
-  }
-  info.outputChannels = value;
-  snd_pcm_close( phandle );
 
- captureProbe:
-  // Now try for capture
-  stream = SND_PCM_STREAM_CAPTURE;
-  snd_pcm_info_set_stream( pcminfo, stream );
+  captureProbe:
+    // Now try for capture
+    stream = SND_PCM_STREAM_CAPTURE;
+    snd_pcm_info_set_stream( pcminfo, stream );
 
-  result = snd_ctl_pcm_info( chandle, pcminfo );
-  snd_ctl_close( chandle );
-  if ( result < 0 ) {
-    // Device probably doesn't support capture.
-    if ( info.outputChannels == 0 ) return info;
-    goto probeParameters;
-  }
+    result = snd_ctl_pcm_info( chandle, pcminfo );
+    snd_ctl_close( chandle );
+    if ( result < 0 ) {
+      // Device probably doesn't support capture.
+      if ( info.outputChannels == 0 ) return info;
+      goto probeParameters;
+    }
 
-  result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
-  if ( result < 0 ) {
-    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    if ( info.outputChannels == 0 ) return info;
-    goto probeParameters;
-  }
+    result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
+    if ( result < 0 ) {
+      errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      if ( info.outputChannels == 0 ) return info;
+      goto probeParameters;
+    }
 
-  // The device is open ... fill the parameter structure.
-  result = snd_pcm_hw_params_any( phandle, params );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    if ( info.outputChannels == 0 ) return info;
-    goto probeParameters;
-  }
+    // The device is open ... fill the parameter structure.
+    result = snd_pcm_hw_params_any( phandle, params );
+    if ( result < 0 ) {
+      snd_pcm_close( phandle );
+      errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      if ( info.outputChannels == 0 ) return info;
+      goto probeParameters;
+    }
 
-  result = snd_pcm_hw_params_get_channels_max( params, &value );
-  if ( result < 0 ) {
+    result = snd_pcm_hw_params_get_channels_max( params, &value );
+    if ( result < 0 ) {
+      snd_pcm_close( phandle );
+      errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      if ( info.outputChannels == 0 ) return info;
+      goto probeParameters;
+    }
+    info.inputChannels = value;
     snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    if ( info.outputChannels == 0 ) return info;
-    goto probeParameters;
-  }
-  info.inputChannels = value;
-  snd_pcm_close( phandle );
 
-  // If device opens for both playback and capture, we determine the channels.
-  if ( info.outputChannels > 0 && info.inputChannels > 0 )
-    info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+    // If device opens for both playback and capture, we determine the channels.
+    if ( info.outputChannels > 0 && info.inputChannels > 0 )
+      info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
 
-  // ALSA doesn't provide default devices so we'll use the first available one.
-  if ( device == 0 && info.outputChannels > 0 )
-    info.isDefaultOutput = true;
-  if ( device == 0 && info.inputChannels > 0 )
-    info.isDefaultInput = true;
+    // ALSA doesn't provide default devices so we'll use the first available one.
+    if ( device == 0 && info.outputChannels > 0 )
+      info.isDefaultOutput = true;
+    if ( device == 0 && info.inputChannels > 0 )
+      info.isDefaultInput = true;
+
+  probeParameters:
+    // At this point, we just need to figure out the supported data
+    // formats and sample rates.  We'll proceed by opening the device in
+    // the direction with the maximum number of channels, or playback if
+    // they are equal.  This might limit our sample rate options, but so
+    // be it.
+
+    if ( info.outputChannels >= info.inputChannels )
+      stream = SND_PCM_STREAM_PLAYBACK;
+    else
+      stream = SND_PCM_STREAM_CAPTURE;
+    snd_pcm_info_set_stream( pcminfo, stream );
 
- probeParameters:
-  // At this point, we just need to figure out the supported data
-  // formats and sample rates.  We'll proceed by opening the device in
-  // the direction with the maximum number of channels, or playback if
-  // they are equal.  This might limit our sample rate options, but so
-  // be it.
+    result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
+    if ( result < 0 ) {
+      errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      return info;
+    }
 
-  if ( info.outputChannels >= info.inputChannels )
-    stream = SND_PCM_STREAM_PLAYBACK;
-  else
-    stream = SND_PCM_STREAM_CAPTURE;
-  snd_pcm_info_set_stream( pcminfo, stream );
+    // The device is open ... fill the parameter structure.
+    result = snd_pcm_hw_params_any( phandle, params );
+    if ( result < 0 ) {
+      snd_pcm_close( phandle );
+      errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      return info;
+    }
 
-  result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
-  if ( result < 0 ) {
-    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
+    // Test our discrete set of sample rate values.
+    info.sampleRates.clear();
+    for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
+      if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 )
+        info.sampleRates.push_back( SAMPLE_RATES[i] );
+    }
+    if ( info.sampleRates.size() == 0 ) {
+      snd_pcm_close( phandle );
+      errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      return info;
+    }
 
-  // The device is open ... fill the parameter structure.
-  result = snd_pcm_hw_params_any( phandle, params );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
+    // Probe the supported data formats ... we don't care about endian-ness just yet
+    snd_pcm_format_t format;
+    info.nativeFormats = 0;
+    format = SND_PCM_FORMAT_S8;
+    if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+      info.nativeFormats |= RTAUDIO_SINT8;
+    format = SND_PCM_FORMAT_S16;
+    if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+      info.nativeFormats |= RTAUDIO_SINT16;
+    format = SND_PCM_FORMAT_S24;
+    if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+      info.nativeFormats |= RTAUDIO_SINT24;
+    format = SND_PCM_FORMAT_S32;
+    if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+      info.nativeFormats |= RTAUDIO_SINT32;
+    format = SND_PCM_FORMAT_FLOAT;
+    if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+      info.nativeFormats |= RTAUDIO_FLOAT32;
+    format = SND_PCM_FORMAT_FLOAT64;
+    if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+      info.nativeFormats |= RTAUDIO_FLOAT64;
+
+    // Check that we have at least one supported format
+    if ( info.nativeFormats == 0 ) {
+      errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      return info;
+    }
 
-  // Test our discrete set of sample rate values.
-  info.sampleRates.clear();
-  for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
-    if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 )
-      info.sampleRates.push_back( SAMPLE_RATES[i] );
-  }
-  if ( info.sampleRates.size() == 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
+    // Get the device name
+    char *cardname;
+    result = snd_card_get_name( card, &cardname );
+    if ( result >= 0 )
+      sprintf( name, "hw:%s,%d", cardname, subdevice );
+    info.name = name;
 
-  // Probe the supported data formats ... we don't care about endian-ness just yet
-  snd_pcm_format_t format;
-  info.nativeFormats = 0;
-  format = SND_PCM_FORMAT_S8;
-  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
-    info.nativeFormats |= RTAUDIO_SINT8;
-  format = SND_PCM_FORMAT_S16;
-  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
-    info.nativeFormats |= RTAUDIO_SINT16;
-  format = SND_PCM_FORMAT_S24;
-  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
-    info.nativeFormats |= RTAUDIO_SINT24;
-  format = SND_PCM_FORMAT_S32;
-  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
-    info.nativeFormats |= RTAUDIO_SINT32;
-  format = SND_PCM_FORMAT_FLOAT;
-  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
-    info.nativeFormats |= RTAUDIO_FLOAT32;
-  format = SND_PCM_FORMAT_FLOAT64;
-  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
-    info.nativeFormats |= RTAUDIO_FLOAT64;
-
-  // Check that we have at least one supported format
-  if ( info.nativeFormats == 0 ) {
-    errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
+    // That's all ... close the device and return
+    snd_pcm_close( phandle );
+    info.probed = true;
     return info;
   }
 
-  // Get the device name
-  char *cardname;
-  result = snd_card_get_name( card, &cardname );
-  if ( result >= 0 )
-    sprintf( name, "hw:%s,%d", cardname, subdevice );
-  info.name = name;
-
-  // That's all ... close the device and return
-  snd_pcm_close( phandle );
-  info.probed = true;
-  return info;
-}
-
-void RtApiAlsa :: saveDeviceInfo( void )
-{
-  devices_.clear();
+  void RtApiAlsa :: saveDeviceInfo( void )
+  {
+    devices_.clear();
 
-  unsigned int nDevices = getDeviceCount();
-  devices_.resize( nDevices );
-  for ( unsigned int i=0; i<nDevices; i++ )
-    devices_[i] = getDeviceInfo( i );
-}
+    unsigned int nDevices = getDeviceCount();
+    devices_.resize( nDevices );
+    for ( unsigned int i=0; i<nDevices; i++ )
+      devices_[i] = getDeviceInfo( i );
+  }
 
-bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-                                   unsigned int firstChannel, unsigned int sampleRate,
-                                   RtAudioFormat format, unsigned int *bufferSize,
-                                   RtAudio::StreamOptions *options )
+  bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+                                     unsigned int firstChannel, unsigned int sampleRate,
+                                     RtAudioFormat format, unsigned int *bufferSize,
+                                     RtAudio::StreamOptions *options )
 
-{
+  {
 #if defined(__RTAUDIO_DEBUG__)
-  snd_output_t *out;
-  snd_output_stdio_attach(&out, stderr, 0);
+    snd_output_t *out;
+    snd_output_stdio_attach(&out, stderr, 0);
 #endif
 
-  // I'm not using the "plug" interface ... too much inconsistent behavior.
+    // I'm not using the "plug" interface ... too much inconsistent behavior.
 
-  unsigned nDevices = 0;
-  int result, subdevice, card;
-  char name[64];
-  snd_ctl_t *chandle;
+    unsigned nDevices = 0;
+    int result, subdevice, card;
+    char name[64];
+    snd_ctl_t *chandle;
 
-  // Count cards and devices
-  card = -1;
-  snd_card_next( &card );
-  while ( card >= 0 ) {
-    sprintf( name, "hw:%d", card );
-    result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
-    if ( result < 0 ) {
-      errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
-      errorText_ = errorStream_.str();
-      return FAILURE;
-		}
-		subdevice = -1;
-		while( 1 ) {
-      result = snd_ctl_pcm_next_device( chandle, &subdevice );
-			if ( result < 0 ) break;
-			if ( subdevice < 0 ) break;
-      if ( nDevices == device ) {
-        sprintf( name, "hw:%d,%d", card, subdevice );
-        snd_ctl_close( chandle );
-        goto foundDevice;
-      }
-      nDevices++;
-    }
-    snd_ctl_close( chandle );
+    // Count cards and devices
+    card = -1;
     snd_card_next( &card );
-  }
-
-  if ( nDevices == 0 ) {
-    // This should not happen because a check is made before this function is called.
-    errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
-    return FAILURE;
-  }
+    while ( card >= 0 ) {
+      sprintf( name, "hw:%d", card );
+      result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
+      if ( result < 0 ) {
+        errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
+      subdevice = -1;
+      while( 1 ) {
+        result = snd_ctl_pcm_next_device( chandle, &subdevice );
+        if ( result < 0 ) break;
+        if ( subdevice < 0 ) break;
+        if ( nDevices == device ) {
+          sprintf( name, "hw:%d,%d", card, subdevice );
+          snd_ctl_close( chandle );
+          goto foundDevice;
+        }
+        nDevices++;
+      }
+      snd_ctl_close( chandle );
+      snd_card_next( &card );
+    }
 
-  if ( device >= nDevices ) {
-    // This should not happen because a check is made before this function is called.
-    errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
-    return FAILURE;
-  }
+    if ( nDevices == 0 ) {
+      // This should not happen because a check is made before this function is called.
+      errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
+      return FAILURE;
+    }
 
- foundDevice:
+    if ( device >= nDevices ) {
+      // This should not happen because a check is made before this function is called.
+      errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
+      return FAILURE;
+    }
 
-  // The getDeviceInfo() function will not work for a device that is
-  // already open.  Thus, we'll probe the system before opening a
-  // stream and save the results for use by getDeviceInfo().
-  if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
-    this->saveDeviceInfo();
+  foundDevice:
 
-  snd_pcm_stream_t stream;
-  if ( mode == OUTPUT )
-    stream = SND_PCM_STREAM_PLAYBACK;
-  else
-    stream = SND_PCM_STREAM_CAPTURE;
+    // The getDeviceInfo() function will not work for a device that is
+    // already open.  Thus, we'll probe the system before opening a
+    // stream and save the results for use by getDeviceInfo().
+    if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
+      this->saveDeviceInfo();
 
-  snd_pcm_t *phandle;
-  int openMode = SND_PCM_ASYNC;
-  result = snd_pcm_open( &phandle, name, stream, openMode );
-  if ( result < 0 ) {
+    snd_pcm_stream_t stream;
     if ( mode == OUTPUT )
-      errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
+      stream = SND_PCM_STREAM_PLAYBACK;
     else
-      errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+      stream = SND_PCM_STREAM_CAPTURE;
 
-  // Fill the parameter structure.
-  snd_pcm_hw_params_t *hw_params;
-  snd_pcm_hw_params_alloca( &hw_params );
-  result = snd_pcm_hw_params_any( phandle, hw_params );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+    snd_pcm_t *phandle;
+    int openMode = SND_PCM_ASYNC;
+    result = snd_pcm_open( &phandle, name, stream, openMode );
+    if ( result < 0 ) {
+      if ( mode == OUTPUT )
+        errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
+      else
+        errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+
+    // Fill the parameter structure.
+    snd_pcm_hw_params_t *hw_params;
+    snd_pcm_hw_params_alloca( &hw_params );
+    result = snd_pcm_hw_params_any( phandle, hw_params );
+    if ( result < 0 ) {
+      snd_pcm_close( phandle );
+      errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
 #if defined(__RTAUDIO_DEBUG__)
-  fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
-  snd_pcm_hw_params_dump( hw_params, out );
+    fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
+    snd_pcm_hw_params_dump( hw_params, out );
 #endif
 
-  // Set access ... check user preference.
-  if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
-    stream_.userInterleaved = false;
-    result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
-    if ( result < 0 ) {
+    // Set access ... check user preference.
+    if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
+      stream_.userInterleaved = false;
+      result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
+      if ( result < 0 ) {
+        result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
+        stream_.deviceInterleaved[mode] =  true;
+      }
+      else
+        stream_.deviceInterleaved[mode] = false;
+    }
+    else {
+      stream_.userInterleaved = true;
       result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
-      stream_.deviceInterleaved[mode] =  true;
+      if ( result < 0 ) {
+        result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
+        stream_.deviceInterleaved[mode] =  false;
+      }
+      else
+        stream_.deviceInterleaved[mode] =  true;
     }
-    else
-      stream_.deviceInterleaved[mode] = false;
-  }
-  else {
-    stream_.userInterleaved = true;
-    result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
+
     if ( result < 0 ) {
-      result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
-      stream_.deviceInterleaved[mode] =  false;
+      snd_pcm_close( phandle );
+      errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      return FAILURE;
     }
-    else
-      stream_.deviceInterleaved[mode] =  true;
-  }
 
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Determine how to set the device format.
-  stream_.userFormat = format;
-  snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
+    // Determine how to set the device format.
+    stream_.userFormat = format;
+    snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
+
+    if ( format == RTAUDIO_SINT8 )
+      deviceFormat = SND_PCM_FORMAT_S8;
+    else if ( format == RTAUDIO_SINT16 )
+      deviceFormat = SND_PCM_FORMAT_S16;
+    else if ( format == RTAUDIO_SINT24 )
+      deviceFormat = SND_PCM_FORMAT_S24;
+    else if ( format == RTAUDIO_SINT32 )
+      deviceFormat = SND_PCM_FORMAT_S32;
+    else if ( format == RTAUDIO_FLOAT32 )
+      deviceFormat = SND_PCM_FORMAT_FLOAT;
+    else if ( format == RTAUDIO_FLOAT64 )
+      deviceFormat = SND_PCM_FORMAT_FLOAT64;
+
+    if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
+      stream_.deviceFormat[mode] = format;
+      goto setFormat;
+    }
 
-  if ( format == RTAUDIO_SINT8 )
-    deviceFormat = SND_PCM_FORMAT_S8;
-  else if ( format == RTAUDIO_SINT16 )
-    deviceFormat = SND_PCM_FORMAT_S16;
-  else if ( format == RTAUDIO_SINT24 )
-    deviceFormat = SND_PCM_FORMAT_S24;
-  else if ( format == RTAUDIO_SINT32 )
-    deviceFormat = SND_PCM_FORMAT_S32;
-  else if ( format == RTAUDIO_FLOAT32 )
-    deviceFormat = SND_PCM_FORMAT_FLOAT;
-  else if ( format == RTAUDIO_FLOAT64 )
+    // The user requested format is not natively supported by the device.
     deviceFormat = SND_PCM_FORMAT_FLOAT64;
+    if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
+      stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
+      goto setFormat;
+    }
 
-  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
-    stream_.deviceFormat[mode] = format;
-    goto setFormat;
-  }
-
-  // The user requested format is not natively supported by the device.
-  deviceFormat = SND_PCM_FORMAT_FLOAT64;
-  if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
-    stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
-    goto setFormat;
-  }
-
-  deviceFormat = SND_PCM_FORMAT_FLOAT;
-  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
-    stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
-    goto setFormat;
-  }
-
-  deviceFormat = SND_PCM_FORMAT_S32;
-  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
-    stream_.deviceFormat[mode] = RTAUDIO_SINT32;
-    goto setFormat;
-  }
+    deviceFormat = SND_PCM_FORMAT_FLOAT;
+    if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+      stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+      goto setFormat;
+    }
 
-  deviceFormat = SND_PCM_FORMAT_S24;
-  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
-    stream_.deviceFormat[mode] = RTAUDIO_SINT24;
-    goto setFormat;
-  }
+    deviceFormat = SND_PCM_FORMAT_S32;
+    if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+      stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+      goto setFormat;
+    }
 
-  deviceFormat = SND_PCM_FORMAT_S16;
-  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
-    stream_.deviceFormat[mode] = RTAUDIO_SINT16;
-    goto setFormat;
-  }
+    deviceFormat = SND_PCM_FORMAT_S24;
+    if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+      stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+      goto setFormat;
+    }
 
-  deviceFormat = SND_PCM_FORMAT_S8;
-  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
-    stream_.deviceFormat[mode] = RTAUDIO_SINT8;
-    goto setFormat;
-  }
+    deviceFormat = SND_PCM_FORMAT_S16;
+    if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+      stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+      goto setFormat;
+    }
 
-  // If we get here, no supported format was found.
-  errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
-  errorText_ = errorStream_.str();
-  return FAILURE;
+    deviceFormat = SND_PCM_FORMAT_S8;
+    if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+      stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+      goto setFormat;
+    }
 
- setFormat:
-  result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
+    // If we get here, no supported format was found.
+    errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
     errorText_ = errorStream_.str();
     return FAILURE;
-  }
 
-  // Determine whether byte-swaping is necessary.
-  stream_.doByteSwap[mode] = false;
-  if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
-    result = snd_pcm_format_cpu_endian( deviceFormat );
-    if ( result == 0 )
-      stream_.doByteSwap[mode] = true;
-    else if (result < 0) {
+  setFormat:
+    result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
+    if ( result < 0 ) {
       snd_pcm_close( phandle );
-      errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
+      errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
       errorText_ = errorStream_.str();
       return FAILURE;
     }
-  }
 
-  // Set the sample rate.
-  result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+    // Determine whether byte-swaping is necessary.
+    stream_.doByteSwap[mode] = false;
+    if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
+      result = snd_pcm_format_cpu_endian( deviceFormat );
+      if ( result == 0 )
+        stream_.doByteSwap[mode] = true;
+      else if (result < 0) {
+        snd_pcm_close( phandle );
+        errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
+    }
 
-  // Determine the number of channels for this device.  We support a possible
-  // minimum device channel number > than the value requested by the user.
-  stream_.nUserChannels[mode] = channels;
-  unsigned int value;
-  result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
-  unsigned int deviceChannels = value;
-  if ( result < 0 || deviceChannels < channels + firstChannel ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+    // Set the sample rate.
+    result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
+    if ( result < 0 ) {
+      snd_pcm_close( phandle );
+      errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
-  result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-  deviceChannels = value;
-  if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
-  stream_.nDeviceChannels[mode] = deviceChannels;
+    // Determine the number of channels for this device.  We support a possible
+    // minimum device channel number > than the value requested by the user.
+    stream_.nUserChannels[mode] = channels;
+    unsigned int value;
+    result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
+    unsigned int deviceChannels = value;
+    if ( result < 0 || deviceChannels < channels + firstChannel ) {
+      snd_pcm_close( phandle );
+      errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
-  // Set the device channels.
-  result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+    result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
+    if ( result < 0 ) {
+      snd_pcm_close( phandle );
+      errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+    deviceChannels = value;
+    if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
+    stream_.nDeviceChannels[mode] = deviceChannels;
 
-  // Set the buffer number, which in ALSA is referred to as the "period".
-  int totalSize, dir;
-  unsigned int periods = 0;
-  if ( options ) periods = options->numberOfBuffers;
-  totalSize = *bufferSize * periods;
+    // Set the device channels.
+    result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
+    if ( result < 0 ) {
+      snd_pcm_close( phandle );
+      errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
-  // Set the buffer (or period) size.
-  snd_pcm_uframes_t periodSize = *bufferSize;
-  result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-  *bufferSize = periodSize;
+    // Set the buffer number, which in ALSA is referred to as the "period".
+    int totalSize, dir;
+    unsigned int periods = 0;
+    if ( options ) periods = options->numberOfBuffers;
+    totalSize = *bufferSize * periods;
 
-  if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
-  else periods = totalSize / *bufferSize;
-  // Even though the hardware might allow 1 buffer, it won't work reliably.
-  if ( periods < 2 ) periods = 2;
-  result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+    // Set the buffer (or period) size.
+    snd_pcm_uframes_t periodSize = *bufferSize;
+    result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
+    if ( result < 0 ) {
+      snd_pcm_close( phandle );
+      errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+    *bufferSize = periodSize;
 
-  // If attempting to setup a duplex stream, the bufferSize parameter
-  // MUST be the same in both directions!
-  if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+    if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
+    else periods = totalSize / *bufferSize;
+    // Even though the hardware might allow 1 buffer, it won't work reliably.
+    if ( periods < 2 ) periods = 2;
+    result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
+    if ( result < 0 ) {
+      snd_pcm_close( phandle );
+      errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
-  stream_.bufferSize = *bufferSize;
+    // If attempting to setup a duplex stream, the bufferSize parameter
+    // MUST be the same in both directions!
+    if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
+      errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
-  // Install the hardware configuration
-  result = snd_pcm_hw_params( phandle, hw_params );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+    stream_.bufferSize = *bufferSize;
+
+    // Install the hardware configuration
+    result = snd_pcm_hw_params( phandle, hw_params );
+    if ( result < 0 ) {
+      snd_pcm_close( phandle );
+      errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
 #if defined(__RTAUDIO_DEBUG__)
-  fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
-  snd_pcm_hw_params_dump( hw_params, out );
+    fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
+    snd_pcm_hw_params_dump( hw_params, out );
 #endif
 
-  // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
-  snd_pcm_sw_params_t *sw_params = NULL;
-  snd_pcm_sw_params_alloca( &sw_params );
-  snd_pcm_sw_params_current( phandle, sw_params );
-  snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
-  snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
-  snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
-
-  // The following two settings were suggested by Theo Veenker
-  //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
-  //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
-
-  // here are two options for a fix
-  //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
-  snd_pcm_uframes_t val;
-  snd_pcm_sw_params_get_boundary( sw_params, &val );
-  snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
-
-  result = snd_pcm_sw_params( phandle, sw_params );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+    // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
+    snd_pcm_sw_params_t *sw_params = NULL;
+    snd_pcm_sw_params_alloca( &sw_params );
+    snd_pcm_sw_params_current( phandle, sw_params );
+    snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
+    snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
+    snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
+
+    // The following two settings were suggested by Theo Veenker
+    //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
+    //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
+
+    // here are two options for a fix
+    //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
+    snd_pcm_uframes_t val;
+    snd_pcm_sw_params_get_boundary( sw_params, &val );
+    snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
+
+    result = snd_pcm_sw_params( phandle, sw_params );
+    if ( result < 0 ) {
+      snd_pcm_close( phandle );
+      errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
 #if defined(__RTAUDIO_DEBUG__)
-  fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
-  snd_pcm_sw_params_dump( sw_params, out );
+    fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
+    snd_pcm_sw_params_dump( sw_params, out );
 #endif
 
-  // Set flags for buffer conversion
-  stream_.doConvertBuffer[mode] = false;
-  if ( stream_.userFormat != stream_.deviceFormat[mode] )
-    stream_.doConvertBuffer[mode] = true;
-  if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
-    stream_.doConvertBuffer[mode] = true;
-  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
-       stream_.nUserChannels[mode] > 1 )
-    stream_.doConvertBuffer[mode] = true;
+    // Set flags for buffer conversion
+    stream_.doConvertBuffer[mode] = false;
+    if ( stream_.userFormat != stream_.deviceFormat[mode] )
+      stream_.doConvertBuffer[mode] = true;
+    if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+      stream_.doConvertBuffer[mode] = true;
+    if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+         stream_.nUserChannels[mode] > 1 )
+      stream_.doConvertBuffer[mode] = true;
+
+    // Allocate the ApiHandle if necessary and then save.
+    AlsaHandle *apiInfo = 0;
+    if ( stream_.apiHandle == 0 ) {
+      try {
+        apiInfo = (AlsaHandle *) new AlsaHandle;
+      }
+      catch ( std::bad_alloc& ) {
+        errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
+        goto error;
+      }
 
-  // Allocate the ApiHandle if necessary and then save.
-  AlsaHandle *apiInfo = 0;
-  if ( stream_.apiHandle == 0 ) {
-    try {
-      apiInfo = (AlsaHandle *) new AlsaHandle;
+      if ( pthread_cond_init( &apiInfo->runnable, NULL ) ) {
+        errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
+        goto error;
+      }
+
+      stream_.apiHandle = (void *) apiInfo;
+      apiInfo->handles[0] = 0;
+      apiInfo->handles[1] = 0;
     }
-    catch ( std::bad_alloc& ) {
-      errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
-      goto error;
+    else {
+      apiInfo = (AlsaHandle *) stream_.apiHandle;
     }
+    apiInfo->handles[mode] = phandle;
 
-    if ( pthread_cond_init( &apiInfo->runnable, NULL ) ) {
-      errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
+    // Allocate necessary internal buffers.
+    unsigned long bufferBytes;
+    bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+    stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+    if ( stream_.userBuffer[mode] == NULL ) {
+      errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
       goto error;
     }
 
-    stream_.apiHandle = (void *) apiInfo;
-    apiInfo->handles[0] = 0;
-    apiInfo->handles[1] = 0;
-  }
-  else {
-    apiInfo = (AlsaHandle *) stream_.apiHandle;
-  }
-  apiInfo->handles[mode] = phandle;
+    if ( stream_.doConvertBuffer[mode] ) {
 
-  // Allocate necessary internal buffers.
-  unsigned long bufferBytes;
-  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
-  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
-  if ( stream_.userBuffer[mode] == NULL ) {
-    errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
-    goto error;
-  }
+      bool makeBuffer = true;
+      bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+      if ( mode == INPUT ) {
+        if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+          unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+          if ( bufferBytes <= bytesOut ) makeBuffer = false;
+        }
+      }
 
-  if ( stream_.doConvertBuffer[mode] ) {
+      if ( makeBuffer ) {
+        bufferBytes *= *bufferSize;
+        if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+        stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+        if ( stream_.deviceBuffer == NULL ) {
+          errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
+          goto error;
+        }
+      }
+    }
 
-    bool makeBuffer = true;
-    bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
-    if ( mode == INPUT ) {
-      if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
-        unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
-        if ( bufferBytes <= bytesOut ) makeBuffer = false;
+    stream_.sampleRate = sampleRate;
+    stream_.nBuffers = periods;
+    stream_.device[mode] = device;
+    stream_.state = STREAM_STOPPED;
+
+    // Setup the buffer conversion information structure.
+    if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+
+    // Setup thread if necessary.
+    if ( stream_.mode == OUTPUT && mode == INPUT ) {
+      // We had already set up an output stream.
+      stream_.mode = DUPLEX;
+      // Link the streams if possible.
+      apiInfo->synchronized = false;
+      if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
+        apiInfo->synchronized = true;
+      else {
+        errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
+        error( RtError::WARNING );
       }
     }
+    else {
+      stream_.mode = mode;
 
-    if ( makeBuffer ) {
-      bufferBytes *= *bufferSize;
-      if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
-      stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
-      if ( stream_.deviceBuffer == NULL ) {
-        errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
+      // Setup callback thread.
+      stream_.callbackInfo.object = (void *) this;
+
+      // Set the thread attributes for joinable and realtime scheduling
+      // priority (optional).  The higher priority will only take affect
+      // if the program is run as root or suid. Note, under Linux
+      // processes with CAP_SYS_NICE privilege, a user can change
+      // scheduling policy and priority (thus need not be root). See
+      // POSIX "capabilities".
+      pthread_attr_t attr;
+      pthread_attr_init( &attr );
+      pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
+#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
+      if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+        struct sched_param param;
+        int priority = options->priority;
+        int min = sched_get_priority_min( SCHED_RR );
+        int max = sched_get_priority_max( SCHED_RR );
+        if ( priority < min ) priority = min;
+        else if ( priority > max ) priority = max;
+        param.sched_priority = priority;
+        pthread_attr_setschedparam( &attr, &param );
+        pthread_attr_setschedpolicy( &attr, SCHED_RR );
+      }
+      else
+        pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#else
+      pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#endif
+
+      stream_.callbackInfo.isRunning = true;
+      result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
+      pthread_attr_destroy( &attr );
+      if ( result ) {
+        stream_.callbackInfo.isRunning = false;
+        errorText_ = "RtApiAlsa::error creating callback thread!";
         goto error;
       }
     }
-  }
-
-  stream_.sampleRate = sampleRate;
-  stream_.nBuffers = periods;
-  stream_.device[mode] = device;
-  stream_.state = STREAM_STOPPED;
 
-  // Setup the buffer conversion information structure.
-  if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+    return SUCCESS;
 
-  // Setup thread if necessary.
-  if ( stream_.mode == OUTPUT && mode == INPUT ) {
-    // We had already set up an output stream.
-    stream_.mode = DUPLEX;
-    // Link the streams if possible.
-    apiInfo->synchronized = false;
-    if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
-      apiInfo->synchronized = true;
-    else {
-      errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
-      error( RtError::WARNING );
+  error:
+    if ( apiInfo ) {
+      pthread_cond_destroy( &apiInfo->runnable );
+      if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
+      if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
+      delete apiInfo;
+      stream_.apiHandle = 0;
     }
-  }
-  else {
-    stream_.mode = mode;
-
-    // Setup callback thread.
-    stream_.callbackInfo.object = (void *) this;
 
-    // Set the thread attributes for joinable and realtime scheduling
-    // priority (optional).  The higher priority will only take affect
-    // if the program is run as root or suid. Note, under Linux
-    // processes with CAP_SYS_NICE privilege, a user can change
-    // scheduling policy and priority (thus need not be root). See
-    // POSIX "capabilities".
-    pthread_attr_t attr;
-    pthread_attr_init( &attr );
-    pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
-#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
-    if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
-      struct sched_param param;
-      int priority = options->priority;
-      int min = sched_get_priority_min( SCHED_RR );
-      int max = sched_get_priority_max( SCHED_RR );
-      if ( priority < min ) priority = min;
-      else if ( priority > max ) priority = max;
-      param.sched_priority = priority;
-      pthread_attr_setschedparam( &attr, &param );
-      pthread_attr_setschedpolicy( &attr, SCHED_RR );
+    for ( int i=0; i<2; i++ ) {
+      if ( stream_.userBuffer[i] ) {
+        free( stream_.userBuffer[i] );
+        stream_.userBuffer[i] = 0;
+      }
     }
-    else
-      pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
-#else
-    pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
-#endif
 
-    stream_.callbackInfo.isRunning = true;
-    result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
-    pthread_attr_destroy( &attr );
-    if ( result ) {
-      stream_.callbackInfo.isRunning = false;
-      errorText_ = "RtApiAlsa::error creating callback thread!";
-      goto error;
+    if ( stream_.deviceBuffer ) {
+      free( stream_.deviceBuffer );
+      stream_.deviceBuffer = 0;
     }
+
+    return FAILURE;
   }
 
-  return SUCCESS;
+  void RtApiAlsa :: closeStream()
+  {
+    if ( stream_.state == STREAM_CLOSED ) {
+      errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
+      error( RtError::WARNING );
+      return;
+    }
 
- error:
-  if ( apiInfo ) {
-    pthread_cond_destroy( &apiInfo->runnable );
-    if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
-    if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
-    delete apiInfo;
-    stream_.apiHandle = 0;
-  }
+    AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+    stream_.callbackInfo.isRunning = false;
+    MUTEX_LOCK( &stream_.mutex );
+    if ( stream_.state == STREAM_STOPPED )
+      pthread_cond_signal( &apiInfo->runnable );
+    MUTEX_UNLOCK( &stream_.mutex );
+    pthread_join( stream_.callbackInfo.thread, NULL );
 
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
+    if ( stream_.state == STREAM_RUNNING ) {
+      stream_.state = STREAM_STOPPED;
+      if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
+        snd_pcm_drop( apiInfo->handles[0] );
+      if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
+        snd_pcm_drop( apiInfo->handles[1] );
     }
-  }
 
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
+    if ( apiInfo ) {
+      pthread_cond_destroy( &apiInfo->runnable );
+      if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
+      if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
+      delete apiInfo;
+      stream_.apiHandle = 0;
+    }
 
-  return FAILURE;
-}
+    for ( int i=0; i<2; i++ ) {
+      if ( stream_.userBuffer[i] ) {
+        free( stream_.userBuffer[i] );
+        stream_.userBuffer[i] = 0;
+      }
+    }
 
-void RtApiAlsa :: closeStream()
-{
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
-    error( RtError::WARNING );
-    return;
+    if ( stream_.deviceBuffer ) {
+      free( stream_.deviceBuffer );
+      stream_.deviceBuffer = 0;
+    }
+
+    stream_.mode = UNINITIALIZED;
+    stream_.state = STREAM_CLOSED;
   }
 
-  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
-  stream_.callbackInfo.isRunning = false;
-  MUTEX_LOCK( &stream_.mutex );
-  if ( stream_.state == STREAM_STOPPED )
-    pthread_cond_signal( &apiInfo->runnable );
-  MUTEX_UNLOCK( &stream_.mutex );
-  pthread_join( stream_.callbackInfo.thread, NULL );
+  void RtApiAlsa :: startStream()
+  {
+    // This method calls snd_pcm_prepare if the device isn't already in that state.
 
-  if ( stream_.state == STREAM_RUNNING ) {
-    stream_.state = STREAM_STOPPED;
-    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
-      snd_pcm_drop( apiInfo->handles[0] );
-    if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
-      snd_pcm_drop( apiInfo->handles[1] );
-  }
+    verifyStream();
+    if ( stream_.state == STREAM_RUNNING ) {
+      errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
+      error( RtError::WARNING );
+      return;
+    }
 
-  if ( apiInfo ) {
-    pthread_cond_destroy( &apiInfo->runnable );
-    if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
-    if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
-    delete apiInfo;
-    stream_.apiHandle = 0;
-  }
+    MUTEX_LOCK( &stream_.mutex );
 
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
+    int result = 0;
+    snd_pcm_state_t state;
+    AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+    snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+      state = snd_pcm_state( handle[0] );
+      if ( state != SND_PCM_STATE_PREPARED ) {
+        result = snd_pcm_prepare( handle[0] );
+        if ( result < 0 ) {
+          errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
+          errorText_ = errorStream_.str();
+          goto unlock;
+        }
+      }
     }
-  }
 
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
+    if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+      state = snd_pcm_state( handle[1] );
+      if ( state != SND_PCM_STATE_PREPARED ) {
+        result = snd_pcm_prepare( handle[1] );
+        if ( result < 0 ) {
+          errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
+          errorText_ = errorStream_.str();
+          goto unlock;
+        }
+      }
+    }
 
-  stream_.mode = UNINITIALIZED;
-  stream_.state = STREAM_CLOSED;
-}
+    stream_.state = STREAM_RUNNING;
 
-void RtApiAlsa :: startStream()
-{
-  // This method calls snd_pcm_prepare if the device isn't already in that state.
+  unlock:
+    MUTEX_UNLOCK( &stream_.mutex );
 
-  verifyStream();
-  if ( stream_.state == STREAM_RUNNING ) {
-    errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
-    error( RtError::WARNING );
-    return;
+    pthread_cond_signal( &apiInfo->runnable );
+
+    if ( result >= 0 ) return;
+    error( RtError::SYSTEM_ERROR );
   }
 
-  MUTEX_LOCK( &stream_.mutex );
+  void RtApiAlsa :: stopStream()
+  {
+    verifyStream();
+    if ( stream_.state == STREAM_STOPPED ) {
+      errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
+      error( RtError::WARNING );
+      return;
+    }
 
-  int result = 0;
-  snd_pcm_state_t state;
-  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
-  snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-    state = snd_pcm_state( handle[0] );
-    if ( state != SND_PCM_STATE_PREPARED ) {
-      result = snd_pcm_prepare( handle[0] );
+    // Change the state before the lock to improve shutdown response.
+    stream_.state = STREAM_STOPPED;
+    MUTEX_LOCK( &stream_.mutex );
+
+    int result = 0;
+    AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+    snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+      if ( apiInfo->synchronized ) 
+        result = snd_pcm_drop( handle[0] );
+      else
+        result = snd_pcm_drain( handle[0] );
       if ( result < 0 ) {
-        errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
+        errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
         errorText_ = errorStream_.str();
         goto unlock;
       }
     }
-  }
 
-  if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
-    state = snd_pcm_state( handle[1] );
-    if ( state != SND_PCM_STATE_PREPARED ) {
-      result = snd_pcm_prepare( handle[1] );
+    if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+      result = snd_pcm_drop( handle[1] );
       if ( result < 0 ) {
-        errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
+        errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
         errorText_ = errorStream_.str();
         goto unlock;
       }
     }
-  }
-
-  stream_.state = STREAM_RUNNING;
-
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
 
-  pthread_cond_signal( &apiInfo->runnable );
-
-  if ( result >= 0 ) return;
-  error( RtError::SYSTEM_ERROR );
-}
+  unlock:
+    MUTEX_UNLOCK( &stream_.mutex );
 
-void RtApiAlsa :: stopStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
+    if ( result >= 0 ) return;
+    error( RtError::SYSTEM_ERROR );
   }
 
-  // Change the state before the lock to improve shutdown response.
-  stream_.state = STREAM_STOPPED;
-  MUTEX_LOCK( &stream_.mutex );
+  void RtApiAlsa :: abortStream()
+  {
+    verifyStream();
+    if ( stream_.state == STREAM_STOPPED ) {
+      errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
+      error( RtError::WARNING );
+      return;
+    }
 
-  int result = 0;
-  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
-  snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-    if ( apiInfo->synchronized ) 
+    // Change the state before the lock to improve shutdown response.
+    stream_.state = STREAM_STOPPED;
+    MUTEX_LOCK( &stream_.mutex );
+
+    int result = 0;
+    AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+    snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
       result = snd_pcm_drop( handle[0] );
-    else
-      result = snd_pcm_drain( handle[0] );
-    if ( result < 0 ) {
-      errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
-      errorText_ = errorStream_.str();
-      goto unlock;
+      if ( result < 0 ) {
+        errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
+        errorText_ = errorStream_.str();
+        goto unlock;
+      }
     }
-  }
 
-  if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
-    result = snd_pcm_drop( handle[1] );
-    if ( result < 0 ) {
-      errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
-      errorText_ = errorStream_.str();
-      goto unlock;
+    if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+      result = snd_pcm_drop( handle[1] );
+      if ( result < 0 ) {
+        errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
+        errorText_ = errorStream_.str();
+        goto unlock;
+      }
     }
-  }
-
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
 
-  if ( result >= 0 ) return;
-  error( RtError::SYSTEM_ERROR );
-}
+  unlock:
+    MUTEX_UNLOCK( &stream_.mutex );
 
-void RtApiAlsa :: abortStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
+    if ( result >= 0 ) return;
+    error( RtError::SYSTEM_ERROR );
   }
 
-  // Change the state before the lock to improve shutdown response.
-  stream_.state = STREAM_STOPPED;
-  MUTEX_LOCK( &stream_.mutex );
-
-  int result = 0;
-  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
-  snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-    result = snd_pcm_drop( handle[0] );
-    if ( result < 0 ) {
-      errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
-      errorText_ = errorStream_.str();
-      goto unlock;
+  void RtApiAlsa :: callbackEvent()
+  {
+    AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+    if ( stream_.state == STREAM_STOPPED ) {
+      MUTEX_LOCK( &stream_.mutex );
+      pthread_cond_wait( &apiInfo->runnable, &stream_.mutex );
+      if ( stream_.state != STREAM_RUNNING ) {
+        MUTEX_UNLOCK( &stream_.mutex );
+        return;
+      }
+      MUTEX_UNLOCK( &stream_.mutex );
     }
-  }
 
-  if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
-    result = snd_pcm_drop( handle[1] );
-    if ( result < 0 ) {
-      errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
-      errorText_ = errorStream_.str();
-      goto unlock;
+    if ( stream_.state == STREAM_CLOSED ) {
+      errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
+      error( RtError::WARNING );
+      return;
     }
-  }
 
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  if ( result >= 0 ) return;
-  error( RtError::SYSTEM_ERROR );
-}
+    int doStopStream = 0;
+    RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+    double streamTime = getStreamTime();
+    RtAudioStreamStatus status = 0;
+    if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
+      status |= RTAUDIO_OUTPUT_UNDERFLOW;
+      apiInfo->xrun[0] = false;
+    }
+    if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
+      status |= RTAUDIO_INPUT_OVERFLOW;
+      apiInfo->xrun[1] = false;
+    }
+    doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+                             stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
 
-void RtApiAlsa :: callbackEvent()
-{
-  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
-  if ( stream_.state == STREAM_STOPPED ) {
-    MUTEX_LOCK( &stream_.mutex );
-    pthread_cond_wait( &apiInfo->runnable, &stream_.mutex );
-    if ( stream_.state != STREAM_RUNNING ) {
-      MUTEX_UNLOCK( &stream_.mutex );
+    if ( doStopStream == 2 ) {
+      abortStream();
       return;
     }
-    MUTEX_UNLOCK( &stream_.mutex );
-  }
-
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
-    error( RtError::WARNING );
-    return;
-  }
 
-  int doStopStream = 0;
-  RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
-  double streamTime = getStreamTime();
-  RtAudioStreamStatus status = 0;
-  if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
-    status |= RTAUDIO_OUTPUT_UNDERFLOW;
-    apiInfo->xrun[0] = false;
-  }
-  if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
-    status |= RTAUDIO_INPUT_OVERFLOW;
-    apiInfo->xrun[1] = false;
-  }
-  doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
-                         stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
+    MUTEX_LOCK( &stream_.mutex );
 
-  if ( doStopStream == 2 ) {
-    abortStream();
-    return;
-  }
+    // The state might change while waiting on a mutex.
+    if ( stream_.state == STREAM_STOPPED ) goto unlock;
 
-  MUTEX_LOCK( &stream_.mutex );
-
-  // The state might change while waiting on a mutex.
-  if ( stream_.state == STREAM_STOPPED ) goto unlock;
+    int result;
+    char *buffer;
+    int channels;
+    snd_pcm_t **handle;
+    snd_pcm_sframes_t frames;
+    RtAudioFormat format;
+    handle = (snd_pcm_t **) apiInfo->handles;
 
-  int result;
-  char *buffer;
-  int channels;
-  snd_pcm_t **handle;
-  snd_pcm_sframes_t frames;
-  RtAudioFormat format;
-  handle = (snd_pcm_t **) apiInfo->handles;
+    if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
 
-  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+      // Setup parameters.
+      if ( stream_.doConvertBuffer[1] ) {
+        buffer = stream_.deviceBuffer;
+        channels = stream_.nDeviceChannels[1];
+        format = stream_.deviceFormat[1];
+      }
+      else {
+        buffer = stream_.userBuffer[1];
+        channels = stream_.nUserChannels[1];
+        format = stream_.userFormat;
+      }
 
-    // Setup parameters.
-    if ( stream_.doConvertBuffer[1] ) {
-      buffer = stream_.deviceBuffer;
-      channels = stream_.nDeviceChannels[1];
-      format = stream_.deviceFormat[1];
-    }
-    else {
-      buffer = stream_.userBuffer[1];
-      channels = stream_.nUserChannels[1];
-      format = stream_.userFormat;
-    }
+      // Read samples from device in interleaved/non-interleaved format.
+      if ( stream_.deviceInterleaved[1] )
+        result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
+      else {
+        void *bufs[channels];
+        size_t offset = stream_.bufferSize * formatBytes( format );
+        for ( int i=0; i<channels; i++ )
+          bufs[i] = (void *) (buffer + (i * offset));
+        result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
+      }
 
-    // Read samples from device in interleaved/non-interleaved format.
-    if ( stream_.deviceInterleaved[1] )
-      result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
-    else {
-      void *bufs[channels];
-      size_t offset = stream_.bufferSize * formatBytes( format );
-      for ( int i=0; i<channels; i++ )
-        bufs[i] = (void *) (buffer + (i * offset));
-      result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
-    }
-
-    if ( result < (int) stream_.bufferSize ) {
-      // Either an error or overrun occured.
-      if ( result == -EPIPE ) {
-        snd_pcm_state_t state = snd_pcm_state( handle[1] );
-        if ( state == SND_PCM_STATE_XRUN ) {
-          apiInfo->xrun[1] = true;
-          result = snd_pcm_prepare( handle[1] );
-          if ( result < 0 ) {
-            errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
+      if ( result < (int) stream_.bufferSize ) {
+        // Either an error or overrun occured.
+        if ( result == -EPIPE ) {
+          snd_pcm_state_t state = snd_pcm_state( handle[1] );
+          if ( state == SND_PCM_STATE_XRUN ) {
+            apiInfo->xrun[1] = true;
+            result = snd_pcm_prepare( handle[1] );
+            if ( result < 0 ) {
+              errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
+              errorText_ = errorStream_.str();
+            }
+          }
+          else {
+            errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
             errorText_ = errorStream_.str();
           }
         }
         else {
-          errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
+          errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
           errorText_ = errorStream_.str();
         }
+        error( RtError::WARNING );
+        goto tryOutput;
       }
-      else {
-        errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
-        errorText_ = errorStream_.str();
-      }
-      error( RtError::WARNING );
-      goto tryOutput;
-    }
 
-    // Do byte swapping if necessary.
-    if ( stream_.doByteSwap[1] )
-      byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
+      // Do byte swapping if necessary.
+      if ( stream_.doByteSwap[1] )
+        byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
 
-    // Do buffer conversion if necessary.
-    if ( stream_.doConvertBuffer[1] )
-      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+      // Do buffer conversion if necessary.
+      if ( stream_.doConvertBuffer[1] )
+        convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
 
-    // Check stream latency
-    result = snd_pcm_delay( handle[1], &frames );
-    if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
-  }
+      // Check stream latency
+      result = snd_pcm_delay( handle[1], &frames );
+      if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
+    }
 
- tryOutput:
+  tryOutput:
 
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
 
-    // Setup parameters and do buffer conversion if necessary.
-    if ( stream_.doConvertBuffer[0] ) {
-      buffer = stream_.deviceBuffer;
-      convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
-      channels = stream_.nDeviceChannels[0];
-      format = stream_.deviceFormat[0];
-    }
-    else {
-      buffer = stream_.userBuffer[0];
-      channels = stream_.nUserChannels[0];
-      format = stream_.userFormat;
-    }
+      // Setup parameters and do buffer conversion if necessary.
+      if ( stream_.doConvertBuffer[0] ) {
+        buffer = stream_.deviceBuffer;
+        convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+        channels = stream_.nDeviceChannels[0];
+        format = stream_.deviceFormat[0];
+      }
+      else {
+        buffer = stream_.userBuffer[0];
+        channels = stream_.nUserChannels[0];
+        format = stream_.userFormat;
+      }
 
-    // Do byte swapping if necessary.
-    if ( stream_.doByteSwap[0] )
-      byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
+      // Do byte swapping if necessary.
+      if ( stream_.doByteSwap[0] )
+        byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
 
-    // Write samples to device in interleaved/non-interleaved format.
-    if ( stream_.deviceInterleaved[0] )
-      result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
-    else {
-      void *bufs[channels];
-      size_t offset = stream_.bufferSize * formatBytes( format );
-      for ( int i=0; i<channels; i++ )
-        bufs[i] = (void *) (buffer + (i * offset));
-      result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
-    }
-
-    if ( result < (int) stream_.bufferSize ) {
-      // Either an error or underrun occured.
-      if ( result == -EPIPE ) {
-        snd_pcm_state_t state = snd_pcm_state( handle[0] );
-        if ( state == SND_PCM_STATE_XRUN ) {
-          apiInfo->xrun[0] = true;
-          result = snd_pcm_prepare( handle[0] );
-          if ( result < 0 ) {
-            errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
+      // Write samples to device in interleaved/non-interleaved format.
+      if ( stream_.deviceInterleaved[0] )
+        result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
+      else {
+        void *bufs[channels];
+        size_t offset = stream_.bufferSize * formatBytes( format );
+        for ( int i=0; i<channels; i++ )
+          bufs[i] = (void *) (buffer + (i * offset));
+        result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
+      }
+
+      if ( result < (int) stream_.bufferSize ) {
+        // Either an error or underrun occured.
+        if ( result == -EPIPE ) {
+          snd_pcm_state_t state = snd_pcm_state( handle[0] );
+          if ( state == SND_PCM_STATE_XRUN ) {
+            apiInfo->xrun[0] = true;
+            result = snd_pcm_prepare( handle[0] );
+            if ( result < 0 ) {
+              errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
+              errorText_ = errorStream_.str();
+            }
+          }
+          else {
+            errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
             errorText_ = errorStream_.str();
           }
         }
         else {
-          errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
+          errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
           errorText_ = errorStream_.str();
         }
+        error( RtError::WARNING );
+        goto unlock;
       }
-      else {
-        errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
-        errorText_ = errorStream_.str();
-      }
-      error( RtError::WARNING );
-      goto unlock;
+
+      // Check stream latency
+      result = snd_pcm_delay( handle[0], &frames );
+      if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
     }
 
-    // Check stream latency
-    result = snd_pcm_delay( handle[0], &frames );
-    if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
-  }
+  unlock:
+    MUTEX_UNLOCK( &stream_.mutex );
 
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
+    RtApi::tickStreamTime();
+    if ( doStopStream == 1 ) this->stopStream();
+  }
 
-  RtApi::tickStreamTime();
-  if ( doStopStream == 1 ) this->stopStream();
-}
+  extern "C" void *alsaCallbackHandler( void *ptr )
+  {
+    CallbackInfo *info = (CallbackInfo *) ptr;
+    RtApiAlsa *object = (RtApiAlsa *) info->object;
+    bool *isRunning = &info->isRunning;
 
-extern "C" void *alsaCallbackHandler( void *ptr )
-{
-  CallbackInfo *info = (CallbackInfo *) ptr;
-  RtApiAlsa *object = (RtApiAlsa *) info->object;
-  bool *isRunning = &info->isRunning;
+    while ( *isRunning == true ) {
+      pthread_testcancel();
+      object->callbackEvent();
+    }
 
-  while ( *isRunning == true ) {
-    pthread_testcancel();
-    object->callbackEvent();
+    pthread_exit( NULL );
   }
 
-  pthread_exit( NULL );
-}
-
-//******************** End of __LINUX_ALSA__ *********************//
+  //******************** End of __LINUX_ALSA__ *********************//
 #endif
 
 
@@ -6099,1594 +6200,1594 @@ extern "C" void *alsaCallbackHandler( void *ptr )
 #include <errno.h>
 #include <math.h>
 
-extern "C" void *ossCallbackHandler(void * ptr);
+  extern "C" void *ossCallbackHandler(void * ptr);
 
-// A structure to hold various information related to the OSS API
-// implementation.
-struct OssHandle {
-  int id[2];    // device ids
-  bool xrun[2];
-  bool triggered;
-  pthread_cond_t runnable;
+  // A structure to hold various information related to the OSS API
+  // implementation.
+  struct OssHandle {
+    int id[2];    // device ids
+    bool xrun[2];
+    bool triggered;
+    pthread_cond_t runnable;
 
-  OssHandle()
-    :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
-};
-
-RtApiOss :: RtApiOss()
-{
-  // Nothing to do here.
-}
-
-RtApiOss :: ~RtApiOss()
-{
-  if ( stream_.state != STREAM_CLOSED ) closeStream();
-}
+    OssHandle()
+      :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
+  };
 
-unsigned int RtApiOss :: getDeviceCount( void )
-{
-  int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
-  if ( mixerfd == -1 ) {
-    errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
-    error( RtError::WARNING );
-    return 0;
+  RtApiOss :: RtApiOss()
+  {
+    // Nothing to do here.
   }
 
-  oss_sysinfo sysinfo;
-  if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
-    close( mixerfd );
-    errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
-    error( RtError::WARNING );
-    return 0;
+  RtApiOss :: ~RtApiOss()
+  {
+    if ( stream_.state != STREAM_CLOSED ) closeStream();
   }
 
-  close( mixerfd );
-  return sysinfo.numaudios;
-}
-
-RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
-{
-  RtAudio::DeviceInfo info;
-  info.probed = false;
+  unsigned int RtApiOss :: getDeviceCount( void )
+  {
+    int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+    if ( mixerfd == -1 ) {
+      errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
+      error( RtError::WARNING );
+      return 0;
+    }
 
-  int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
-  if ( mixerfd == -1 ) {
-    errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
-    error( RtError::WARNING );
-    return info;
-  }
+    oss_sysinfo sysinfo;
+    if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
+      close( mixerfd );
+      errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
+      error( RtError::WARNING );
+      return 0;
+    }
 
-  oss_sysinfo sysinfo;
-  int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
-  if ( result == -1 ) {
     close( mixerfd );
-    errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
-    error( RtError::WARNING );
-    return info;
+    return sysinfo.numaudios;
   }
 
-  unsigned nDevices = sysinfo.numaudios;
-  if ( nDevices == 0 ) {
-    close( mixerfd );
-    errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
-    error( RtError::INVALID_USE );
-  }
+  RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
+  {
+    RtAudio::DeviceInfo info;
+    info.probed = false;
 
-  if ( device >= nDevices ) {
-    close( mixerfd );
-    errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
-    error( RtError::INVALID_USE );
-  }
+    int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+    if ( mixerfd == -1 ) {
+      errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
+      error( RtError::WARNING );
+      return info;
+    }
 
-  oss_audioinfo ainfo;
-  ainfo.dev = device;
-  result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
-  close( mixerfd );
-  if ( result == -1 ) {
-    errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
+    oss_sysinfo sysinfo;
+    int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
+    if ( result == -1 ) {
+      close( mixerfd );
+      errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
+      error( RtError::WARNING );
+      return info;
+    }
 
-  // Probe channels
-  if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
-  if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
-  if ( ainfo.caps & PCM_CAP_DUPLEX ) {
-    if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
-      info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
-  }
+    unsigned nDevices = sysinfo.numaudios;
+    if ( nDevices == 0 ) {
+      close( mixerfd );
+      errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
+      error( RtError::INVALID_USE );
+    }
 
-  // Probe data formats ... do for input
-  unsigned long mask = ainfo.iformats;
-  if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
-    info.nativeFormats |= RTAUDIO_SINT16;
-  if ( mask & AFMT_S8 )
-    info.nativeFormats |= RTAUDIO_SINT8;
-  if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
-    info.nativeFormats |= RTAUDIO_SINT32;
-  if ( mask & AFMT_FLOAT )
-    info.nativeFormats |= RTAUDIO_FLOAT32;
-  if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
-    info.nativeFormats |= RTAUDIO_SINT24;
-
-  // Check that we have at least one supported format
-  if ( info.nativeFormats == 0 ) {
-    errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
+    if ( device >= nDevices ) {
+      close( mixerfd );
+      errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
+      error( RtError::INVALID_USE );
+    }
 
-  // Probe the supported sample rates.
-  info.sampleRates.clear();
-  if ( ainfo.nrates ) {
-    for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
+    oss_audioinfo ainfo;
+    ainfo.dev = device;
+    result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
+    close( mixerfd );
+    if ( result == -1 ) {
+      errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      return info;
+    }
+
+    // Probe channels
+    if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
+    if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
+    if ( ainfo.caps & PCM_CAP_DUPLEX ) {
+      if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
+        info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+    }
+
+    // Probe data formats ... do for input
+    unsigned long mask = ainfo.iformats;
+    if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
+      info.nativeFormats |= RTAUDIO_SINT16;
+    if ( mask & AFMT_S8 )
+      info.nativeFormats |= RTAUDIO_SINT8;
+    if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
+      info.nativeFormats |= RTAUDIO_SINT32;
+    if ( mask & AFMT_FLOAT )
+      info.nativeFormats |= RTAUDIO_FLOAT32;
+    if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
+      info.nativeFormats |= RTAUDIO_SINT24;
+
+    // Check that we have at least one supported format
+    if ( info.nativeFormats == 0 ) {
+      errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+      return info;
+    }
+
+    // Probe the supported sample rates.
+    info.sampleRates.clear();
+    if ( ainfo.nrates ) {
+      for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
+        for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+          if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
+            info.sampleRates.push_back( SAMPLE_RATES[k] );
+            break;
+          }
+        }
+      }
+    }
+    else {
+      // Check min and max rate values;
       for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
-        if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
+        if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] )
           info.sampleRates.push_back( SAMPLE_RATES[k] );
-          break;
-        }
       }
     }
-  }
-  else {
-    // Check min and max rate values;
-    for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
-      if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] )
-        info.sampleRates.push_back( SAMPLE_RATES[k] );
+
+    if ( info.sampleRates.size() == 0 ) {
+      errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+    }
+    else {
+      info.probed = true;
+      info.name = ainfo.name;
     }
-  }
 
-  if ( info.sampleRates.size() == 0 ) {
-    errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-  }
-  else {
-    info.probed = true;
-    info.name = ainfo.name;
+    return info;
   }
 
-  return info;
-}
 
+  bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+                                    unsigned int firstChannel, unsigned int sampleRate,
+                                    RtAudioFormat format, unsigned int *bufferSize,
+                                    RtAudio::StreamOptions *options )
+  {
+    int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+    if ( mixerfd == -1 ) {
+      errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
+      return FAILURE;
+    }
 
-bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-                                   unsigned int firstChannel, unsigned int sampleRate,
-                                  RtAudioFormat format, unsigned int *bufferSize,
-                                  RtAudio::StreamOptions *options )
-{
-  int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
-  if ( mixerfd == -1 ) {
-    errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
-    return FAILURE;
-  }
+    oss_sysinfo sysinfo;
+    int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
+    if ( result == -1 ) {
+      close( mixerfd );
+      errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
+      return FAILURE;
+    }
 
-  oss_sysinfo sysinfo;
-  int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
-  if ( result == -1 ) {
-    close( mixerfd );
-    errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
-    return FAILURE;
-  }
+    unsigned nDevices = sysinfo.numaudios;
+    if ( nDevices == 0 ) {
+      // This should not happen because a check is made before this function is called.
+      close( mixerfd );
+      errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
+      return FAILURE;
+    }
 
-  unsigned nDevices = sysinfo.numaudios;
-  if ( nDevices == 0 ) {
-    // This should not happen because a check is made before this function is called.
-    close( mixerfd );
-    errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
-    return FAILURE;
-  }
+    if ( device >= nDevices ) {
+      // This should not happen because a check is made before this function is called.
+      close( mixerfd );
+      errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
+      return FAILURE;
+    }
 
-  if ( device >= nDevices ) {
-    // This should not happen because a check is made before this function is called.
+    oss_audioinfo ainfo;
+    ainfo.dev = device;
+    result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
     close( mixerfd );
-    errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
-    return FAILURE;
-  }
+    if ( result == -1 ) {
+      errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
-  oss_audioinfo ainfo;
-  ainfo.dev = device;
-  result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
-  close( mixerfd );
-  if ( result == -1 ) {
-    errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+    // Check if device supports input or output
+    if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
+         ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
+      if ( mode == OUTPUT )
+        errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
+      else
+        errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
-  // Check if device supports input or output
-  if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
-       ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
+    int flags = 0;
+    OssHandle *handle = (OssHandle *) stream_.apiHandle;
     if ( mode == OUTPUT )
-      errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
-    else
-      errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  int flags = 0;
-  OssHandle *handle = (OssHandle *) stream_.apiHandle;
-  if ( mode == OUTPUT )
-    flags |= O_WRONLY;
-  else { // mode == INPUT
-    if (stream_.mode == OUTPUT && stream_.device[0] == device) {
-      // We just set the same device for playback ... close and reopen for duplex (OSS only).
-      close( handle->id[0] );
-      handle->id[0] = 0;
-      if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
-        errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
-        errorText_ = errorStream_.str();
-        return FAILURE;
-      }
-      // Check that the number previously set channels is the same.
-      if ( stream_.nUserChannels[0] != channels ) {
-        errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
-        errorText_ = errorStream_.str();
-        return FAILURE;
+      flags |= O_WRONLY;
+    else { // mode == INPUT
+      if (stream_.mode == OUTPUT && stream_.device[0] == device) {
+        // We just set the same device for playback ... close and reopen for duplex (OSS only).
+        close( handle->id[0] );
+        handle->id[0] = 0;
+        if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
+          errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
+          errorText_ = errorStream_.str();
+          return FAILURE;
+        }
+        // Check that the number previously set channels is the same.
+        if ( stream_.nUserChannels[0] != channels ) {
+          errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
+          errorText_ = errorStream_.str();
+          return FAILURE;
+        }
+        flags |= O_RDWR;
       }
-      flags |= O_RDWR;
+      else
+        flags |= O_RDONLY;
     }
-    else
-      flags |= O_RDONLY;
-  }
 
-  // Set exclusive access if specified.
-  if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
+    // Set exclusive access if specified.
+    if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
 
-  // Try to open the device.
-  int fd;
-  fd = open( ainfo.devnode, flags, 0 );
-  if ( fd == -1 ) {
-    if ( errno == EBUSY )
-      errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
-    else
-      errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // For duplex operation, specifically set this mode (this doesn't seem to work).
-  /*
-  if ( flags | O_RDWR ) {
-    result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
-    if ( result == -1) {
-      errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
+    // Try to open the device.
+    int fd;
+    fd = open( ainfo.devnode, flags, 0 );
+    if ( fd == -1 ) {
+      if ( errno == EBUSY )
+        errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
+      else
+        errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
       errorText_ = errorStream_.str();
       return FAILURE;
     }
-  }
-  */
 
-  // Check the device channel support.
-  stream_.nUserChannels[mode] = channels;
-  if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
-    close( fd );
-    errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Set the number of channels.
-  int deviceChannels = channels + firstChannel;
-  result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
-  if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
-    close( fd );
-    errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-  stream_.nDeviceChannels[mode] = deviceChannels;
-
-  // Get the data format mask
-  int mask;
-  result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
-  if ( result == -1 ) {
-    close( fd );
-    errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+    // For duplex operation, specifically set this mode (this doesn't seem to work).
+    /*
+      if ( flags | O_RDWR ) {
+      result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
+      if ( result == -1) {
+      errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+      }
+      }
+    */
 
-  // Determine how to set the device format.
-  stream_.userFormat = format;
-  int deviceFormat = -1;
-  stream_.doByteSwap[mode] = false;
-  if ( format == RTAUDIO_SINT8 ) {
-    if ( mask & AFMT_S8 ) {
-      deviceFormat = AFMT_S8;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT8;
-    }
-  }
-  else if ( format == RTAUDIO_SINT16 ) {
-    if ( mask & AFMT_S16_NE ) {
-      deviceFormat = AFMT_S16_NE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT16;
-    }
-    else if ( mask & AFMT_S16_OE ) {
-      deviceFormat = AFMT_S16_OE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT16;
-      stream_.doByteSwap[mode] = true;
-    }
-  }
-  else if ( format == RTAUDIO_SINT24 ) {
-    if ( mask & AFMT_S24_NE ) {
-      deviceFormat = AFMT_S24_NE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT24;
-    }
-    else if ( mask & AFMT_S24_OE ) {
-      deviceFormat = AFMT_S24_OE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT24;
-      stream_.doByteSwap[mode] = true;
-    }
-  }
-  else if ( format == RTAUDIO_SINT32 ) {
-    if ( mask & AFMT_S32_NE ) {
-      deviceFormat = AFMT_S32_NE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT32;
-    }
-    else if ( mask & AFMT_S32_OE ) {
-      deviceFormat = AFMT_S32_OE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT32;
-      stream_.doByteSwap[mode] = true;
+    // Check the device channel support.
+    stream_.nUserChannels[mode] = channels;
+    if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
+      close( fd );
+      errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
+      errorText_ = errorStream_.str();
+      return FAILURE;
     }
-  }
 
-  if ( deviceFormat == -1 ) {
-    // The user requested format is not natively supported by the device.
-    if ( mask & AFMT_S16_NE ) {
-      deviceFormat = AFMT_S16_NE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+    // Set the number of channels.
+    int deviceChannels = channels + firstChannel;
+    result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
+    if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
+      close( fd );
+      errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
     }
-    else if ( mask & AFMT_S32_NE ) {
-      deviceFormat = AFMT_S32_NE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+    stream_.nDeviceChannels[mode] = deviceChannels;
+
+    // Get the data format mask
+    int mask;
+    result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
+    if ( result == -1 ) {
+      close( fd );
+      errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
+      errorText_ = errorStream_.str();
+      return FAILURE;
     }
-    else if ( mask & AFMT_S24_NE ) {
-      deviceFormat = AFMT_S24_NE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+
+    // Determine how to set the device format.
+    stream_.userFormat = format;
+    int deviceFormat = -1;
+    stream_.doByteSwap[mode] = false;
+    if ( format == RTAUDIO_SINT8 ) {
+      if ( mask & AFMT_S8 ) {
+        deviceFormat = AFMT_S8;
+        stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+      }
     }
-    else if ( mask & AFMT_S16_OE ) {
-      deviceFormat = AFMT_S16_OE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT16;
-      stream_.doByteSwap[mode] = true;
+    else if ( format == RTAUDIO_SINT16 ) {
+      if ( mask & AFMT_S16_NE ) {
+        deviceFormat = AFMT_S16_NE;
+        stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+      }
+      else if ( mask & AFMT_S16_OE ) {
+        deviceFormat = AFMT_S16_OE;
+        stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+        stream_.doByteSwap[mode] = true;
+      }
     }
-    else if ( mask & AFMT_S32_OE ) {
-      deviceFormat = AFMT_S32_OE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT32;
-      stream_.doByteSwap[mode] = true;
+    else if ( format == RTAUDIO_SINT24 ) {
+      if ( mask & AFMT_S24_NE ) {
+        deviceFormat = AFMT_S24_NE;
+        stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+      }
+      else if ( mask & AFMT_S24_OE ) {
+        deviceFormat = AFMT_S24_OE;
+        stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+        stream_.doByteSwap[mode] = true;
+      }
     }
-    else if ( mask & AFMT_S24_OE ) {
-      deviceFormat = AFMT_S24_OE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT24;
-      stream_.doByteSwap[mode] = true;
+    else if ( format == RTAUDIO_SINT32 ) {
+      if ( mask & AFMT_S32_NE ) {
+        deviceFormat = AFMT_S32_NE;
+        stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+      }
+      else if ( mask & AFMT_S32_OE ) {
+        deviceFormat = AFMT_S32_OE;
+        stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+        stream_.doByteSwap[mode] = true;
+      }
     }
-    else if ( mask & AFMT_S8) {
-      deviceFormat = AFMT_S8;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+
+    if ( deviceFormat == -1 ) {
+      // The user requested format is not natively supported by the device.
+      if ( mask & AFMT_S16_NE ) {
+        deviceFormat = AFMT_S16_NE;
+        stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+      }
+      else if ( mask & AFMT_S32_NE ) {
+        deviceFormat = AFMT_S32_NE;
+        stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+      }
+      else if ( mask & AFMT_S24_NE ) {
+        deviceFormat = AFMT_S24_NE;
+        stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+      }
+      else if ( mask & AFMT_S16_OE ) {
+        deviceFormat = AFMT_S16_OE;
+        stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+        stream_.doByteSwap[mode] = true;
+      }
+      else if ( mask & AFMT_S32_OE ) {
+        deviceFormat = AFMT_S32_OE;
+        stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+        stream_.doByteSwap[mode] = true;
+      }
+      else if ( mask & AFMT_S24_OE ) {
+        deviceFormat = AFMT_S24_OE;
+        stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+        stream_.doByteSwap[mode] = true;
+      }
+      else if ( mask & AFMT_S8) {
+        deviceFormat = AFMT_S8;
+        stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+      }
     }
-  }
 
-  if ( stream_.deviceFormat[mode] == 0 ) {
-    // This really shouldn't happen ...
-    close( fd );
-    errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+    if ( stream_.deviceFormat[mode] == 0 ) {
+      // This really shouldn't happen ...
+      close( fd );
+      errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
-  // Set the data format.
-  int temp = deviceFormat;
-  result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
-  if ( result == -1 || deviceFormat != temp ) {
-    close( fd );
-    errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+    // Set the data format.
+    int temp = deviceFormat;
+    result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
+    if ( result == -1 || deviceFormat != temp ) {
+      close( fd );
+      errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
-  // Attempt to set the buffer size.  According to OSS, the minimum
-  // number of buffers is two.  The supposed minimum buffer size is 16
-  // bytes, so that will be our lower bound.  The argument to this
-  // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
-  // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
-  // We'll check the actual value used near the end of the setup
-  // procedure.
-  int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
-  if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
-  int buffers = 0;
-  if ( options ) buffers = options->numberOfBuffers;
-  if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
-  if ( buffers < 2 ) buffers = 3;
-  temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
-  result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
-  if ( result == -1 ) {
-    close( fd );
-    errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-  stream_.nBuffers = buffers;
+    // Attempt to set the buffer size.  According to OSS, the minimum
+    // number of buffers is two.  The supposed minimum buffer size is 16
+    // bytes, so that will be our lower bound.  The argument to this
+    // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
+    // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
+    // We'll check the actual value used near the end of the setup
+    // procedure.
+    int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
+    if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
+    int buffers = 0;
+    if ( options ) buffers = options->numberOfBuffers;
+    if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
+    if ( buffers < 2 ) buffers = 3;
+    temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
+    result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
+    if ( result == -1 ) {
+      close( fd );
+      errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+    stream_.nBuffers = buffers;
 
-  // Save buffer size (in sample frames).
-  *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
-  stream_.bufferSize = *bufferSize;
+    // Save buffer size (in sample frames).
+    *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
+    stream_.bufferSize = *bufferSize;
 
-  // Set the sample rate.
-  int srate = sampleRate;
-  result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
-  if ( result == -1 ) {
-    close( fd );
-    errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
+    // Set the sample rate.
+    int srate = sampleRate;
+    result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
+    if ( result == -1 ) {
+      close( fd );
+      errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
 
-  // Verify the sample rate setup worked.
-  if ( abs( srate - sampleRate ) > 100 ) {
-    close( fd );
-    errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-  stream_.sampleRate = sampleRate;
+    // Verify the sample rate setup worked.
+    if ( abs( srate - sampleRate ) > 100 ) {
+      close( fd );
+      errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+    stream_.sampleRate = sampleRate;
 
-  if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
-    // We're doing duplex setup here.
-    stream_.deviceFormat[0] = stream_.deviceFormat[1];
-    stream_.nDeviceChannels[0] = deviceChannels;
-  }
+    if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
+      // We're doing duplex setup here.
+      stream_.deviceFormat[0] = stream_.deviceFormat[1];
+      stream_.nDeviceChannels[0] = deviceChannels;
+    }
 
-  // Set interleaving parameters.
-  stream_.userInterleaved = true;
-  stream_.deviceInterleaved[mode] =  true;
-  if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
-    stream_.userInterleaved = false;
+    // Set interleaving parameters.
+    stream_.userInterleaved = true;
+    stream_.deviceInterleaved[mode] =  true;
+    if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
+      stream_.userInterleaved = false;
+
+    // Set flags for buffer conversion
+    stream_.doConvertBuffer[mode] = false;
+    if ( stream_.userFormat != stream_.deviceFormat[mode] )
+      stream_.doConvertBuffer[mode] = true;
+    if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+      stream_.doConvertBuffer[mode] = true;
+    if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+         stream_.nUserChannels[mode] > 1 )
+      stream_.doConvertBuffer[mode] = true;
+
+    // Allocate the stream handles if necessary and then save.
+    if ( stream_.apiHandle == 0 ) {
+      try {
+        handle = new OssHandle;
+      }
+      catch ( std::bad_alloc& ) {
+        errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
+        goto error;
+      }
 
-  // Set flags for buffer conversion
-  stream_.doConvertBuffer[mode] = false;
-  if ( stream_.userFormat != stream_.deviceFormat[mode] )
-    stream_.doConvertBuffer[mode] = true;
-  if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
-    stream_.doConvertBuffer[mode] = true;
-  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
-       stream_.nUserChannels[mode] > 1 )
-    stream_.doConvertBuffer[mode] = true;
+      if ( pthread_cond_init( &handle->runnable, NULL ) ) {
+        errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
+        goto error;
+      }
 
-  // Allocate the stream handles if necessary and then save.
-  if ( stream_.apiHandle == 0 ) {
-    try {
-      handle = new OssHandle;
+      stream_.apiHandle = (void *) handle;
     }
-    catch ( std::bad_alloc& ) {
-      errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
-      goto error;
+    else {
+      handle = (OssHandle *) stream_.apiHandle;
     }
+    handle->id[mode] = fd;
 
-    if ( pthread_cond_init( &handle->runnable, NULL ) ) {
-      errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
+    // Allocate necessary internal buffers.
+    unsigned long bufferBytes;
+    bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+    stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+    if ( stream_.userBuffer[mode] == NULL ) {
+      errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
       goto error;
     }
 
-    stream_.apiHandle = (void *) handle;
-  }
-  else {
-    handle = (OssHandle *) stream_.apiHandle;
-  }
-  handle->id[mode] = fd;
+    if ( stream_.doConvertBuffer[mode] ) {
 
-  // Allocate necessary internal buffers.
-  unsigned long bufferBytes;
-  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
-  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
-  if ( stream_.userBuffer[mode] == NULL ) {
-    errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
-    goto error;
-  }
-
-  if ( stream_.doConvertBuffer[mode] ) {
-
-    bool makeBuffer = true;
-    bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
-    if ( mode == INPUT ) {
-      if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
-        unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
-        if ( bufferBytes <= bytesOut ) makeBuffer = false;
+      bool makeBuffer = true;
+      bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+      if ( mode == INPUT ) {
+        if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+          unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+          if ( bufferBytes <= bytesOut ) makeBuffer = false;
+        }
       }
-    }
 
-    if ( makeBuffer ) {
-      bufferBytes *= *bufferSize;
-      if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
-      stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
-      if ( stream_.deviceBuffer == NULL ) {
-        errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
-        goto error;
+      if ( makeBuffer ) {
+        bufferBytes *= *bufferSize;
+        if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+        stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+        if ( stream_.deviceBuffer == NULL ) {
+          errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
+          goto error;
+        }
       }
     }
-  }
 
-  stream_.device[mode] = device;
-  stream_.state = STREAM_STOPPED;
+    stream_.device[mode] = device;
+    stream_.state = STREAM_STOPPED;
 
-  // Setup the buffer conversion information structure.
-  if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+    // Setup the buffer conversion information structure.
+    if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
 
-  // Setup thread if necessary.
-  if ( stream_.mode == OUTPUT && mode == INPUT ) {
-    // We had already set up an output stream.
-    stream_.mode = DUPLEX;
-    if ( stream_.device[0] == device ) handle->id[0] = fd;
-  }
-  else {
-    stream_.mode = mode;
+    // Setup thread if necessary.
+    if ( stream_.mode == OUTPUT && mode == INPUT ) {
+      // We had already set up an output stream.
+      stream_.mode = DUPLEX;
+      if ( stream_.device[0] == device ) handle->id[0] = fd;
+    }
+    else {
+      stream_.mode = mode;
 
-    // Setup callback thread.
-    stream_.callbackInfo.object = (void *) this;
+      // Setup callback thread.
+      stream_.callbackInfo.object = (void *) this;
 
-    // Set the thread attributes for joinable and realtime scheduling
-    // priority.  The higher priority will only take affect if the
-    // program is run as root or suid.
-    pthread_attr_t attr;
-    pthread_attr_init( &attr );
-    pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
+      // Set the thread attributes for joinable and realtime scheduling
+      // priority.  The higher priority will only take affect if the
+      // program is run as root or suid.
+      pthread_attr_t attr;
+      pthread_attr_init( &attr );
+      pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
 #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
-    if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
-      struct sched_param param;
-      int priority = options->priority;
-      int min = sched_get_priority_min( SCHED_RR );
-      int max = sched_get_priority_max( SCHED_RR );
-      if ( priority < min ) priority = min;
-      else if ( priority > max ) priority = max;
-      param.sched_priority = priority;
-      pthread_attr_setschedparam( &attr, &param );
-      pthread_attr_setschedpolicy( &attr, SCHED_RR );
-    }
-    else
-      pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+      if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+        struct sched_param param;
+        int priority = options->priority;
+        int min = sched_get_priority_min( SCHED_RR );
+        int max = sched_get_priority_max( SCHED_RR );
+        if ( priority < min ) priority = min;
+        else if ( priority > max ) priority = max;
+        param.sched_priority = priority;
+        pthread_attr_setschedparam( &attr, &param );
+        pthread_attr_setschedpolicy( &attr, SCHED_RR );
+      }
+      else
+        pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
 #else
-    pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+      pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
 #endif
 
-    stream_.callbackInfo.isRunning = true;
-    result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
-    pthread_attr_destroy( &attr );
-    if ( result ) {
-      stream_.callbackInfo.isRunning = false;
-      errorText_ = "RtApiOss::error creating callback thread!";
-      goto error;
+      stream_.callbackInfo.isRunning = true;
+      result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
+      pthread_attr_destroy( &attr );
+      if ( result ) {
+        stream_.callbackInfo.isRunning = false;
+        errorText_ = "RtApiOss::error creating callback thread!";
+        goto error;
+      }
     }
-  }
-
-  return SUCCESS;
 
- error:
-  if ( handle ) {
-    pthread_cond_destroy( &handle->runnable );
-    if ( handle->id[0] ) close( handle->id[0] );
-    if ( handle->id[1] ) close( handle->id[1] );
-    delete handle;
-    stream_.apiHandle = 0;
-  }
+    return SUCCESS;
 
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
+  error:
+    if ( handle ) {
+      pthread_cond_destroy( &handle->runnable );
+      if ( handle->id[0] ) close( handle->id[0] );
+      if ( handle->id[1] ) close( handle->id[1] );
+      delete handle;
+      stream_.apiHandle = 0;
     }
-  }
 
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
+    for ( int i=0; i<2; i++ ) {
+      if ( stream_.userBuffer[i] ) {
+        free( stream_.userBuffer[i] );
+        stream_.userBuffer[i] = 0;
+      }
+    }
 
-  return FAILURE;
-}
+    if ( stream_.deviceBuffer ) {
+      free( stream_.deviceBuffer );
+      stream_.deviceBuffer = 0;
+    }
 
-void RtApiOss :: closeStream()
-{
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiOss::closeStream(): no open stream to close!";
-    error( RtError::WARNING );
-    return;
+    return FAILURE;
   }
 
-  OssHandle *handle = (OssHandle *) stream_.apiHandle;
-  stream_.callbackInfo.isRunning = false;
-  MUTEX_LOCK( &stream_.mutex );
-  if ( stream_.state == STREAM_STOPPED )
-    pthread_cond_signal( &handle->runnable );
-  MUTEX_UNLOCK( &stream_.mutex );
-  pthread_join( stream_.callbackInfo.thread, NULL );
+  void RtApiOss :: closeStream()
+  {
+    if ( stream_.state == STREAM_CLOSED ) {
+      errorText_ = "RtApiOss::closeStream(): no open stream to close!";
+      error( RtError::WARNING );
+      return;
+    }
 
-  if ( stream_.state == STREAM_RUNNING ) {
-    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
-      ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
-    else
-      ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
-    stream_.state = STREAM_STOPPED;
-  }
+    OssHandle *handle = (OssHandle *) stream_.apiHandle;
+    stream_.callbackInfo.isRunning = false;
+    MUTEX_LOCK( &stream_.mutex );
+    if ( stream_.state == STREAM_STOPPED )
+      pthread_cond_signal( &handle->runnable );
+    MUTEX_UNLOCK( &stream_.mutex );
+    pthread_join( stream_.callbackInfo.thread, NULL );
 
-  if ( handle ) {
-    pthread_cond_destroy( &handle->runnable );
-    if ( handle->id[0] ) close( handle->id[0] );
-    if ( handle->id[1] ) close( handle->id[1] );
-    delete handle;
-    stream_.apiHandle = 0;
-  }
+    if ( stream_.state == STREAM_RUNNING ) {
+      if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
+        ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+      else
+        ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+      stream_.state = STREAM_STOPPED;
+    }
 
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
+    if ( handle ) {
+      pthread_cond_destroy( &handle->runnable );
+      if ( handle->id[0] ) close( handle->id[0] );
+      if ( handle->id[1] ) close( handle->id[1] );
+      delete handle;
+      stream_.apiHandle = 0;
     }
-  }
 
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
+    for ( int i=0; i<2; i++ ) {
+      if ( stream_.userBuffer[i] ) {
+        free( stream_.userBuffer[i] );
+        stream_.userBuffer[i] = 0;
+      }
+    }
 
-  stream_.mode = UNINITIALIZED;
-  stream_.state = STREAM_CLOSED;
-}
+    if ( stream_.deviceBuffer ) {
+      free( stream_.deviceBuffer );
+      stream_.deviceBuffer = 0;
+    }
 
-void RtApiOss :: startStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_RUNNING ) {
-    errorText_ = "RtApiOss::startStream(): the stream is already running!";
-    error( RtError::WARNING );
-    return;
+    stream_.mode = UNINITIALIZED;
+    stream_.state = STREAM_CLOSED;
   }
 
-  MUTEX_LOCK( &stream_.mutex );
+  void RtApiOss :: startStream()
+  {
+    verifyStream();
+    if ( stream_.state == STREAM_RUNNING ) {
+      errorText_ = "RtApiOss::startStream(): the stream is already running!";
+      error( RtError::WARNING );
+      return;
+    }
 
-  stream_.state = STREAM_RUNNING;
+    MUTEX_LOCK( &stream_.mutex );
 
-  // No need to do anything else here ... OSS automatically starts
-  // when fed samples.
+    stream_.state = STREAM_RUNNING;
 
-  MUTEX_UNLOCK( &stream_.mutex );
+    // No need to do anything else here ... OSS automatically starts
+    // when fed samples.
 
-  OssHandle *handle = (OssHandle *) stream_.apiHandle;
-  pthread_cond_signal( &handle->runnable );
-}
+    MUTEX_UNLOCK( &stream_.mutex );
 
-void RtApiOss :: stopStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
+    OssHandle *handle = (OssHandle *) stream_.apiHandle;
+    pthread_cond_signal( &handle->runnable );
   }
 
-  // Change the state before the lock to improve shutdown response
-  // when using a callback.
-  stream_.state = STREAM_STOPPED;
-  MUTEX_LOCK( &stream_.mutex );
+  void RtApiOss :: stopStream()
+  {
+    verifyStream();
+    if ( stream_.state == STREAM_STOPPED ) {
+      errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
+      error( RtError::WARNING );
+      return;
+    }
 
-  int result = 0;
-  OssHandle *handle = (OssHandle *) stream_.apiHandle;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+    // Change the state before the lock to improve shutdown response
+    // when using a callback.
+    stream_.state = STREAM_STOPPED;
+    MUTEX_LOCK( &stream_.mutex );
 
-    // Flush the output with zeros a few times.
-    char *buffer;
-    int samples;
-    RtAudioFormat format;
+    int result = 0;
+    OssHandle *handle = (OssHandle *) stream_.apiHandle;
+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
 
-    if ( stream_.doConvertBuffer[0] ) {
-      buffer = stream_.deviceBuffer;
-      samples = stream_.bufferSize * stream_.nDeviceChannels[0];
-      format = stream_.deviceFormat[0];
-    }
-    else {
-      buffer = stream_.userBuffer[0];
-      samples = stream_.bufferSize * stream_.nUserChannels[0];
-      format = stream_.userFormat;
-    }
+      // Flush the output with zeros a few times.
+      char *buffer;
+      int samples;
+      RtAudioFormat format;
 
-    memset( buffer, 0, samples * formatBytes(format) );
-    for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
-      result = write( handle->id[0], buffer, samples * formatBytes(format) );
+      if ( stream_.doConvertBuffer[0] ) {
+        buffer = stream_.deviceBuffer;
+        samples = stream_.bufferSize * stream_.nDeviceChannels[0];
+        format = stream_.deviceFormat[0];
+      }
+      else {
+        buffer = stream_.userBuffer[0];
+        samples = stream_.bufferSize * stream_.nUserChannels[0];
+        format = stream_.userFormat;
+      }
+
+      memset( buffer, 0, samples * formatBytes(format) );
+      for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
+        result = write( handle->id[0], buffer, samples * formatBytes(format) );
+        if ( result == -1 ) {
+          errorText_ = "RtApiOss::stopStream: audio write error.";
+          error( RtError::WARNING );
+        }
+      }
+
+      result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
       if ( result == -1 ) {
-        errorText_ = "RtApiOss::stopStream: audio write error.";
-        error( RtError::WARNING );
+        errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
+        errorText_ = errorStream_.str();
+        goto unlock;
       }
+      handle->triggered = false;
     }
 
-    result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
-    if ( result == -1 ) {
-      errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
-      errorText_ = errorStream_.str();
-      goto unlock;
+    if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
+      result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+      if ( result == -1 ) {
+        errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
+        errorText_ = errorStream_.str();
+        goto unlock;
+      }
     }
-    handle->triggered = false;
+
+  unlock:
+    MUTEX_UNLOCK( &stream_.mutex );
+
+    stream_.state = STREAM_STOPPED;
+    if ( result != -1 ) return;
+    error( RtError::SYSTEM_ERROR );
   }
 
-  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
-    result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
-    if ( result == -1 ) {
-      errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
-      errorText_ = errorStream_.str();
-      goto unlock;
+  void RtApiOss :: abortStream()
+  {
+    verifyStream();
+    if ( stream_.state == STREAM_STOPPED ) {
+      errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
+      error( RtError::WARNING );
+      return;
     }
-  }
 
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
+    // Change the state before the lock to improve shutdown response
+    // when using a callback.
+    stream_.state = STREAM_STOPPED;
+    MUTEX_LOCK( &stream_.mutex );
 
-  stream_.state = STREAM_STOPPED;
-  if ( result != -1 ) return;
-  error( RtError::SYSTEM_ERROR );
-}
+    int result = 0;
+    OssHandle *handle = (OssHandle *) stream_.apiHandle;
+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+      result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+      if ( result == -1 ) {
+        errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
+        errorText_ = errorStream_.str();
+        goto unlock;
+      }
+      handle->triggered = false;
+    }
 
-void RtApiOss :: abortStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
-  }
+    if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
+      result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+      if ( result == -1 ) {
+        errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
+        errorText_ = errorStream_.str();
+        goto unlock;
+      }
+    }
 
-  // Change the state before the lock to improve shutdown response
-  // when using a callback.
-  stream_.state = STREAM_STOPPED;
-  MUTEX_LOCK( &stream_.mutex );
+  unlock:
+    MUTEX_UNLOCK( &stream_.mutex );
 
-  int result = 0;
-  OssHandle *handle = (OssHandle *) stream_.apiHandle;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-    result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
-    if ( result == -1 ) {
-      errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-    handle->triggered = false;
+    stream_.state = STREAM_STOPPED;
+    if ( result != -1 ) return;
+    error( RtError::SYSTEM_ERROR );
   }
 
-  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
-    result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
-    if ( result == -1 ) {
-      errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
-      errorText_ = errorStream_.str();
-      goto unlock;
+  void RtApiOss :: callbackEvent()
+  {
+    OssHandle *handle = (OssHandle *) stream_.apiHandle;
+    if ( stream_.state == STREAM_STOPPED ) {
+      MUTEX_LOCK( &stream_.mutex );
+      pthread_cond_wait( &handle->runnable, &stream_.mutex );
+      if ( stream_.state != STREAM_RUNNING ) {
+        MUTEX_UNLOCK( &stream_.mutex );
+        return;
+      }
+      MUTEX_UNLOCK( &stream_.mutex );
     }
-  }
 
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  stream_.state = STREAM_STOPPED;
-  if ( result != -1 ) return;
-  error( RtError::SYSTEM_ERROR );
-}
+    if ( stream_.state == STREAM_CLOSED ) {
+      errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
+      error( RtError::WARNING );
+      return;
+    }
 
-void RtApiOss :: callbackEvent()
-{
-  OssHandle *handle = (OssHandle *) stream_.apiHandle;
-  if ( stream_.state == STREAM_STOPPED ) {
-    MUTEX_LOCK( &stream_.mutex );
-    pthread_cond_wait( &handle->runnable, &stream_.mutex );
-    if ( stream_.state != STREAM_RUNNING ) {
-      MUTEX_UNLOCK( &stream_.mutex );
+    // Invoke user callback to get fresh output data.
+    int doStopStream = 0;
+    RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+    double streamTime = getStreamTime();
+    RtAudioStreamStatus status = 0;
+    if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+      status |= RTAUDIO_OUTPUT_UNDERFLOW;
+      handle->xrun[0] = false;
+    }
+    if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+      status |= RTAUDIO_INPUT_OVERFLOW;
+      handle->xrun[1] = false;
+    }
+    doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+                             stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
+    if ( doStopStream == 2 ) {
+      this->abortStream();
       return;
     }
-    MUTEX_UNLOCK( &stream_.mutex );
-  }
 
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
-    error( RtError::WARNING );
-    return;
-  }
+    MUTEX_LOCK( &stream_.mutex );
 
-  // Invoke user callback to get fresh output data.
-  int doStopStream = 0;
-  RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
-  double streamTime = getStreamTime();
-  RtAudioStreamStatus status = 0;
-  if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
-    status |= RTAUDIO_OUTPUT_UNDERFLOW;
-    handle->xrun[0] = false;
-  }
-  if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
-    status |= RTAUDIO_INPUT_OVERFLOW;
-    handle->xrun[1] = false;
-  }
-  doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
-                           stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
-  if ( doStopStream == 2 ) {
-    this->abortStream();
-    return;
-  }
+    // The state might change while waiting on a mutex.
+    if ( stream_.state == STREAM_STOPPED ) goto unlock;
 
-  MUTEX_LOCK( &stream_.mutex );
+    int result;
+    char *buffer;
+    int samples;
+    RtAudioFormat format;
 
-  // The state might change while waiting on a mutex.
-  if ( stream_.state == STREAM_STOPPED ) goto unlock;
+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
 
-  int result;
-  char *buffer;
-  int samples;
-  RtAudioFormat format;
+      // Setup parameters and do buffer conversion if necessary.
+      if ( stream_.doConvertBuffer[0] ) {
+        buffer = stream_.deviceBuffer;
+        convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+        samples = stream_.bufferSize * stream_.nDeviceChannels[0];
+        format = stream_.deviceFormat[0];
+      }
+      else {
+        buffer = stream_.userBuffer[0];
+        samples = stream_.bufferSize * stream_.nUserChannels[0];
+        format = stream_.userFormat;
+      }
 
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+      // Do byte swapping if necessary.
+      if ( stream_.doByteSwap[0] )
+        byteSwapBuffer( buffer, samples, format );
+
+      if ( stream_.mode == DUPLEX && handle->triggered == false ) {
+        int trig = 0;
+        ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
+        result = write( handle->id[0], buffer, samples * formatBytes(format) );
+        trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
+        ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
+        handle->triggered = true;
+      }
+      else
+        // Write samples to device.
+        result = write( handle->id[0], buffer, samples * formatBytes(format) );
 
-    // Setup parameters and do buffer conversion if necessary.
-    if ( stream_.doConvertBuffer[0] ) {
-      buffer = stream_.deviceBuffer;
-      convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
-      samples = stream_.bufferSize * stream_.nDeviceChannels[0];
-      format = stream_.deviceFormat[0];
-    }
-    else {
-      buffer = stream_.userBuffer[0];
-      samples = stream_.bufferSize * stream_.nUserChannels[0];
-      format = stream_.userFormat;
+      if ( result == -1 ) {
+        // We'll assume this is an underrun, though there isn't a
+        // specific means for determining that.
+        handle->xrun[0] = true;
+        errorText_ = "RtApiOss::callbackEvent: audio write error.";
+        error( RtError::WARNING );
+        // Continue on to input section.
+      }
     }
 
-    // Do byte swapping if necessary.
-    if ( stream_.doByteSwap[0] )
-      byteSwapBuffer( buffer, samples, format );
+    if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
 
-    if ( stream_.mode == DUPLEX && handle->triggered == false ) {
-      int trig = 0;
-      ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
-      result = write( handle->id[0], buffer, samples * formatBytes(format) );
-      trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
-      ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
-      handle->triggered = true;
-    }
-    else
-      // Write samples to device.
-      result = write( handle->id[0], buffer, samples * formatBytes(format) );
-
-    if ( result == -1 ) {
-      // We'll assume this is an underrun, though there isn't a
-      // specific means for determining that.
-      handle->xrun[0] = true;
-      errorText_ = "RtApiOss::callbackEvent: audio write error.";
-      error( RtError::WARNING );
-      // Continue on to input section.
-    }
-  }
+      // Setup parameters.
+      if ( stream_.doConvertBuffer[1] ) {
+        buffer = stream_.deviceBuffer;
+        samples = stream_.bufferSize * stream_.nDeviceChannels[1];
+        format = stream_.deviceFormat[1];
+      }
+      else {
+        buffer = stream_.userBuffer[1];
+        samples = stream_.bufferSize * stream_.nUserChannels[1];
+        format = stream_.userFormat;
+      }
 
-  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+      // Read samples from device.
+      result = read( handle->id[1], buffer, samples * formatBytes(format) );
 
-    // Setup parameters.
-    if ( stream_.doConvertBuffer[1] ) {
-      buffer = stream_.deviceBuffer;
-      samples = stream_.bufferSize * stream_.nDeviceChannels[1];
-      format = stream_.deviceFormat[1];
-    }
-    else {
-      buffer = stream_.userBuffer[1];
-      samples = stream_.bufferSize * stream_.nUserChannels[1];
-      format = stream_.userFormat;
-    }
+      if ( result == -1 ) {
+        // We'll assume this is an overrun, though there isn't a
+        // specific means for determining that.
+        handle->xrun[1] = true;
+        errorText_ = "RtApiOss::callbackEvent: audio read error.";
+        error( RtError::WARNING );
+        goto unlock;
+      }
 
-    // Read samples from device.
-    result = read( handle->id[1], buffer, samples * formatBytes(format) );
+      // Do byte swapping if necessary.
+      if ( stream_.doByteSwap[1] )
+        byteSwapBuffer( buffer, samples, format );
 
-    if ( result == -1 ) {
-      // We'll assume this is an overrun, though there isn't a
-      // specific means for determining that.
-      handle->xrun[1] = true;
-      errorText_ = "RtApiOss::callbackEvent: audio read error.";
-      error( RtError::WARNING );
-      goto unlock;
+      // Do buffer conversion if necessary.
+      if ( stream_.doConvertBuffer[1] )
+        convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
     }
 
-    // Do byte swapping if necessary.
-    if ( stream_.doByteSwap[1] )
-      byteSwapBuffer( buffer, samples, format );
+  unlock:
+    MUTEX_UNLOCK( &stream_.mutex );
 
-    // Do buffer conversion if necessary.
-    if ( stream_.doConvertBuffer[1] )
-      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+    RtApi::tickStreamTime();
+    if ( doStopStream == 1 ) this->stopStream();
   }
 
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  RtApi::tickStreamTime();
-  if ( doStopStream == 1 ) this->stopStream();
-}
+  extern "C" void *ossCallbackHandler( void *ptr )
+  {
+    CallbackInfo *info = (CallbackInfo *) ptr;
+    RtApiOss *object = (RtApiOss *) info->object;
+    bool *isRunning = &info->isRunning;
 
-extern "C" void *ossCallbackHandler( void *ptr )
-{
-  CallbackInfo *info = (CallbackInfo *) ptr;
-  RtApiOss *object = (RtApiOss *) info->object;
-  bool *isRunning = &info->isRunning;
+    while ( *isRunning == true ) {
+      pthread_testcancel();
+      object->callbackEvent();
+    }
 
-  while ( *isRunning == true ) {
-    pthread_testcancel();
-    object->callbackEvent();
+    pthread_exit( NULL );
   }
 
-  pthread_exit( NULL );
-}
-
-//******************** End of __LINUX_OSS__ *********************//
+  //******************** End of __LINUX_OSS__ *********************//
 #endif
 
 
-// *************************************************** //
-//
-// Protected common (OS-independent) RtAudio methods.
-//
-// *************************************************** //
+  // *************************************************** //
+  //
+  // Protected common (OS-independent) RtAudio methods.
+  //
+  // *************************************************** //
 
-// This method can be modified to control the behavior of error
-// message printing.
-void RtApi :: error( RtError::Type type )
-{
-  errorStream_.str(""); // clear the ostringstream
-  if ( type == RtError::WARNING && showWarnings_ == true )
-    std::cerr << '\n' << errorText_ << "\n\n";
-  else
-    throw( RtError( errorText_, type ) );
-}
-
-void RtApi :: verifyStream()
-{
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApi:: a stream is not open!";
-    error( RtError::INVALID_USE );
+  // This method can be modified to control the behavior of error
+  // message printing.
+  void RtApi :: error( RtError::Type type )
+  {
+    errorStream_.str(""); // clear the ostringstream
+    if ( type == RtError::WARNING && showWarnings_ == true )
+      std::cerr << '\n' << errorText_ << "\n\n";
+    else
+      throw( RtError( errorText_, type ) );
   }
-}
 
-void RtApi :: clearStreamInfo()
-{
-  stream_.mode = UNINITIALIZED;
-  stream_.state = STREAM_CLOSED;
-  stream_.sampleRate = 0;
-  stream_.bufferSize = 0;
-  stream_.nBuffers = 0;
-  stream_.userFormat = 0;
-  stream_.userInterleaved = true;
-  stream_.streamTime = 0.0;
-  stream_.apiHandle = 0;
-  stream_.deviceBuffer = 0;
-  stream_.callbackInfo.callback = 0;
-  stream_.callbackInfo.userData = 0;
-  stream_.callbackInfo.isRunning = false;
-  for ( int i=0; i<2; i++ ) {
-    stream_.device[i] = 11111;
-    stream_.doConvertBuffer[i] = false;
-    stream_.deviceInterleaved[i] = true;
-    stream_.doByteSwap[i] = false;
-    stream_.nUserChannels[i] = 0;
-    stream_.nDeviceChannels[i] = 0;
-    stream_.channelOffset[i] = 0;
-    stream_.deviceFormat[i] = 0;
-    stream_.latency[i] = 0;
-    stream_.userBuffer[i] = 0;
-    stream_.convertInfo[i].channels = 0;
-    stream_.convertInfo[i].inJump = 0;
-    stream_.convertInfo[i].outJump = 0;
-    stream_.convertInfo[i].inFormat = 0;
-    stream_.convertInfo[i].outFormat = 0;
-    stream_.convertInfo[i].inOffset.clear();
-    stream_.convertInfo[i].outOffset.clear();
+  void RtApi :: verifyStream()
+  {
+    if ( stream_.state == STREAM_CLOSED ) {
+      errorText_ = "RtApi:: a stream is not open!";
+      error( RtError::INVALID_USE );
+    }
   }
-}
 
-unsigned int RtApi :: formatBytes( RtAudioFormat format )
-{
-  if ( format == RTAUDIO_SINT16 )
-    return 2;
-  else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 ||
-            format == RTAUDIO_FLOAT32 )
-    return 4;
-  else if ( format == RTAUDIO_FLOAT64 )
-    return 8;
-  else if ( format == RTAUDIO_SINT8 )
-    return 1;
-
-  errorText_ = "RtApi::formatBytes: undefined format.";
-  error( RtError::WARNING );
+  void RtApi :: clearStreamInfo()
+  {
+    stream_.mode = UNINITIALIZED;
+    stream_.state = STREAM_CLOSED;
+    stream_.sampleRate = 0;
+    stream_.bufferSize = 0;
+    stream_.nBuffers = 0;
+    stream_.userFormat = 0;
+    stream_.userInterleaved = true;
+    stream_.streamTime = 0.0;
+    stream_.apiHandle = 0;
+    stream_.deviceBuffer = 0;
+    stream_.callbackInfo.callback = 0;
+    stream_.callbackInfo.userData = 0;
+    stream_.callbackInfo.isRunning = false;
+    for ( int i=0; i<2; i++ ) {
+      stream_.device[i] = 11111;
+      stream_.doConvertBuffer[i] = false;
+      stream_.deviceInterleaved[i] = true;
+      stream_.doByteSwap[i] = false;
+      stream_.nUserChannels[i] = 0;
+      stream_.nDeviceChannels[i] = 0;
+      stream_.channelOffset[i] = 0;
+      stream_.deviceFormat[i] = 0;
+      stream_.latency[i] = 0;
+      stream_.userBuffer[i] = 0;
+      stream_.convertInfo[i].channels = 0;
+      stream_.convertInfo[i].inJump = 0;
+      stream_.convertInfo[i].outJump = 0;
+      stream_.convertInfo[i].inFormat = 0;
+      stream_.convertInfo[i].outFormat = 0;
+      stream_.convertInfo[i].inOffset.clear();
+      stream_.convertInfo[i].outOffset.clear();
+    }
+  }
 
-  return 0;
-}
+  unsigned int RtApi :: formatBytes( RtAudioFormat format )
+  {
+    if ( format == RTAUDIO_SINT16 )
+      return 2;
+    else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 ||
+              format == RTAUDIO_FLOAT32 )
+      return 4;
+    else if ( format == RTAUDIO_FLOAT64 )
+      return 8;
+    else if ( format == RTAUDIO_SINT8 )
+      return 1;
+
+    errorText_ = "RtApi::formatBytes: undefined format.";
+    error( RtError::WARNING );
 
-void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
-{
-  if ( mode == INPUT ) { // convert device to user buffer
-    stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
-    stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
-    stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
-    stream_.convertInfo[mode].outFormat = stream_.userFormat;
-  }
-  else { // convert user to device buffer
-    stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
-    stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
-    stream_.convertInfo[mode].inFormat = stream_.userFormat;
-    stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
+    return 0;
   }
 
-  if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
-    stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
-  else
-    stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
-
-  // Set up the interleave/deinterleave offsets.
-  if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
-    if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
-         ( mode == INPUT && stream_.userInterleaved ) ) {
-      for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
-        stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
-        stream_.convertInfo[mode].outOffset.push_back( k );
-        stream_.convertInfo[mode].inJump = 1;
-      }
+  void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
+  {
+    if ( mode == INPUT ) { // convert device to user buffer
+      stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
+      stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
+      stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
+      stream_.convertInfo[mode].outFormat = stream_.userFormat;
     }
-    else {
-      for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
-        stream_.convertInfo[mode].inOffset.push_back( k );
-        stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
-        stream_.convertInfo[mode].outJump = 1;
-      }
+    else { // convert user to device buffer
+      stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
+      stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
+      stream_.convertInfo[mode].inFormat = stream_.userFormat;
+      stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
     }
-  }
-  else { // no (de)interleaving
-    if ( stream_.userInterleaved ) {
-      for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
-        stream_.convertInfo[mode].inOffset.push_back( k );
-        stream_.convertInfo[mode].outOffset.push_back( k );
+
+    if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
+      stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
+    else
+      stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
+
+    // Set up the interleave/deinterleave offsets.
+    if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
+      if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
+           ( mode == INPUT && stream_.userInterleaved ) ) {
+        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+          stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
+          stream_.convertInfo[mode].outOffset.push_back( k );
+          stream_.convertInfo[mode].inJump = 1;
+        }
       }
-    }
-    else {
-      for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
-        stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
-        stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
-        stream_.convertInfo[mode].inJump = 1;
-        stream_.convertInfo[mode].outJump = 1;
+      else {
+        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+          stream_.convertInfo[mode].inOffset.push_back( k );
+          stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
+          stream_.convertInfo[mode].outJump = 1;
+        }
       }
     }
-  }
-
-  // Add channel offset.
-  if ( firstChannel > 0 ) {
-    if ( stream_.deviceInterleaved[mode] ) {
-      if ( mode == OUTPUT ) {
-        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
-          stream_.convertInfo[mode].outOffset[k] += firstChannel;
+    else { // no (de)interleaving
+      if ( stream_.userInterleaved ) {
+        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+          stream_.convertInfo[mode].inOffset.push_back( k );
+          stream_.convertInfo[mode].outOffset.push_back( k );
+        }
       }
       else {
-        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
-          stream_.convertInfo[mode].inOffset[k] += firstChannel;
+        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+          stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
+          stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
+          stream_.convertInfo[mode].inJump = 1;
+          stream_.convertInfo[mode].outJump = 1;
+        }
       }
     }
-    else {
-      if ( mode == OUTPUT ) {
-        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
-          stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
+
+    // Add channel offset.
+    if ( firstChannel > 0 ) {
+      if ( stream_.deviceInterleaved[mode] ) {
+        if ( mode == OUTPUT ) {
+          for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+            stream_.convertInfo[mode].outOffset[k] += firstChannel;
+        }
+        else {
+          for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+            stream_.convertInfo[mode].inOffset[k] += firstChannel;
+        }
       }
       else {
-        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
-          stream_.convertInfo[mode].inOffset[k] += ( firstChannel  * stream_.bufferSize );
+        if ( mode == OUTPUT ) {
+          for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+            stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
+        }
+        else {
+          for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+            stream_.convertInfo[mode].inOffset[k] += ( firstChannel  * stream_.bufferSize );
+        }
       }
     }
   }
-}
 
-void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
-{
-  // This function does format conversion, input/output channel compensation, and
-  // data interleaving/deinterleaving.  24-bit integers are assumed to occupy
-  // the upper three bytes of a 32-bit integer.
-
-  // Clear our device buffer when in/out duplex device channels are different
-  if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
-       ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
-    memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
-
-  int j;
-  if (info.outFormat == RTAUDIO_FLOAT64) {
-    Float64 scale;
-    Float64 *out = (Float64 *)outBuffer;
-
-    if (info.inFormat == RTAUDIO_SINT8) {
-      signed char *in = (signed char *)inBuffer;
-      scale = 1.0 / 127.5;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
-          out[info.outOffset[j]] += 0.5;
-          out[info.outOffset[j]] *= scale;
+  void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
+  {
+    // This function does format conversion, input/output channel compensation, and
+    // data interleaving/deinterleaving.  24-bit integers are assumed to occupy
+    // the upper three bytes of a 32-bit integer.
+
+    // Clear our device buffer when in/out duplex device channels are different
+    if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
+         ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
+      memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
+
+    int j;
+    if (info.outFormat == RTAUDIO_FLOAT64) {
+      Float64 scale;
+      Float64 *out = (Float64 *)outBuffer;
+
+      if (info.inFormat == RTAUDIO_SINT8) {
+        signed char *in = (signed char *)inBuffer;
+        scale = 1.0 / 127.5;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+            out[info.outOffset[j]] += 0.5;
+            out[info.outOffset[j]] *= scale;
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT16) {
-      Int16 *in = (Int16 *)inBuffer;
-      scale = 1.0 / 32767.5;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
-          out[info.outOffset[j]] += 0.5;
-          out[info.outOffset[j]] *= scale;
+      }
+      else if (info.inFormat == RTAUDIO_SINT16) {
+        Int16 *in = (Int16 *)inBuffer;
+        scale = 1.0 / 32767.5;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+            out[info.outOffset[j]] += 0.5;
+            out[info.outOffset[j]] *= scale;
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT24) {
-      Int32 *in = (Int32 *)inBuffer;
-      scale = 1.0 / 8388607.5;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]] & 0x00ffffff);
-          out[info.outOffset[j]] += 0.5;
-          out[info.outOffset[j]] *= scale;
+      }
+      else if (info.inFormat == RTAUDIO_SINT24) {
+        Int32 *in = (Int32 *)inBuffer;
+        scale = 1.0 / 8388607.5;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]] & 0x00ffffff);
+            out[info.outOffset[j]] += 0.5;
+            out[info.outOffset[j]] *= scale;
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT32) {
-      Int32 *in = (Int32 *)inBuffer;
-      scale = 1.0 / 2147483647.5;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
-          out[info.outOffset[j]] += 0.5;
-          out[info.outOffset[j]] *= scale;
+      }
+      else if (info.inFormat == RTAUDIO_SINT32) {
+        Int32 *in = (Int32 *)inBuffer;
+        scale = 1.0 / 2147483647.5;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+            out[info.outOffset[j]] += 0.5;
+            out[info.outOffset[j]] *= scale;
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT32) {
-      Float32 *in = (Float32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+      else if (info.inFormat == RTAUDIO_FLOAT32) {
+        Float32 *in = (Float32 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT64) {
-      // Channel compensation and/or (de)interleaving only.
-      Float64 *in = (Float64 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = in[info.inOffset[j]];
+      else if (info.inFormat == RTAUDIO_FLOAT64) {
+        // Channel compensation and/or (de)interleaving only.
+        Float64 *in = (Float64 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = in[info.inOffset[j]];
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
     }
-  }
-  else if (info.outFormat == RTAUDIO_FLOAT32) {
-    Float32 scale;
-    Float32 *out = (Float32 *)outBuffer;
-
-    if (info.inFormat == RTAUDIO_SINT8) {
-      signed char *in = (signed char *)inBuffer;
-      scale = (Float32) ( 1.0 / 127.5 );
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
-          out[info.outOffset[j]] += 0.5;
-          out[info.outOffset[j]] *= scale;
+    else if (info.outFormat == RTAUDIO_FLOAT32) {
+      Float32 scale;
+      Float32 *out = (Float32 *)outBuffer;
+
+      if (info.inFormat == RTAUDIO_SINT8) {
+        signed char *in = (signed char *)inBuffer;
+        scale = (Float32) ( 1.0 / 127.5 );
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+            out[info.outOffset[j]] += 0.5;
+            out[info.outOffset[j]] *= scale;
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT16) {
-      Int16 *in = (Int16 *)inBuffer;
-      scale = (Float32) ( 1.0 / 32767.5 );
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
-          out[info.outOffset[j]] += 0.5;
-          out[info.outOffset[j]] *= scale;
+      }
+      else if (info.inFormat == RTAUDIO_SINT16) {
+        Int16 *in = (Int16 *)inBuffer;
+        scale = (Float32) ( 1.0 / 32767.5 );
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+            out[info.outOffset[j]] += 0.5;
+            out[info.outOffset[j]] *= scale;
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT24) {
-      Int32 *in = (Int32 *)inBuffer;
-      scale = (Float32) ( 1.0 / 8388607.5 );
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]] & 0x00ffffff);
-          out[info.outOffset[j]] += 0.5;
-          out[info.outOffset[j]] *= scale;
+      }
+      else if (info.inFormat == RTAUDIO_SINT24) {
+        Int32 *in = (Int32 *)inBuffer;
+        scale = (Float32) ( 1.0 / 8388607.5 );
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]] & 0x00ffffff);
+            out[info.outOffset[j]] += 0.5;
+            out[info.outOffset[j]] *= scale;
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT32) {
-      Int32 *in = (Int32 *)inBuffer;
-      scale = (Float32) ( 1.0 / 2147483647.5 );
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
-          out[info.outOffset[j]] += 0.5;
-          out[info.outOffset[j]] *= scale;
+      }
+      else if (info.inFormat == RTAUDIO_SINT32) {
+        Int32 *in = (Int32 *)inBuffer;
+        scale = (Float32) ( 1.0 / 2147483647.5 );
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+            out[info.outOffset[j]] += 0.5;
+            out[info.outOffset[j]] *= scale;
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT32) {
-      // Channel compensation and/or (de)interleaving only.
-      Float32 *in = (Float32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = in[info.inOffset[j]];
+      else if (info.inFormat == RTAUDIO_FLOAT32) {
+        // Channel compensation and/or (de)interleaving only.
+        Float32 *in = (Float32 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = in[info.inOffset[j]];
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT64) {
-      Float64 *in = (Float64 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+      else if (info.inFormat == RTAUDIO_FLOAT64) {
+        Float64 *in = (Float64 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
     }
-  }
-  else if (info.outFormat == RTAUDIO_SINT32) {
-    Int32 *out = (Int32 *)outBuffer;
-    if (info.inFormat == RTAUDIO_SINT8) {
-      signed char *in = (signed char *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
-          out[info.outOffset[j]] <<= 24;
+    else if (info.outFormat == RTAUDIO_SINT32) {
+      Int32 *out = (Int32 *)outBuffer;
+      if (info.inFormat == RTAUDIO_SINT8) {
+        signed char *in = (signed char *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+            out[info.outOffset[j]] <<= 24;
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_SINT16) {
-      Int16 *in = (Int16 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
-          out[info.outOffset[j]] <<= 16;
+      else if (info.inFormat == RTAUDIO_SINT16) {
+        Int16 *in = (Int16 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+            out[info.outOffset[j]] <<= 16;
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_SINT24) {
-      Int32 *in = (Int32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
-          out[info.outOffset[j]] <<= 8;
+      else if (info.inFormat == RTAUDIO_SINT24) {
+        Int32 *in = (Int32 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+            out[info.outOffset[j]] <<= 8;
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_SINT32) {
-      // Channel compensation and/or (de)interleaving only.
-      Int32 *in = (Int32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = in[info.inOffset[j]];
+      else if (info.inFormat == RTAUDIO_SINT32) {
+        // Channel compensation and/or (de)interleaving only.
+        Int32 *in = (Int32 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = in[info.inOffset[j]];
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT32) {
-      Float32 *in = (Float32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
+      else if (info.inFormat == RTAUDIO_FLOAT32) {
+        Float32 *in = (Float32 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT64) {
-      Float64 *in = (Float64 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
+      else if (info.inFormat == RTAUDIO_FLOAT64) {
+        Float64 *in = (Float64 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
     }
-  }
-  else if (info.outFormat == RTAUDIO_SINT24) {
-    Int32 *out = (Int32 *)outBuffer;
-    if (info.inFormat == RTAUDIO_SINT8) {
-      signed char *in = (signed char *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
-          out[info.outOffset[j]] <<= 16;
+    else if (info.outFormat == RTAUDIO_SINT24) {
+      Int32 *out = (Int32 *)outBuffer;
+      if (info.inFormat == RTAUDIO_SINT8) {
+        signed char *in = (signed char *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+            out[info.outOffset[j]] <<= 16;
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_SINT16) {
-      Int16 *in = (Int16 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
-          out[info.outOffset[j]] <<= 8;
+      else if (info.inFormat == RTAUDIO_SINT16) {
+        Int16 *in = (Int16 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+            out[info.outOffset[j]] <<= 8;
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_SINT24) {
-      // Channel compensation and/or (de)interleaving only.
-      Int32 *in = (Int32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = in[info.inOffset[j]];
+      else if (info.inFormat == RTAUDIO_SINT24) {
+        // Channel compensation and/or (de)interleaving only.
+        Int32 *in = (Int32 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = in[info.inOffset[j]];
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_SINT32) {
-      Int32 *in = (Int32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
-          out[info.outOffset[j]] >>= 8;
+      else if (info.inFormat == RTAUDIO_SINT32) {
+        Int32 *in = (Int32 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+            out[info.outOffset[j]] >>= 8;
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT32) {
-      Float32 *in = (Float32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
+      else if (info.inFormat == RTAUDIO_FLOAT32) {
+        Float32 *in = (Float32 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT64) {
-      Float64 *in = (Float64 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
+      else if (info.inFormat == RTAUDIO_FLOAT64) {
+        Float64 *in = (Float64 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
     }
-  }
-  else if (info.outFormat == RTAUDIO_SINT16) {
-    Int16 *out = (Int16 *)outBuffer;
-    if (info.inFormat == RTAUDIO_SINT8) {
-      signed char *in = (signed char *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
-          out[info.outOffset[j]] <<= 8;
+    else if (info.outFormat == RTAUDIO_SINT16) {
+      Int16 *out = (Int16 *)outBuffer;
+      if (info.inFormat == RTAUDIO_SINT8) {
+        signed char *in = (signed char *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
+            out[info.outOffset[j]] <<= 8;
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_SINT16) {
-      // Channel compensation and/or (de)interleaving only.
-      Int16 *in = (Int16 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = in[info.inOffset[j]];
+      else if (info.inFormat == RTAUDIO_SINT16) {
+        // Channel compensation and/or (de)interleaving only.
+        Int16 *in = (Int16 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = in[info.inOffset[j]];
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_SINT24) {
-      Int32 *in = (Int32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 8) & 0x0000ffff);
+      else if (info.inFormat == RTAUDIO_SINT24) {
+        Int32 *in = (Int32 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 8) & 0x0000ffff);
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_SINT32) {
-      Int32 *in = (Int32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
+      else if (info.inFormat == RTAUDIO_SINT32) {
+        Int32 *in = (Int32 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT32) {
-      Float32 *in = (Float32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
+      else if (info.inFormat == RTAUDIO_FLOAT32) {
+        Float32 *in = (Float32 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT64) {
-      Float64 *in = (Float64 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
+      else if (info.inFormat == RTAUDIO_FLOAT64) {
+        Float64 *in = (Float64 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
     }
-  }
-  else if (info.outFormat == RTAUDIO_SINT8) {
-    signed char *out = (signed char *)outBuffer;
-    if (info.inFormat == RTAUDIO_SINT8) {
-      // Channel compensation and/or (de)interleaving only.
-      signed char *in = (signed char *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = in[info.inOffset[j]];
+    else if (info.outFormat == RTAUDIO_SINT8) {
+      signed char *out = (signed char *)outBuffer;
+      if (info.inFormat == RTAUDIO_SINT8) {
+        // Channel compensation and/or (de)interleaving only.
+        signed char *in = (signed char *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = in[info.inOffset[j]];
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    if (info.inFormat == RTAUDIO_SINT16) {
-      Int16 *in = (Int16 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
+      if (info.inFormat == RTAUDIO_SINT16) {
+        Int16 *in = (Int16 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_SINT24) {
-      Int32 *in = (Int32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 16) & 0x000000ff);
+      else if (info.inFormat == RTAUDIO_SINT24) {
+        Int32 *in = (Int32 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 16) & 0x000000ff);
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_SINT32) {
-      Int32 *in = (Int32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
+      else if (info.inFormat == RTAUDIO_SINT32) {
+        Int32 *in = (Int32 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT32) {
-      Float32 *in = (Float32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
+      else if (info.inFormat == RTAUDIO_FLOAT32) {
+        Float32 *in = (Float32 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT64) {
-      Float64 *in = (Float64 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
+      else if (info.inFormat == RTAUDIO_FLOAT64) {
+        Float64 *in = (Float64 *)inBuffer;
+        for (unsigned int i=0; i<stream_.bufferSize; i++) {
+          for (j=0; j<info.channels; j++) {
+            out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
+          }
+          in += info.inJump;
+          out += info.outJump;
         }
-        in += info.inJump;
-        out += info.outJump;
       }
     }
   }
-}
 
-//static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
-//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
-//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
+  //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
+  //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
+  //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
 
-void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
-{
-  register char val;
-  register char *ptr;
-
-  ptr = buffer;
-  if ( format == RTAUDIO_SINT16 ) {
-    for ( unsigned int i=0; i<samples; i++ ) {
-      // Swap 1st and 2nd bytes.
-      val = *(ptr);
-      *(ptr) = *(ptr+1);
-      *(ptr+1) = val;
-
-      // Increment 2 bytes.
-      ptr += 2;
-    }
-  }
-  else if ( format == RTAUDIO_SINT24 ||
-            format == RTAUDIO_SINT32 ||
-            format == RTAUDIO_FLOAT32 ) {
-    for ( unsigned int i=0; i<samples; i++ ) {
-      // Swap 1st and 4th bytes.
-      val = *(ptr);
-      *(ptr) = *(ptr+3);
-      *(ptr+3) = val;
-
-      // Swap 2nd and 3rd bytes.
-      ptr += 1;
-      val = *(ptr);
-      *(ptr) = *(ptr+1);
-      *(ptr+1) = val;
-
-      // Increment 3 more bytes.
-      ptr += 3;
-    }
-  }
-  else if ( format == RTAUDIO_FLOAT64 ) {
-    for ( unsigned int i=0; i<samples; i++ ) {
-      // Swap 1st and 8th bytes
-      val = *(ptr);
-      *(ptr) = *(ptr+7);
-      *(ptr+7) = val;
-
-      // Swap 2nd and 7th bytes
-      ptr += 1;
-      val = *(ptr);
-      *(ptr) = *(ptr+5);
-      *(ptr+5) = val;
-
-      // Swap 3rd and 6th bytes
-      ptr += 1;
-      val = *(ptr);
-      *(ptr) = *(ptr+3);
-      *(ptr+3) = val;
-
-      // Swap 4th and 5th bytes
-      ptr += 1;
-      val = *(ptr);
-      *(ptr) = *(ptr+1);
-      *(ptr+1) = val;
-
-      // Increment 5 more bytes.
-      ptr += 5;
+  void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
+  {
+    register char val;
+    register char *ptr;
+
+    ptr = buffer;
+    if ( format == RTAUDIO_SINT16 ) {
+      for ( unsigned int i=0; i<samples; i++ ) {
+        // Swap 1st and 2nd bytes.
+        val = *(ptr);
+        *(ptr) = *(ptr+1);
+        *(ptr+1) = val;
+
+        // Increment 2 bytes.
+        ptr += 2;
+      }
+    }
+    else if ( format == RTAUDIO_SINT24 ||
+              format == RTAUDIO_SINT32 ||
+              format == RTAUDIO_FLOAT32 ) {
+      for ( unsigned int i=0; i<samples; i++ ) {
+        // Swap 1st and 4th bytes.
+        val = *(ptr);
+        *(ptr) = *(ptr+3);
+        *(ptr+3) = val;
+
+        // Swap 2nd and 3rd bytes.
+        ptr += 1;
+        val = *(ptr);
+        *(ptr) = *(ptr+1);
+        *(ptr+1) = val;
+
+        // Increment 3 more bytes.
+        ptr += 3;
+      }
+    }
+    else if ( format == RTAUDIO_FLOAT64 ) {
+      for ( unsigned int i=0; i<samples; i++ ) {
+        // Swap 1st and 8th bytes
+        val = *(ptr);
+        *(ptr) = *(ptr+7);
+        *(ptr+7) = val;
+
+        // Swap 2nd and 7th bytes
+        ptr += 1;
+        val = *(ptr);
+        *(ptr) = *(ptr+5);
+        *(ptr+5) = val;
+
+        // Swap 3rd and 6th bytes
+        ptr += 1;
+        val = *(ptr);
+        *(ptr) = *(ptr+3);
+        *(ptr+3) = val;
+
+        // Swap 4th and 5th bytes
+        ptr += 1;
+        val = *(ptr);
+        *(ptr) = *(ptr+1);
+        *(ptr+1) = val;
+
+        // Increment 5 more bytes.
+        ptr += 5;
+      }
     }
   }
-}
 
-// Indentation settings for Vim and Emacs
-//
-// Local Variables:
-// c-basic-offset: 2
-// indent-tabs-mode: nil
-// End:
-//
-// vim: et sts=2 sw=2
+  // Indentation settings for Vim and Emacs
+  //
+  // Local Variables:
+  // c-basic-offset: 2
+  // indent-tabs-mode: nil
+  // End:
+  //
+  // vim: et sts=2 sw=2