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diff --git a/drivers/rtaudio/RtAudio.cpp b/drivers/rtaudio/RtAudio.cpp
new file mode 100644
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+++ b/drivers/rtaudio/RtAudio.cpp
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+#ifdef RTAUDIO_ENABLED
+/************************************************************************/
+/*! \class RtAudio
+    \brief Realtime audio i/o C++ classes.
+
+    RtAudio provides a common API (Application Programming Interface)
+    for realtime audio input/output across Linux (native ALSA, Jack,
+    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/
+
+    RtAudio: realtime audio i/o C++ classes
+    Copyright (c) 2001-2009 Gary P. Scavone
+
+    Permission is hereby granted, free of charge, to any person
+    obtaining a copy of this software and associated documentation files
+    (the "Software"), to deal in the Software without restriction,
+    including without limitation the rights to use, copy, modify, merge,
+    publish, distribute, sublicense, and/or sell copies of the Software,
+    and to permit persons to whom the Software is furnished to do so,
+    subject to the following conditions:
+
+    The above copyright notice and this permission notice shall be
+    included in all copies or substantial portions of the Software.
+
+    Any person wishing to distribute modifications to the Software is
+    asked to send the modifications to the original developer so that
+    they can be incorporated into the canonical version.  This is,
+    however, not a binding provision of this license.
+
+    THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+    EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+    MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+    IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+    ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+    CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+    WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+*/
+/************************************************************************/
+
+// RtAudio: Version 4.0.6
+
+#include "RtAudio.h"
+#include <iostream>
+#include <cstdlib>
+#include <cstring>
+#include <limits.h>
+
+// Static variable definitions.
+const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
+const unsigned int RtApi::SAMPLE_RATES[] = {
+  4000, 5512, 8000, 9600, 11025, 16000, 22050,
+  32000, 44100, 48000, 88200, 96000, 176400, 192000
+};
+
+#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__)
+  #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
+  #define MUTEX_DESTROY(A)    DeleteCriticalSection(A)
+  #define MUTEX_LOCK(A)       EnterCriticalSection(A)
+  #define MUTEX_UNLOCK(A)     LeaveCriticalSection(A)
+#elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
+  // pthread API
+  #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
+  #define MUTEX_DESTROY(A)    pthread_mutex_destroy(A)
+  #define MUTEX_LOCK(A)       pthread_mutex_lock(A)
+  #define MUTEX_UNLOCK(A)     pthread_mutex_unlock(A)
+#else
+  #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions
+  #define MUTEX_DESTROY(A)    abs(*A) // dummy definitions
+#endif
+
+// *************************************************** //
+//
+// RtAudio definitions.
+//
+// *************************************************** //
+
+void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()
+{
+  apis.clear();
+
+  // The order here will control the order of RtAudio's API search in
+  // the constructor.
+#if defined(__UNIX_JACK__)
+  apis.push_back( UNIX_JACK );
+#endif
+#if defined(__LINUX_ALSA__)
+  apis.push_back( LINUX_ALSA );
+#endif
+#if defined(__LINUX_OSS__)
+  apis.push_back( LINUX_OSS );
+#endif
+#if defined(__WINDOWS_ASIO__)
+  apis.push_back( WINDOWS_ASIO );
+#endif
+#if defined(__WINDOWS_DS__)
+  apis.push_back( WINDOWS_DS );
+#endif
+#if defined(__MACOSX_CORE__)
+  apis.push_back( MACOSX_CORE );
+#endif
+#if defined(__RTAUDIO_DUMMY__)
+  apis.push_back( RTAUDIO_DUMMY );
+#endif
+}
+
+void RtAudio :: openRtApi( RtAudio::Api api )
+{
+#if defined(__UNIX_JACK__)
+  if ( api == UNIX_JACK )
+    rtapi_ = new RtApiJack();
+#endif
+#if defined(__LINUX_ALSA__)
+  if ( api == LINUX_ALSA )
+    rtapi_ = new RtApiAlsa();
+#endif
+#if defined(__LINUX_OSS__)
+  if ( api == LINUX_OSS )
+    rtapi_ = new RtApiOss();
+#endif
+#if defined(__WINDOWS_ASIO__)
+  if ( api == WINDOWS_ASIO )
+    rtapi_ = new RtApiAsio();
+#endif
+#if defined(__WINDOWS_DS__)
+  if ( api == WINDOWS_DS )
+    rtapi_ = new RtApiDs();
+#endif
+#if defined(__MACOSX_CORE__)
+  if ( api == MACOSX_CORE )
+    rtapi_ = new RtApiCore();
+#endif
+#if defined(__RTAUDIO_DUMMY__)
+  if ( api == RTAUDIO_DUMMY )
+    rtapi_ = new RtApiDummy();
+#endif
+}
+
+RtAudio :: RtAudio( RtAudio::Api api ) throw()
+{
+  rtapi_ = 0;
+
+  if ( api != UNSPECIFIED ) {
+    // Attempt to open the specified API.
+    openRtApi( api );
+    if ( rtapi_ ) return;
+
+    // No compiled support for specified API value.  Issue a debug
+    // warning and continue as if no API was specified.
+    std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
+  }
+
+  // Iterate through the compiled APIs and return as soon as we find
+  // one with at least one device or we reach the end of the list.
+  std::vector< RtAudio::Api > apis;
+  getCompiledApi( apis );
+  for ( unsigned int i=0; i<apis.size(); i++ ) {
+    openRtApi( apis[i] );
+    if ( rtapi_->getDeviceCount() ) break;
+  }
+
+  if ( rtapi_ ) return;
+
+  // It should not be possible to get here because the preprocessor
+  // definition __RTAUDIO_DUMMY__ is automatically defined if no
+  // API-specific definitions are passed to the compiler. But just in
+  // case something weird happens, we'll print out an error message.
+  std::cerr << "\nRtAudio: no compiled API support found ... critical error!!\n\n";
+}
+
+RtAudio :: ~RtAudio() throw()
+{
+  delete rtapi_;
+}
+
+void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
+                            RtAudio::StreamParameters *inputParameters,
+                            RtAudioFormat format, unsigned int sampleRate,
+                            unsigned int *bufferFrames,
+                            RtAudioCallback callback, void *userData,
+                            RtAudio::StreamOptions *options )
+{
+  return rtapi_->openStream( outputParameters, inputParameters, format,
+                             sampleRate, bufferFrames, callback,
+                             userData, options );
+}
+
+// *************************************************** //
+//
+// Public RtApi definitions (see end of file for
+// private or protected utility functions).
+//
+// *************************************************** //
+
+RtApi :: RtApi()
+{
+  stream_.state = STREAM_CLOSED;
+  stream_.mode = UNINITIALIZED;
+  stream_.apiHandle = 0;
+  stream_.userBuffer[0] = 0;
+  stream_.userBuffer[1] = 0;
+  MUTEX_INITIALIZE( &stream_.mutex );
+  showWarnings_ = true;
+}
+
+RtApi :: ~RtApi()
+{
+  MUTEX_DESTROY( &stream_.mutex );
+}
+
+void RtApi :: openStream( RtAudio::StreamParameters *oParams,
+                          RtAudio::StreamParameters *iParams,
+                          RtAudioFormat format, unsigned int sampleRate,
+                          unsigned int *bufferFrames,
+                          RtAudioCallback callback, void *userData,
+                          RtAudio::StreamOptions *options )
+{
+  if ( stream_.state != STREAM_CLOSED ) {
+    errorText_ = "RtApi::openStream: a stream is already open!";
+    error( RtError::INVALID_USE );
+  }
+
+  if ( oParams && oParams->nChannels < 1 ) {
+    errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
+    error( RtError::INVALID_USE );
+  }
+
+  if ( iParams && iParams->nChannels < 1 ) {
+    errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
+    error( RtError::INVALID_USE );
+  }
+
+  if ( oParams == NULL && iParams == NULL ) {
+    errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
+    error( RtError::INVALID_USE );
+  }
+
+  if ( formatBytes(format) == 0 ) {
+    errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
+    error( RtError::INVALID_USE );
+  }
+
+  unsigned int nDevices = getDeviceCount();
+  unsigned int oChannels = 0;
+  if ( oParams ) {
+    oChannels = oParams->nChannels;
+    if ( oParams->deviceId >= nDevices ) {
+      errorText_ = "RtApi::openStream: output device parameter value is invalid.";
+      error( RtError::INVALID_USE );
+    }
+  }
+
+  unsigned int iChannels = 0;
+  if ( iParams ) {
+    iChannels = iParams->nChannels;
+    if ( iParams->deviceId >= nDevices ) {
+      errorText_ = "RtApi::openStream: input device parameter value is invalid.";
+      error( RtError::INVALID_USE );
+    }
+  }
+
+  clearStreamInfo();
+  bool result;
+
+  if ( oChannels > 0 ) {
+
+    result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
+                              sampleRate, format, bufferFrames, options );
+    if ( result == false ) error( RtError::SYSTEM_ERROR );
+  }
+
+  if ( iChannels > 0 ) {
+
+    result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
+                              sampleRate, format, bufferFrames, options );
+    if ( result == false ) {
+      if ( oChannels > 0 ) closeStream();
+      error( RtError::SYSTEM_ERROR );
+    }
+  }
+
+  stream_.callbackInfo.callback = (void *) callback;
+  stream_.callbackInfo.userData = userData;
+
+  if ( options ) options->numberOfBuffers = stream_.nBuffers;
+  stream_.state = STREAM_STOPPED;
+}
+
+unsigned int RtApi :: getDefaultInputDevice( void )
+{
+  // Should be implemented in subclasses if possible.
+  return 0;
+}
+
+unsigned int RtApi :: getDefaultOutputDevice( void )
+{
+  // Should be implemented in subclasses if possible.
+  return 0;
+}
+
+void RtApi :: closeStream( void )
+{
+  // MUST be implemented in subclasses!
+  return;
+}
+
+bool RtApi :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+                               unsigned int firstChannel, unsigned int sampleRate,
+                               RtAudioFormat format, unsigned int *bufferSize,
+                               RtAudio::StreamOptions *options )
+{
+  // MUST be implemented in subclasses!
+  return FAILURE;
+}
+
+void RtApi :: tickStreamTime( void )
+{
+  // Subclasses that do not provide their own implementation of
+  // getStreamTime should call this function once per buffer I/O to
+  // provide basic stream time support.
+
+  stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
+
+#if defined( HAVE_GETTIMEOFDAY )
+  gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
+}
+
+long RtApi :: getStreamLatency( void )
+{
+  verifyStream();
+
+  long totalLatency = 0;
+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
+    totalLatency = stream_.latency[0];
+  if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
+    totalLatency += stream_.latency[1];
+
+  return totalLatency;
+}
+
+double RtApi :: getStreamTime( void )
+{
+  verifyStream();
+
+#if defined( HAVE_GETTIMEOFDAY )
+  // Return a very accurate estimate of the stream time by
+  // adding in the elapsed time since the last tick.
+  struct timeval then;
+  struct timeval now;
+
+  if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
+    return stream_.streamTime;
+
+  gettimeofday( &now, NULL );
+  then = stream_.lastTickTimestamp;
+  return stream_.streamTime +
+    ((now.tv_sec + 0.000001 * now.tv_usec) -
+     (then.tv_sec + 0.000001 * then.tv_usec));     
+#else
+  return stream_.streamTime;
+#endif
+}
+
+unsigned int RtApi :: getStreamSampleRate( void )
+{
+ verifyStream();
+
+ return stream_.sampleRate;
+}
+
+
+// *************************************************** //
+//
+// OS/API-specific methods.
+//
+// *************************************************** //
+
+#if defined(__MACOSX_CORE__)
+
+// The OS X CoreAudio API is designed to use a separate callback
+// procedure for each of its audio devices.  A single RtAudio duplex
+// stream using two different devices is supported here, though it
+// cannot be guaranteed to always behave correctly because we cannot
+// synchronize these two callbacks.
+//
+// A property listener is installed for over/underrun information.
+// However, no functionality is currently provided to allow property
+// listeners to trigger user handlers because it is unclear what could
+// be done if a critical stream parameter (buffer size, sample rate,
+// device disconnect) notification arrived.  The listeners entail
+// quite a bit of extra code and most likely, a user program wouldn't
+// be prepared for the result anyway.  However, we do provide a flag
+// to the client callback function to inform of an over/underrun.
+//
+// The mechanism for querying and setting system parameters was
+// updated (and perhaps simplified) in OS-X version 10.4.  However,
+// since 10.4 support is not necessarily available to all users, I've
+// decided not to update the respective code at this time.  Perhaps
+// this will happen when Apple makes 10.4 free for everyone. :-)
+
+// A structure to hold various information related to the CoreAudio API
+// implementation.
+struct CoreHandle {
+  AudioDeviceID id[2];    // device ids
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+  AudioDeviceIOProcID procId[2];
+#endif
+  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;
+  int drainCounter;       // Tracks callback counts when draining
+  bool internalDrain;     // Indicates if stop is initiated from callback or not.
+
+  CoreHandle()
+    :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()
+{
+  // Nothing to do here.
+}
+
+RtApiCore :: ~RtApiCore()
+{
+  // The subclass destructor gets called before the base class
+  // destructor, so close an existing stream before deallocating
+  // apiDeviceId memory.
+  if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
+
+unsigned int RtApiCore :: getDeviceCount( void )
+{
+  // Find out how many audio devices there are, if any.
+  UInt32 dataSize;
+  OSStatus result = AudioHardwareGetPropertyInfo( kAudioHardwarePropertyDevices, &dataSize, NULL );
+  if ( result != noErr ) {
+    errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
+    error( RtError::WARNING );
+    return 0;
+  }
+
+  return dataSize / sizeof( AudioDeviceID );
+}
+
+unsigned int RtApiCore :: getDefaultInputDevice( void )
+{
+  unsigned int nDevices = getDeviceCount();
+  if ( nDevices <= 1 ) return 0;
+
+  AudioDeviceID id;
+  UInt32 dataSize = sizeof( AudioDeviceID );
+  OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultInputDevice,
+                                              &dataSize, &id );
+
+  if ( result != noErr ) {
+    errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
+    error( RtError::WARNING );
+    return 0;
+  }
+
+  dataSize *= nDevices;
+  AudioDeviceID deviceList[ nDevices ];
+  result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
+  if ( result != noErr ) {
+    errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
+    error( RtError::WARNING );
+    return 0;
+  }
+
+  for ( unsigned int i=0; i<nDevices; i++ )
+    if ( id == deviceList[i] ) return i;
+
+  errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
+  error( RtError::WARNING );
+  return 0;
+}
+
+unsigned int RtApiCore :: getDefaultOutputDevice( void )
+{
+  unsigned int nDevices = getDeviceCount();
+  if ( nDevices <= 1 ) return 0;
+
+  AudioDeviceID id;
+  UInt32 dataSize = sizeof( AudioDeviceID );
+  OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultOutputDevice,
+                                              &dataSize, &id );
+
+  if ( result != noErr ) {
+    errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
+    error( RtError::WARNING );
+    return 0;
+  }
+
+  dataSize *= nDevices;
+  AudioDeviceID deviceList[ nDevices ];
+  result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
+  if ( result != noErr ) {
+    errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
+    error( RtError::WARNING );
+    return 0;
+  }
+
+  for ( unsigned int i=0; i<nDevices; i++ )
+    if ( id == deviceList[i] ) return i;
+
+  errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
+  error( RtError::WARNING );
+  return 0;
+}
+
+RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
+{
+  RtAudio::DeviceInfo info;
+  info.probed = false;
+
+  // Get device ID
+  unsigned int nDevices = getDeviceCount();
+  if ( nDevices == 0 ) {
+    errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
+    error( RtError::INVALID_USE );
+  }
+
+  if ( device >= nDevices ) {
+    errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
+    error( RtError::INVALID_USE );
+  }
+
+  AudioDeviceID deviceList[ nDevices ];
+  UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
+  OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
+  if ( result != noErr ) {
+    errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
+    error( RtError::WARNING );
+    return info;
+  }
+
+  AudioDeviceID id = deviceList[ device ];
+
+  // Get the device name.
+  info.name.erase();
+  char	name[256];
+  dataSize = 256;
+  result = AudioDeviceGetProperty( id, 0, false,
+                                   kAudioDevicePropertyDeviceManufacturer,
+                                   &dataSize, name );
+
+  if ( result != noErr ) {
+    errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
+    errorText_ = errorStream_.str();
+    error( RtError::WARNING );
+    return info;
+  }
+  info.name.append( (const char *)name, strlen(name) );
+  info.name.append( ": " );
+
+  dataSize = 256;
+  result = AudioDeviceGetProperty( id, 0, false,
+                                   kAudioDevicePropertyDeviceName,
+                                   &dataSize, name );
+  if ( result != noErr ) {
+    errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
+    errorText_ = errorStream_.str();
+    error( RtError::WARNING );
+    return info;
+  }
+  info.name.append( (const char *)name, strlen(name) );
+
+  // Get the output stream "configuration".
+  AudioBufferList	*bufferList = nil;
+  result = AudioDeviceGetPropertyInfo( id, 0, false,
+                                       kAudioDevicePropertyStreamConfiguration,
+                                       &dataSize, NULL );
+  if (result != noErr || dataSize == 0) {
+    errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
+    errorText_ = errorStream_.str();
+    error( RtError::WARNING );
+    return info;
+  }
+
+  // Allocate the AudioBufferList.
+  bufferList = (AudioBufferList *) malloc( dataSize );
+  if ( bufferList == NULL ) {
+    errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
+    error( RtError::WARNING );
+    return info;
+  }
+
+  result = AudioDeviceGetProperty( id, 0, false,
+                                   kAudioDevicePropertyStreamConfiguration,
+                                   &dataSize, bufferList );
+  if ( result != noErr ) {
+    free( bufferList );
+    errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
+    errorText_ = errorStream_.str();
+    error( RtError::WARNING );
+    return info;
+  }
+
+  // Get output channel information.
+  unsigned int i, nStreams = bufferList->mNumberBuffers;
+  for ( i=0; i<nStreams; i++ )
+    info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
+  free( bufferList );
+
+  // Get the input stream "configuration".
+  result = AudioDeviceGetPropertyInfo( id, 0, true,
+                                       kAudioDevicePropertyStreamConfiguration,
+                                       &dataSize, NULL );
+  if (result != noErr || dataSize == 0) {
+    errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
+    errorText_ = errorStream_.str();
+    error( RtError::WARNING );
+    return info;
+  }
+
+  // Allocate the AudioBufferList.
+  bufferList = (AudioBufferList *) malloc( dataSize );
+  if ( bufferList == NULL ) {
+    errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
+    error( RtError::WARNING );
+    return info;
+  }
+
+  result = AudioDeviceGetProperty( id, 0, true,
+                                   kAudioDevicePropertyStreamConfiguration,
+                                   &dataSize, bufferList );
+  if ( result != noErr ) {
+    free( bufferList );
+    errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
+    errorText_ = errorStream_.str();
+    error( RtError::WARNING );
+    return info;
+  }
+
+  // Get input channel information.
+  nStreams = bufferList->mNumberBuffers;
+  for ( i=0; i<nStreams; i++ )
+    info.inputChannels += bufferList->mBuffers[i].mNumberChannels;
+  free( bufferList );
+
+  // 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;
+
+  // Probe the device sample rates.
+  bool isInput = false;
+  if ( info.outputChannels == 0 ) isInput = true;
+
+  // Determine the supported sample rates.
+  result = AudioDeviceGetPropertyInfo( id, 0, isInput,
+                                       kAudioDevicePropertyAvailableNominalSampleRates,
+                                       &dataSize, NULL );
+
+  if ( result != kAudioHardwareNoError || dataSize == 0 ) {
+    errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
+    errorText_ = errorStream_.str();
+    error( RtError::WARNING );
+    return info;
+  }
+
+  UInt32 nRanges = dataSize / sizeof( AudioValueRange );
+  AudioValueRange rangeList[ nRanges ];
+  result = AudioDeviceGetProperty( id, 0, isInput,
+                                   kAudioDevicePropertyAvailableNominalSampleRates,
+                                   &dataSize, &rangeList );
+
+  if ( result != kAudioHardwareNoError ) {
+    errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
+    errorText_ = errorStream_.str();
+    error( RtError::WARNING );
+    return info;
+  }
+
+  Float64 minimumRate = 100000000.0, maximumRate = 0.0;
+  for ( UInt32 i=0; i<nRanges; i++ ) {
+    if ( rangeList[i].mMinimum < minimumRate ) minimumRate = rangeList[i].mMinimum;
+    if ( rangeList[i].mMaximum > maximumRate ) maximumRate = rangeList[i].mMaximum;
+  }
+
+  info.sampleRates.clear();
+  for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+    if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate )
+      info.sampleRates.push_back( SAMPLE_RATES[k] );
+  }
+
+  if ( info.sampleRates.size() == 0 ) {
+    errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
+    errorText_ = errorStream_.str();
+    error( RtError::WARNING );
+    return info;
+  }
+
+  // CoreAudio always uses 32-bit floating point data for PCM streams.
+  // Thus, any other "physical" formats supported by the device are of
+  // no interest to the client.
+  info.nativeFormats = RTAUDIO_FLOAT32;
+
+  if ( getDefaultOutputDevice() == device )
+    info.isDefaultOutput = true;
+  if ( getDefaultInputDevice() == device )
+    info.isDefaultInput = true;
+
+  info.probed = true;
+  return info;
+}
+
+OSStatus callbackHandler( AudioDeviceID inDevice,
+                          const AudioTimeStamp* inNow,
+                          const AudioBufferList* inInputData,
+                          const AudioTimeStamp* inInputTime,
+                          AudioBufferList* outOutputData,
+                          const AudioTimeStamp* inOutputTime, 
+                          void* infoPointer )
+{
+  CallbackInfo *info = (CallbackInfo *) infoPointer;
+
+  RtApiCore *object = (RtApiCore *) info->object;
+  if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
+    return kAudioHardwareUnspecifiedError;
+  else
+    return kAudioHardwareNoError;
+}
+
+OSStatus deviceListener( AudioDeviceID inDevice,
+                         UInt32 channel,
+                         Boolean isInput,
+                         AudioDevicePropertyID propertyID,
+                         void* handlePointer )
+{
+  CoreHandle *handle = (CoreHandle *) handlePointer;
+  if ( propertyID == kAudioDeviceProcessorOverload ) {
+    if ( isInput )
+      handle->xrun[1] = true;
+    else
+      handle->xrun[0] = true;
+  }
+
+  return kAudioHardwareNoError;
+}
+
+static bool hasProperty( AudioDeviceID id, UInt32 channel, bool isInput, AudioDevicePropertyID property )
+{
+  OSStatus result = AudioDeviceGetPropertyInfo( id, channel, isInput, property, NULL, NULL );
+  return result == 0;
+}
+
+bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+                                   unsigned int firstChannel, unsigned int sampleRate,
+                                   RtAudioFormat format, unsigned int *bufferSize,
+                                   RtAudio::StreamOptions *options )
+{
+  // Get device ID
+  unsigned int nDevices = getDeviceCount();
+  if ( nDevices == 0 ) {
+    // This should not happen because a check is made before this function is called.
+    errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";
+    return FAILURE;
+  }
+
+  if ( device >= nDevices ) {
+    // This should not happen because a check is made before this function is called.
+    errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!";
+    return FAILURE;
+  }
+
+  AudioDeviceID deviceList[ nDevices ];
+  UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
+  OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
+  if ( result != noErr ) {
+    errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";
+    return FAILURE;
+  }
+
+  AudioDeviceID id = deviceList[ device ];
+
+  // Setup for stream mode.
+  bool isInput = false;
+  if ( mode == INPUT ) isInput = true;
+
+  // Set or disable "hog" mode.
+  dataSize = sizeof( UInt32 );
+  UInt32 doHog = 0;
+  if ( options && options->flags & RTAUDIO_HOG_DEVICE ) doHog = 1;
+  result = AudioHardwareSetProperty( kAudioHardwarePropertyHogModeIsAllowed, dataSize, &doHog );
+  if ( result != noErr ) {
+    errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  // Get the stream "configuration".
+  AudioBufferList	*bufferList;
+  result = AudioDeviceGetPropertyInfo( id, 0, isInput,
+                                       kAudioDevicePropertyStreamConfiguration,
+                                       &dataSize, NULL );
+  if (result != noErr || dataSize == 0) {
+    errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  // Allocate the AudioBufferList.
+  bufferList = (AudioBufferList *) malloc( dataSize );
+  if ( bufferList == NULL ) {
+    errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";
+    return FAILURE;
+  }
+
+  result = AudioDeviceGetProperty( id, 0, isInput,
+                                   kAudioDevicePropertyStreamConfiguration,
+                                   &dataSize, bufferList );
+  if ( result != noErr ) {
+    free( bufferList );
+    errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  // 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 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 ) {
+      firstStream = iStream;
+      channelOffset = offsetCounter;
+      foundStream = true;
+      break;
+    }
+    if ( streamChannels > offsetCounter ) break;
+    offsetCounter -= streamChannels;
+  }
+
+  // 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 ) {
+    monoMode = true;
+    offsetCounter = firstChannel;
+    for ( iStream=0; iStream<nStreams; iStream++ ) {
+      streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
+      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 );
+
+  // Determine the buffer size.
+  AudioValueRange	bufferRange;
+  dataSize = sizeof( AudioValueRange );
+  result = AudioDeviceGetProperty( id, 0, isInput,
+                                   kAudioDevicePropertyBufferFrameSizeRange,
+                                   &dataSize, &bufferRange );
+  if ( result != noErr ) {
+    errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;
+  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 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,
+                                   kAudioDevicePropertyBufferFrameSize,
+                                   dataSize, &theSize );
+
+  if ( result != noErr ) {
+    errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  // If attempting to setup a duplex stream, the bufferSize parameter
+  // MUST be the same in both directions!
+  *bufferSize = theSize;
+  if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
+    errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  stream_.bufferSize = *bufferSize;
+  stream_.nBuffers = 1;
+
+  // 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,
+                                   kAudioDevicePropertyStreams,
+                                   &dataSize, &streamIDs );
+  if ( result != noErr ) {
+    errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream ID(s) for device (" << device << ").";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  // Now set the stream format.  Also, check the physical format of the
+  // device and change that if necessary.
+  AudioStreamBasicDescription	description;
+  dataSize = sizeof( AudioStreamBasicDescription );
+
+  bool updateFormat;
+  for ( UInt32 i=0; i<streamCount; i++ ) {
+
+    result = AudioStreamGetProperty( streamIDs[firstStream+i], 0,
+                                     kAudioStreamPropertyVirtualFormat,
+                                     &dataSize, &description );
+
+    if ( result != noErr ) {
+      errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+
+    // Set the sample rate and data format id.  However, only make the
+    // change if the sample rate is not within 1.0 of the desired
+    // rate and the format is not linear pcm.
+    updateFormat = false;
+    if ( fabs( description.mSampleRate - (double)sampleRate ) > 1.0 ) {
+      description.mSampleRate = (double) sampleRate;
+      updateFormat = true;
+    }
+
+    if ( description.mFormatID != kAudioFormatLinearPCM ) {
+      description.mFormatID = kAudioFormatLinearPCM;
+      updateFormat = true;
+    }
+
+    if ( updateFormat ) {
+      result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0,
+                                       kAudioStreamPropertyVirtualFormat,
+                                       dataSize, &description );
+      if ( result != noErr ) {
+        errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
+    }
+
+    // Now check the physical format.
+    result = AudioStreamGetProperty( streamIDs[firstStream+i], 0,
+                                     kAudioStreamPropertyPhysicalFormat,
+                                     &dataSize, &description );
+    if ( result != noErr ) {
+      errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+
+    if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 24 ) {
+      description.mFormatID = kAudioFormatLinearPCM;
+      AudioStreamBasicDescription	testDescription = description;
+      unsigned long formatFlags;
+
+      // We'll try higher bit rates first and then work our way down.
+      testDescription.mBitsPerChannel = 32;
+      formatFlags = description.mFormatFlags | kLinearPCMFormatFlagIsFloat & ~kLinearPCMFormatFlagIsSignedInteger;
+      testDescription.mFormatFlags = formatFlags;
+      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[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
+      if ( result == noErr ) continue;
+
+      testDescription = description;
+      testDescription.mBitsPerChannel = 24;
+      testDescription.mFormatFlags = formatFlags;
+      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[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
+      if ( result == noErr ) continue;
+
+      testDescription = description;
+      testDescription.mBitsPerChannel = 8;
+      testDescription.mFormatFlags = formatFlags;
+      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();
+        return FAILURE;
+      }
+    }
+  }
+
+  // Get the stream latency.  There can be latency in both the device
+  // 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;
+  if ( hasProperty( id, channel, isInput, property ) == true ) {
+    result = AudioDeviceGetProperty( id, channel, isInput, property, &dataSize, &latency );
+    if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
+    else {
+      errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";
+      errorText_ = errorStream_.str();
+      error( RtError::WARNING );
+    }
+  }
+
+  // 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 << ").";
+    errorText_ = errorStream_.str();
+    error( RtError::WARNING );
+  }
+
+  // Byte-swapping: According to AudioHardware.h, the stream data will
+  // always be presented in native-endian format, so we should never
+  // need to byte swap.
+  stream_.doByteSwap[mode] = false;
+
+  // From the CoreAudio documentation, PCM data must be supplied as
+  // 32-bit floats.
+  stream_.userFormat = format;
+  stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+
+  if ( streamCount == 1 )
+    stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
+  else // multiple streams
+    stream_.nDeviceChannels[mode] = channels;
+  stream_.nUserChannels[mode] = channels;
+  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;
+  if ( stream_.userFormat != stream_.deviceFormat[mode] )
+    stream_.doConvertBuffer[mode] = true;
+  if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+    stream_.doConvertBuffer[mode] = true;
+  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;
+
+  // Allocate our CoreHandle structure for the stream.
+  CoreHandle *handle = 0;
+  if ( stream_.apiHandle == 0 ) {
+    try {
+      handle = new CoreHandle;
+    }
+    catch ( std::bad_alloc& ) {
+      errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";
+      goto error;
+    }
+
+    if ( pthread_cond_init( &handle->condition, NULL ) ) {
+      errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";
+      goto error;
+    }
+    stream_.apiHandle = (void *) handle;
+  }
+  else
+    handle = (CoreHandle *) stream_.apiHandle;
+  handle->iStream[mode] = firstStream;
+  handle->nStreams[mode] = streamCount;
+  handle->id[mode] = id;
+
+  // 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_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";
+    goto error;
+  }
+
+  // If possible, we will make use of the CoreAudio stream buffers as
+  // "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] );
+    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_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
+        goto error;
+      }
+    }
+  }
+
+  stream_.sampleRate = sampleRate;
+  stream_.device[mode] = device;
+  stream_.state = STREAM_STOPPED;
+  stream_.callbackInfo.object = (void *) this;
+
+  // Setup the buffer conversion information structure.
+  if ( stream_.doConvertBuffer[mode] ) {
+    if ( streamCount > 1 ) setConvertInfo( mode, 0 );
+    else setConvertInfo( mode, channelOffset );
+  }
+
+  if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
+    // Only one callback procedure per device.
+    stream_.mode = DUPLEX;
+  else {
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+    result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );
+#else
+    // deprecated in favor of AudioDeviceCreateIOProcID()
+    result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
+#endif
+    if ( result != noErr ) {
+      errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";
+      errorText_ = errorStream_.str();
+      goto error;
+    }
+    if ( stream_.mode == OUTPUT && mode == INPUT )
+      stream_.mode = DUPLEX;
+    else
+      stream_.mode = mode;
+  }
+
+  // Setup the device property listener for over/underload.
+  result = AudioDeviceAddPropertyListener( id, 0, isInput,
+                                           kAudioDeviceProcessorOverload,
+                                           deviceListener, (void *) handle );
+
+  return SUCCESS;
+
+ error:
+  if ( handle ) {
+    pthread_cond_destroy( &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;
+  }
+
+  return FAILURE;
+}
+
+void RtApiCore :: closeStream( void )
+{
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiCore::closeStream(): no open stream to close!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+    if ( stream_.state == STREAM_RUNNING )
+      AudioDeviceStop( handle->id[0], callbackHandler );
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+    AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
+#else
+    // deprecated in favor of AudioDeviceDestroyIOProcID()
+    AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
+#endif
+  }
+
+  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+    if ( stream_.state == STREAM_RUNNING )
+      AudioDeviceStop( handle->id[1], callbackHandler );
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+    AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
+#else
+    // deprecated in favor of AudioDeviceDestroyIOProcID()
+    AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
+#endif
+  }
+
+  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;
+  }
+
+  // Destroy pthread condition variable.
+  pthread_cond_destroy( &handle->condition );
+  delete handle;
+  stream_.apiHandle = 0;
+
+  stream_.mode = UNINITIALIZED;
+  stream_.state = STREAM_CLOSED;
+}
+
+void RtApiCore :: startStream( void )
+{
+  verifyStream();
+  if ( stream_.state == STREAM_RUNNING ) {
+    errorText_ = "RtApiCore::startStream(): the stream is already running!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  OSStatus result = noErr;
+  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+    result = AudioDeviceStart( handle->id[0], callbackHandler );
+    if ( result != noErr ) {
+      errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
+      errorText_ = errorStream_.str();
+      goto unlock;
+    }
+  }
+
+  if ( stream_.mode == INPUT ||
+       ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+
+    result = AudioDeviceStart( handle->id[1], callbackHandler );
+    if ( result != noErr ) {
+      errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
+      errorText_ = errorStream_.str();
+      goto unlock;
+    }
+  }
+
+  handle->drainCounter = 0;
+  handle->internalDrain = false;
+  stream_.state = STREAM_RUNNING;
+
+ unlock:
+  MUTEX_UNLOCK( &stream_.mutex );
+
+  if ( result == noErr ) return;
+  error( RtError::SYSTEM_ERROR );
+}
+
+void RtApiCore :: stopStream( void )
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  if ( stream_.state == STREAM_STOPPED ) {
+    MUTEX_UNLOCK( &stream_.mutex );
+    return;
+  }
+
+  OSStatus result = noErr;
+  CoreHandle *handle = (CoreHandle *) 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
+    }
+
+    result = AudioDeviceStop( handle->id[0], callbackHandler );
+    if ( result != noErr ) {
+      errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
+      errorText_ = errorStream_.str();
+      goto unlock;
+    }
+  }
+
+  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+
+    result = AudioDeviceStop( handle->id[1], callbackHandler );
+    if ( result != noErr ) {
+      errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
+      errorText_ = errorStream_.str();
+      goto unlock;
+    }
+  }
+
+  stream_.state = STREAM_STOPPED;
+
+ unlock:
+  MUTEX_UNLOCK( &stream_.mutex );
+
+  if ( result == noErr ) return;
+  error( RtError::SYSTEM_ERROR );
+}
+
+void RtApiCore :: abortStream( void )
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+  handle->drainCounter = 1;
+
+  stopStream();
+}
+
+bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
+                                 const AudioBufferList *inBufferList,
+                                 const AudioBufferList *outBufferList )
+{
+  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;
+  }
+
+  CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+  CoreHandle *handle = (CoreHandle *) 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;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  // The state might change while waiting on a mutex.
+  if ( stream_.state == STREAM_STOPPED ) {
+    MUTEX_UNLOCK( &stream_.mutex );
+    return SUCCESS;
+  }
+
+  AudioDeviceID outputDevice = handle->id[0];
+
+  // Invoke user callback to get fresh output data UNLESS we are
+  // draining stream or duplex mode AND the input/output devices are
+  // different AND this function is called for the input device.
+  if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
+    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;
+  }
+
+  if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
+
+    if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+
+      if ( handle->nStreams[0] == 1 ) {
+        memset( outBufferList->mBuffers[handle->iStream[0]].mData,
+                0,
+                outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
+      }
+      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 ( 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 { // copy from user buffer
+        memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
+                stream_.userBuffer[0],
+                outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
+      }
+    }
+    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_.nUserChannels[0]; i++ ) {
+          memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
+                  (void *)&inBuffer[i*stream_.bufferSize], 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 and not mono mode
+          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;
+        }
+      }
+    }
+
+    if ( handle->drainCounter ) {
+      handle->drainCounter++;
+      goto unlock;
+    }
+  }
+
+  AudioDeviceID inputDevice;
+  inputDevice = handle->id[1];
+  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
+
+    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] == false ) { // mono mode
+        UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
+        for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+          memcpy( (void *)&outBuffer[i*stream_.bufferSize],
+                  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 and not mono mode
+          outChannels = stream_.nDeviceChannels[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;
+          }
+
+          // 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] );
+      }
+    }
+  }
+
+ unlock:
+  MUTEX_UNLOCK( &stream_.mutex );
+
+  RtApi::tickStreamTime();
+  return SUCCESS;
+}
+
+const char* RtApiCore :: getErrorCode( OSStatus code )
+{
+  switch( code ) {
+
+  case kAudioHardwareNotRunningError:
+    return "kAudioHardwareNotRunningError";
+
+  case kAudioHardwareUnspecifiedError:
+    return "kAudioHardwareUnspecifiedError";
+
+  case kAudioHardwareUnknownPropertyError:
+    return "kAudioHardwareUnknownPropertyError";
+
+  case kAudioHardwareBadPropertySizeError:
+    return "kAudioHardwareBadPropertySizeError";
+
+  case kAudioHardwareIllegalOperationError:
+    return "kAudioHardwareIllegalOperationError";
+
+  case kAudioHardwareBadObjectError:
+    return "kAudioHardwareBadObjectError";
+
+  case kAudioHardwareBadDeviceError:
+    return "kAudioHardwareBadDeviceError";
+
+  case kAudioHardwareBadStreamError:
+    return "kAudioHardwareBadStreamError";
+
+  case kAudioHardwareUnsupportedOperationError:
+    return "kAudioHardwareUnsupportedOperationError";
+
+  case kAudioDeviceUnsupportedFormatError:
+    return "kAudioDeviceUnsupportedFormatError";
+
+  case kAudioDevicePermissionsError:
+    return "kAudioDevicePermissionsError";
+
+  default:
+    return "CoreAudio unknown error";
+  }
+}
+
+  //******************** 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.
+
+#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; }
+};
+
+void jackSilentError( const char * ) {};
+
+RtApiJack :: RtApiJack()
+{
+  // Nothing to do here.
+#if !defined(__RTAUDIO_DEBUG__)
+  // Turn off Jack's internal error reporting.
+  jack_set_error_function( &jackSilentError );
+#endif
+}
+
+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;
+        }
+      }
+    } while ( ports[++nChannels] );
+    free( ports );
+  }
+
+  jack_client_close( client );
+  return nDevices;
+}
+
+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;
+  }
+
+  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 );
+  }
+
+  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;
+  }
+
+  // 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 ( info.outputChannels == 0 && info.inputChannels == 0 ) {
+    jack_client_close(client);
+    errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
+    error( RtError::WARNING );
+    return info;
+  }
+
+  // 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 always uses 32-bit floats.
+  info.nativeFormats = RTAUDIO_FLOAT32;
+
+  // 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_client_close(client);
+  info.probed = true;
+  return info;
+}
+
+int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
+{
+  CallbackInfo *info = (CallbackInfo *) infoPointer;
+
+  RtApiJack *object = (RtApiJack *) info->object;
+  if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
+
+  return 0;
+}
+
+void jackShutdown( void *infoPointer )
+{
+  CallbackInfo *info = (CallbackInfo *) infoPointer;
+  RtApiJack *object = (RtApiJack *) info->object;
+
+  // 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;
+
+  object->closeStream();
+  std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
+}
+
+int jackXrun( void *infoPointer )
+{
+  JackHandle *handle = (JackHandle *) infoPointer;
+
+  if ( handle->ports[0] ) handle->xrun[0] = true;
+  if ( handle->ports[1] ) handle->xrun[1] = true;
+
+  return 0;
+}
+
+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;
+        }
+      }
+    } while ( ports[++nPorts] );
+    free( ports );
+  }
+
+  if ( device >= nDevices ) {
+    errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
+    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 );
+  }
+
+  // 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;
+  }
+
+  // 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;
+
+  // 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 );
+
+  // The jack server always uses 32-bit floating-point data.
+  stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+  stream_.userFormat = format;
+
+  if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+  else stream_.userInterleaved = true;
+
+  // Jack always uses non-interleaved buffers.
+  stream_.deviceInterleaved[mode] = false;
+
+  // Jack always provides host byte-ordered data.
+  stream_.doByteSwap[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;
+
+  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;
+    }
+
+    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;
+
+  // 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] ) {
+
+    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;
+      }
+    }
+  }
+
+  // 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;
+
+  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 );
+    }
+  }
+  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 );
+
+  return SUCCESS;
+
+ 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] );
+
+    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;
+  }
+
+  return FAILURE;
+}
+
+void RtApiJack :: closeStream( void )
+{
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiJack::closeStream(): no open stream to close!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  JackHandle *handle = (JackHandle *) stream_.apiHandle;
+  if ( handle ) {
+
+    if ( stream_.state == STREAM_RUNNING )
+      jack_deactivate( handle->client );
+
+    jack_client_close( 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;
+  }
+
+  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;
+  }
+
+  stream_.mode = UNINITIALIZED;
+  stream_.state = STREAM_CLOSED;
+}
+
+void RtApiJack :: startStream( void )
+{
+  verifyStream();
+  if ( stream_.state == STREAM_RUNNING ) {
+    errorText_ = "RtApiJack::startStream(): the stream is already running!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  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;
+  }
+
+  const char **ports;
+
+  // 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!";
+      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);
+  }
+
+  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.  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);
+  }
+
+  handle->drainCounter = 0;
+  handle->internalDrain = false;
+  stream_.state = STREAM_RUNNING;
+
+ 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;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  if ( stream_.state == STREAM_STOPPED ) {
+    MUTEX_UNLOCK( &stream_.mutex );
+    return;
+  }
+
+  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
+    }
+  }
+
+  jack_deactivate( handle->client );
+  stream_.state = STREAM_STOPPED;
+
+  MUTEX_UNLOCK( &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;
+  handle->drainCounter = 1;
+
+  stopStream();
+}
+
+// This function will be called by a spawned thread when the user
+// callback function signals that the stream should be stopped or
+// aborted.  It is necessary to handle it this way because the
+// callbackEvent() function must return before the jack_deactivate()
+// function will return.
+extern "C" void *jackStopStream( void *ptr )
+{
+  CallbackInfo *info = (CallbackInfo *) ptr;
+  RtApiJack *object = (RtApiJack *) info->object;
+
+  object->stopStream();
+
+  pthread_exit( NULL );
+}
+
+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;
+  }
+
+  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 == true ) {
+      ThreadHandle id;
+      pthread_create( &id, NULL, jackStopStream, info );
+    }
+    else
+      pthread_cond_signal( &handle->condition );
+    return SUCCESS;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  // The state might change while waiting on a mutex.
+  if ( stream_.state == STREAM_STOPPED ) {
+    MUTEX_UNLOCK( &stream_.mutex );
+    return SUCCESS;
+  }
+
+  // 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 );
+      ThreadHandle id;
+      pthread_create( &id, NULL, jackStopStream, info );
+      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
+
+      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 );
+      }
+
+    }
+    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 ( 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 );
+      }
+      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 );
+      }
+    }
+  }
+
+ unlock:
+  MUTEX_UNLOCK(&stream_.mutex);
+
+  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.
+
+#include "asiosys.h"
+#include "asio.h"
+#include "iasiothiscallresolver.h"
+#include "asiodrivers.h"
+#include <cmath>
+
+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;
+
+  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 );
+
+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();
+}
+
+RtApiAsio :: ~RtApiAsio()
+{
+  if ( stream_.state != STREAM_CLOSED ) closeStream();
+  if ( coInitialized_ ) CoUninitialize();
+}
+
+unsigned int RtApiAsio :: getDeviceCount( void )
+{
+  return (unsigned int) drivers.asioGetNumDev();
+}
+
+RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
+{
+  RtAudio::DeviceInfo info;
+  info.probed = false;
+
+  // Get device ID
+  unsigned int nDevices = getDeviceCount();
+  if ( nDevices == 0 ) {
+    errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
+    error( RtError::INVALID_USE );
+  }
+
+  if ( device >= nDevices ) {
+    errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
+    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 ];
+  }
+
+  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;
+  }
+
+  info.name = driverName;
+
+  if ( !drivers.loadDriver( driverName ) ) {
+    errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
+    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;
+  }
+
+  // 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;
+  }
+
+  info.outputChannels = outputChannels;
+  info.inputChannels = inputChannels;
+  if ( info.outputChannels > 0 && info.inputChannels > 0 )
+    info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+
+  // 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 ) {
+    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;
+  }
+
+  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;
+  }
+
+  // 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();
+
+  // 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;
+    }
+
+    result = ASIOInit( &driverInfo );
+    if ( result != ASE_OK ) {
+      errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
+      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;
+
+  // 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;
+  }
+
+  // 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 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;
+    }
+  }
+
+  // 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;
+  }
+
+  // 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;
+  }
+
+  if ( stream_.deviceFormat[mode] == 0 ) {
+    drivers.removeCurrentDriver();
+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
+    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;
+  }
+
+  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;
+
+    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;
+    }
+
+    long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
+    int min_delta_num = log2_of_min_size;
+
+    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;
+      }
+    }
+
+    *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 ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) {
+    drivers.removeCurrentDriver();
+    errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
+    return FAILURE;
+  }
+
+  stream_.bufferSize = *bufferSize;
+  stream_.nBuffers = 2;
+
+  if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+  else stream_.userInterleaved = true;
+
+  // ASIO always uses non-interleaved buffers.
+  stream_.deviceInterleaved[mode] = false;
+
+  // 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;
+  }
+
+  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;
+
+  // 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] ) {
+
+    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;
+      }
+    }
+  }
+
+  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;
+  }
+
+  // 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;
+
+ error:
+  if ( buffersAllocated )
+    ASIODisposeBuffers();
+  drivers.removeCurrentDriver();
+
+  if ( handle ) {
+    CloseHandle( handle->condition );
+    if ( handle->bufferInfos )
+      free( handle->bufferInfos );
+    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;
+  }
+
+  return FAILURE;
+}
+
+void RtApiAsio :: closeStream()
+{
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  if ( stream_.state == STREAM_RUNNING ) {
+    stream_.state = STREAM_STOPPED;
+    ASIOStop();
+  }
+  ASIODisposeBuffers();
+  drivers.removeCurrentDriver();
+
+  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+  if ( handle ) {
+    CloseHandle( handle->condition );
+    if ( handle->bufferInfos )
+      free( handle->bufferInfos );
+    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;
+  }
+
+  stream_.mode = UNINITIALIZED;
+  stream_.state = STREAM_CLOSED;
+}
+
+void RtApiAsio :: startStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_RUNNING ) {
+    errorText_ = "RtApiAsio::startStream(): the stream is already running!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  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;
+  }
+
+  handle->drainCounter = 0;
+  handle->internalDrain = false;
+  stream_.state = STREAM_RUNNING;
+  asioXRun = false;
+
+ unlock:
+  MUTEX_UNLOCK( &stream_.mutex );
+
+  if ( result == ASE_OK ) return;
+  error( RtError::SYSTEM_ERROR );
+}
+
+void RtApiAsio :: stopStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  if ( stream_.state == STREAM_STOPPED ) {
+    MUTEX_UNLOCK( &stream_.mutex );
+    return;
+  }
+
+  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 );
+    }
+  }
+
+  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 );
+
+  if ( result == ASE_OK ) return;
+  error( RtError::SYSTEM_ERROR );
+}
+
+void RtApiAsio :: abortStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  // 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 completely
+  // disposed.  So now, calling abort is the same as calling stop.
+  // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+  // handle->drainCounter = 1;
+  stopStream();
+}
+
+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;
+  }
+
+  CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+
+  // 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;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  // The state might change while waiting on a mutex.
+  if ( stream_.state == STREAM_STOPPED ) goto unlock;
+
+  // 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;
+  }
+
+  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] );
+
+    if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+
+      for ( i=0, j=0; i<nChannels; i++ ) {
+        if ( handle->bufferInfos[i].isInput != ASIOTrue )
+          memset( handle->bufferInfos[i].buffers[bufferIndex], 0, 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 );
+      }
+
+    }
+    else {
+
+      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 )
+          memcpy( handle->bufferInfos[i].buffers[bufferIndex],
+                  &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
+      }
+
+    }
+
+    if ( handle->drainCounter ) {
+      handle->drainCounter++;
+      goto unlock;
+    }
+  }
+
+  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+    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 );
+    }
+  }
+
+ 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 );
+
+  RtApi::tickStreamTime();
+  return SUCCESS;
+}
+
+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;
+  }
+
+  std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
+}
+
+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;
+}
+
+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
+
+#include <dsound.h>
+#include <assert.h>
+
+#if defined(__MINGW32__)
+  // 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 */
+#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
+#endif
+
+#define MINIMUM_DEVICE_BUFFER_SIZE 32768
+
+#ifdef _MSC_VER // if Microsoft Visual C++
+#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;
+
+  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;
+
+// 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 )
+    s.latency += s.readDeviceSafeLeadBytes * 1.0 / s.inputFrameSize / s.sampleRate;
+
+  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 );
+  }
+
+  // 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 );
+  }
+
+  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();
+    error( RtError::WARNING );
+  }
+
+  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;
+
+  // 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] );
+  }
+
+  // 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 );
+  }
+
+  if ( dsinfo.name.empty() ) return info;
+
+  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;
+  }
+
+  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;
+  }
+
+  // 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;
+  }
+
+  // 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 ( 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;
+    }
+  }
+
+  // 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;
+    }
+
+    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 {
+      waveFormat.wBitsPerSample = 8;
+      stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+    }
+    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;
+    }
+
+    // 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;
+    }
+
+    // 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;
+    }
+
+    // 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 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;
+    }
+
+    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;
+    }
+
+    // Zero the DS buffer
+    ZeroMemory( audioPtr, dataLen );
+
+    // 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;
+  }
+
+  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();
+      return FAILURE;
+    }
+
+    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;
+    }
+
+    // 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;
+    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;
+
+    // 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;
+    }
+
+    // Get the buffer size ... might be different from what we specified.
+    DSCBCAPS dscbcaps;
+    dscbcaps.dwSize = sizeof( DSCBCAPS );
+    result = buffer->GetCaps( &dscbcaps );
+    if ( FAILED( result ) ) {
+      input->Release();
+      buffer->Release();
+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsinfo.name << ")!";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+
+    bufferBytes = dscbcaps.dwBufferBytes;
+
+    // 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;
+  }
+
+  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 <= (long) bytesOut ) makeBuffer = false;
+      }
+    }
+
+    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;
+      }
+    }
+  }
+
+  // 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;
+    }
+
+    // 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;
+
+  // 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;
+  }
+
+  // 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;
+  }
+
+  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;
+  }
+
+  return FAILURE;
+}
+
+void RtApiDs :: closeStream()
+{
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiDs::closeStream(): no open stream to close!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  // 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();
+    }
+    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;
+  }
+
+  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;
+  }
+
+  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;
+  }
+
+  // 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 );
+  
+  DsHandle *handle = (DsHandle *) stream_.apiHandle;
+
+  timeBeginPeriod( 1 ); 
+
+  /*
+    memset( &statistics, 0, sizeof( statistics ) );
+    statistics.sampleRate = stream_.sampleRate;
+    statistics.writeDeviceBufferLeadBytes = handle->dsPointerLeadTime[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] );
+  }
+
+  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;
+    }
+  }
+
+  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;
+    }
+  }
+
+  handle->drainCounter = 0;
+  handle->internalDrain = false;
+  stream_.state = STREAM_RUNNING;
+
+ unlock:
+  MUTEX_UNLOCK( &stream_.mutex );
+
+  if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
+}
+
+void RtApiDs :: stopStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  if ( stream_.state == STREAM_STOPPED ) {
+    MUTEX_UNLOCK( &stream_.mutex );
+    return;
+  }
+
+  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::stopStream: error (" << getErrorString( result ) << ") stopping 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[0], &audioPtr, &dataLen, NULL, NULL, 0 );
+    if ( FAILED( result ) ) {
+      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
+      errorText_ = errorStream_.str();
+      goto unlock;
+    }
+
+    // Zero the DS buffer
+    ZeroMemory( audioPtr, dataLen );
+
+    // Unlock the DS buffer
+    result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+    if ( FAILED( result ) ) {
+      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
+      errorText_ = errorStream_.str();
+      goto unlock;
+    }
+
+    // If we start playing again, we must begin at beginning of buffer.
+    handle->bufferPointer[0] = 0;
+  }
+
+  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+    LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+    audioPtr = NULL;
+    dataLen = 0;
+
+    result = buffer->Stop();
+    if ( FAILED( result ) ) {
+      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input 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::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
+      errorText_ = errorStream_.str();
+      goto unlock;
+    }
+
+    // Zero the DS buffer
+    ZeroMemory( audioPtr, dataLen );
+
+    // Unlock the DS buffer
+    result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+    if ( FAILED( result ) ) {
+      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
+      errorText_ = errorStream_.str();
+      goto unlock;
+    }
+
+    // If we start recording again, we must begin at beginning of buffer.
+    handle->bufferPointer[1] = 0;
+  }
+
+ 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 );
+}
+
+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;
+
+  stopStream();
+}
+
+void RtApiDs :: callbackEvent()
+{
+  if ( stream_.state == STREAM_STOPPED ) {
+    Sleep(50); // sleep 50 milliseconds
+    return;
+  }
+
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  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;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  // The state might change while waiting on a mutex.
+  if ( stream_.state == STREAM_STOPPED ) {
+    MUTEX_UNLOCK( &stream_.mutex );
+    return;
+  }
+
+  // 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;
+#endif
+
+  LPVOID buffer1 = NULL;
+  LPVOID buffer2 = NULL;
+  DWORD bufferSize1 = 0;
+  DWORD bufferSize2 = 0;
+
+  char *buffer;
+  long bufferBytes;
+
+  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.
+
+    // 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];
+
+    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 );
+      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] );
+
+    buffersRolling = true;
+    handle->bufferPointer[0] = ( safeWritePos + handle->dsPointerLeadTime[0] );
+    handle->bufferPointer[1] = safeReadPos;
+  }
+
+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+    
+    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 );
+    }
+
+    // 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.
+
+    // 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 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 );
+    }
+
+    //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;
+    }
+  }
+
+  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+    // 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 );
+    }
+
+    LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+    long nextReadPos = handle->bufferPointer[1];
+    DWORD dsBufferSize = handle->dsBufferSize[1];
+
+    // 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 ( 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 { // 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 (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 );
+    }
+
+    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;
+
+    // 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 );
+
+    // 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;
+    }
+#endif
+
+ unlock:
+  MUTEX_UNLOCK( &stream_.mutex );
+
+  RtApi::tickStreamTime();
+}
+
+// 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;
+
+  while ( *isRunning == true ) {
+    object->callbackEvent();
+  }
+
+  _endthreadex( 0 );
+  return 0;
+}
+
+#include "tchar.h"
+
+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] );
+#else
+  s.append( std::string( name ) );
+#endif
+
+  return s;
+}
+
+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;
+
+    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++;
+    }
+    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++;
+    }
+    object->Release();
+  }
+
+  if ( info->getDefault && lpguid == NULL ) return FALSE;
+
+  if ( info->findIndex && info->counter > info->index ) {
+    info->id = lpguid;
+    info->name = convertTChar( description );
+    return FALSE;
+  }
+
+  return TRUE;
+}
+
+static char* getErrorString( int code )
+{
+  switch ( code ) {
+
+  case DSERR_ALLOCATED:
+    return "Already allocated";
+
+  case DSERR_CONTROLUNAVAIL:
+    return "Control unavailable";
+
+  case DSERR_INVALIDPARAM:
+    return "Invalid parameter";
+
+  case DSERR_INVALIDCALL:
+    return "Invalid call";
+
+  case DSERR_GENERIC:
+    return "Generic error";
+
+  case DSERR_PRIOLEVELNEEDED:
+    return "Priority level needed";
+
+  case DSERR_OUTOFMEMORY:
+    return "Out of memory";
+
+  case DSERR_BADFORMAT:
+    return "The sample rate or the channel format is not supported";
+
+  case DSERR_UNSUPPORTED:
+    return "Not supported";
+
+  case DSERR_NODRIVER:
+    return "No driver";
+
+  case DSERR_ALREADYINITIALIZED:
+    return "Already initialized";
+
+  case DSERR_NOAGGREGATION:
+    return "No aggregation";
+
+  case DSERR_BUFFERLOST:
+    return "Buffer lost";
+
+  case DSERR_OTHERAPPHASPRIO:
+    return "Another application already has priority";
+
+  case DSERR_UNINITIALIZED:
+    return "Uninitialized";
+
+  default:
+    return "DirectSound unknown error";
+  }
+}
+//******************** End of __WINDOWS_DS__ *********************//
+#endif
+
+
+#if defined(__LINUX_ALSA__)
+
+#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;
+
+  AlsaHandle()
+    :synchronized(false) { xrun[0] = false; xrun[1] = false; }
+};
+
+extern "C" void *alsaCallbackHandler( void * ptr );
+
+RtApiAlsa :: RtApiAlsa()
+{
+  // Nothing to do here.
+}
+
+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 ) << ".";
+        errorText_ = errorStream_.str();
+        error( RtError::WARNING );
+        break;
+      }
+      if ( subdevice < 0 )
+        break;
+      nDevices++;
+    }
+  nextcard:
+    snd_ctl_close( handle );
+    snd_card_next( &card );
+  }
+
+  return nDevices;
+}
+
+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;
+
+  // 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 ) << ".";
+        errorText_ = errorStream_.str();
+        error( RtError::WARNING );
+        break;
+      }
+      if ( subdevice < 0 ) break;
+      if ( nDevices == device ) {
+        sprintf( name, "hw:%d,%d", card, subdevice );
+        goto foundDevice;
+      }
+      nDevices++;
+    }
+  nextcard:
+    snd_ctl_close( chandle );
+    snd_card_next( &card );
+  }
+
+  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 );
+  }
+
+ 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;
+    }
+    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 );
+
+  // 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_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;
+  }
+
+  // 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 );
+
+ 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_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;
+  }
+
+  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 );
+
+  // 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;
+
+ 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 );
+
+  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;
+  }
+
+  // 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;
+  }
+
+  // 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;
+  }
+
+  // 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;
+  }
+
+  // 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();
+
+  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 )
+
+{
+#if defined(__RTAUDIO_DEBUG__)
+  snd_output_t *out;
+  snd_output_stdio_attach(&out, stderr, 0);
+#endif
+
+  // 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;
+
+  // 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 );
+    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;
+  }
+
+  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;
+  }
+
+ foundDevice:
+
+  // 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_stream_t stream;
+  if ( mode == OUTPUT )
+    stream = SND_PCM_STREAM_PLAYBACK;
+  else
+    stream = SND_PCM_STREAM_CAPTURE;
+
+  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 );
+#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 ) {
+      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 );
+    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;
+  }
+
+  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;
+
+  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;
+  }
+
+  // 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_S24;
+  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+    stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+    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_S8;
+  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+    stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+    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;
+
+ 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 ) << ".";
+    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;
+    }
+  }
+
+  // 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 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;
+  }
+
+  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 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 number, which in ALSA is referred to as the "period".
+  int totalSize, dir = 0;
+  unsigned int periods = 0;
+  if ( options ) periods = options->numberOfBuffers;
+  totalSize = *bufferSize * periods;
+
+  // 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 ( 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;
+  }
+
+  // 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;
+  }
+
+  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 );
+#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;
+  }
+
+#if defined(__RTAUDIO_DEBUG__)
+  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;
+
+  // 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;
+    }
+
+    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;
+  }
+  else {
+    apiInfo = (AlsaHandle *) stream_.apiHandle;
+  }
+  apiInfo->handles[mode] = phandle;
+
+  // 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;
+  }
+
+  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;
+      }
+    }
+
+    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;
+      }
+    }
+  }
+
+  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;
+
+    // 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;
+    }
+  }
+
+  return SUCCESS;
+
+ 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;
+  }
+
+  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;
+  }
+
+  return FAILURE;
+}
+
+void RtApiAlsa :: closeStream()
+{
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  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 );
+
+  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 ( 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;
+  }
+
+  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;
+  }
+
+  stream_.mode = UNINITIALIZED;
+  stream_.state = STREAM_CLOSED;
+}
+
+void RtApiAlsa :: startStream()
+{
+  // This method calls snd_pcm_prepare if the device isn't already in that state.
+
+  verifyStream();
+  if ( stream_.state == STREAM_RUNNING ) {
+    errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  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_.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_.state = STREAM_RUNNING;
+
+ unlock:
+  MUTEX_UNLOCK( &stream_.mutex );
+
+  pthread_cond_signal( &apiInfo->runnable );
+
+  if ( result >= 0 ) return;
+  error( RtError::SYSTEM_ERROR );
+}
+
+void RtApiAlsa :: stopStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  if ( stream_.state == STREAM_STOPPED ) {
+    MUTEX_UNLOCK( &stream_.mutex );
+    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 ) 
+      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 ( ( 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;
+    }
+  }
+
+ unlock:
+  stream_.state = STREAM_STOPPED;
+  MUTEX_UNLOCK( &stream_.mutex );
+
+  if ( result >= 0 ) return;
+  error( RtError::SYSTEM_ERROR );
+}
+
+void RtApiAlsa :: abortStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  if ( stream_.state == STREAM_STOPPED ) {
+    MUTEX_UNLOCK( &stream_.mutex );
+    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 ) {
+    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;
+    }
+  }
+
+  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:
+  stream_.state = STREAM_STOPPED;
+  MUTEX_UNLOCK( &stream_.mutex );
+
+  if ( result >= 0 ) return;
+  error( RtError::SYSTEM_ERROR );
+}
+
+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_.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 );
+
+  if ( doStopStream == 2 ) {
+    abortStream();
+    return;
+  }
+
+  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;
+
+  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;
+    }
+
+    // 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 ) << ".";
+            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: 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 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;
+  }
+
+ tryOutput:
+
+  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;
+    }
+
+    // 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 ) << ".";
+            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: 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;
+  }
+
+ unlock:
+  MUTEX_UNLOCK( &stream_.mutex );
+
+  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;
+
+  while ( *isRunning == true ) {
+    pthread_testcancel();
+    object->callbackEvent();
+  }
+
+  pthread_exit( NULL );
+}
+
+//******************** End of __LINUX_ALSA__ *********************//
+#endif
+
+
+#if defined(__LINUX_OSS__)
+
+#include <unistd.h>
+#include <sys/ioctl.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include "soundcard.h"
+#include <errno.h>
+#include <math.h>
+
+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;
+
+  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();
+}
+
+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;
+  }
+
+  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;
+  }
+
+  close( mixerfd );
+  return sysinfo.numaudios;
+}
+
+RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
+{
+  RtAudio::DeviceInfo info;
+  info.probed = false;
+
+  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;
+  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;
+  }
+
+  unsigned nDevices = sysinfo.numaudios;
+  if ( nDevices == 0 ) {
+    close( mixerfd );
+    errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
+    error( RtError::INVALID_USE );
+  }
+
+  if ( device >= nDevices ) {
+    close( mixerfd );
+    errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
+    error( RtError::INVALID_USE );
+  }
+
+  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.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;
+  }
+
+  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;
+  }
+
+  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;
+  }
+
+  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;
+  }
+
+  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;
+  }
+
+  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_RDWR;
+    }
+    else
+      flags |= O_RDONLY;
+  }
+
+  // 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 << ").";
+    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;
+  }
+
+  // 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;
+    }
+  }
+
+  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;
+  }
+
+  // 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;
+
+  // 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;
+  }
+
+  // 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;
+  }
+
+  // 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;
+    }
+
+    if ( pthread_cond_init( &handle->runnable, NULL ) ) {
+      errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
+      goto error;
+    }
+
+    stream_.apiHandle = (void *) handle;
+  }
+  else {
+    handle = (OssHandle *) stream_.apiHandle;
+  }
+  handle->id[mode] = fd;
+
+  // 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;
+      }
+    }
+
+    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;
+
+  // 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 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 );
+#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, 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;
+  }
+
+  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;
+  }
+
+  return FAILURE;
+}
+
+void RtApiOss :: closeStream()
+{
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiOss::closeStream(): no open stream to close!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  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 ( 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;
+  }
+
+  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;
+  }
+
+  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;
+  }
+
+  stream_.mode = UNINITIALIZED;
+  stream_.state = STREAM_CLOSED;
+}
+
+void RtApiOss :: startStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_RUNNING ) {
+    errorText_ = "RtApiOss::startStream(): the stream is already running!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  stream_.state = STREAM_RUNNING;
+
+  // No need to do anything else here ... OSS automatically starts
+  // when fed samples.
+
+  MUTEX_UNLOCK( &stream_.mutex );
+
+  OssHandle *handle = (OssHandle *) stream_.apiHandle;
+  pthread_cond_signal( &handle->runnable );
+}
+
+void RtApiOss :: stopStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  // The state might change while waiting on a mutex.
+  if ( stream_.state == STREAM_STOPPED ) {
+    MUTEX_UNLOCK( &stream_.mutex );
+    return;
+  }
+
+  int result = 0;
+  OssHandle *handle = (OssHandle *) stream_.apiHandle;
+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+    // Flush the output with zeros a few times.
+    char *buffer;
+    int samples;
+    RtAudioFormat 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 ) {
+      errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
+      errorText_ = errorStream_.str();
+      goto unlock;
+    }
+    handle->triggered = false;
+  }
+
+  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;
+    }
+  }
+
+ unlock:
+  stream_.state = STREAM_STOPPED;
+  MUTEX_UNLOCK( &stream_.mutex );
+
+  if ( result != -1 ) return;
+  error( RtError::SYSTEM_ERROR );
+}
+
+void RtApiOss :: abortStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  // The state might change while waiting on a mutex.
+  if ( stream_.state == STREAM_STOPPED ) {
+    MUTEX_UNLOCK( &stream_.mutex );
+    return;
+  }
+
+  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;
+  }
+
+  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;
+    }
+  }
+
+ unlock:
+  stream_.state = STREAM_STOPPED;
+  MUTEX_UNLOCK( &stream_.mutex );
+
+  if ( result != -1 ) return;
+  error( RtError::SYSTEM_ERROR );
+}
+
+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 );
+  }
+
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
+    error( RtError::WARNING );
+    return;
+  }
+
+  // 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_LOCK( &stream_.mutex );
+
+  // The state might change while waiting on a mutex.
+  if ( stream_.state == STREAM_STOPPED ) goto unlock;
+
+  int result;
+  char *buffer;
+  int samples;
+  RtAudioFormat format;
+
+  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] );
+      samples = stream_.bufferSize * stream_.nDeviceChannels[0];
+      format = stream_.deviceFormat[0];
+    }
+    else {
+      buffer = stream_.userBuffer[0];
+      samples = stream_.bufferSize * stream_.nUserChannels[0];
+      format = stream_.userFormat;
+    }
+
+    // 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) );
+
+    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.
+    }
+  }
+
+  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+    // 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;
+    }
+
+    // Read samples from device.
+    result = read( handle->id[1], buffer, samples * formatBytes(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 byte swapping if necessary.
+    if ( stream_.doByteSwap[1] )
+      byteSwapBuffer( buffer, samples, format );
+
+    // Do buffer conversion if necessary.
+    if ( stream_.doConvertBuffer[1] )
+      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+  }
+
+ 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;
+
+  while ( *isRunning == true ) {
+    pthread_testcancel();
+    object->callbackEvent();
+  }
+
+  pthread_exit( NULL );
+}
+
+//******************** End of __LINUX_OSS__ *********************//
+#endif
+
+
+// *************************************************** //
+//
+// 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 );
+  }
+}
+
+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();
+  }
+}
+
+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 );
+
+  return 0;
+}
+
+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];
+  }
+
+  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_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
+        stream_.convertInfo[mode].outJump = 1;
+      }
+    }
+  }
+  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.push_back( k * stream_.bufferSize );
+        stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
+        stream_.convertInfo[mode].inJump = 1;
+        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 {
+        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+          stream_.convertInfo[mode].inOffset[k] += firstChannel;
+      }
+    }
+    else {
+      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;
+        }
+        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;
+        }
+        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;
+        }
+        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;
+        }
+        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]];
+        }
+        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]];
+        }
+        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;
+        }
+        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;
+        }
+        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;
+        }
+        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;
+        }
+        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]];
+        }
+        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]];
+        }
+        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;
+        }
+        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;
+        }
+        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;
+        }
+        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]];
+        }
+        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);
+        }
+        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);
+        }
+        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;
+        }
+        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;
+        }
+        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]];
+        }
+        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;
+        }
+        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);
+        }
+        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);
+        }
+        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;
+        }
+        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]];
+        }
+        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);
+        }
+        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);
+        }
+        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);
+        }
+        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);
+        }
+        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]];
+        }
+        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);
+        }
+        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);
+        }
+        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);
+        }
+        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);
+        }
+        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);
+        }
+        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)); }
+
+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
+
+#endif
diff --git a/drivers/rtaudio/RtAudio.h b/drivers/rtaudio/RtAudio.h
new file mode 100644
index 0000000000..03924450b9
--- /dev/null
+++ b/drivers/rtaudio/RtAudio.h
@@ -0,0 +1,985 @@
+#ifdef RTAUDIO_ENABLED
+
+#if defined(OSX_ENABLED)
+
+#define __MACOSX_CORE__
+
+#elif defined(UNIX_ENABLED)
+
+#define __LINUX_ALSA__
+
+#elif defined(WINDOWS_ENABLED)
+
+#define __WINDOWS_DS__
+
+#endif
+
+
+/************************************************************************/
+/*! \class RtAudio
+    \brief Realtime audio i/o C++ classes.
+
+    RtAudio provides a common API (Application Programming Interface)
+    for realtime audio input/output across Linux (native ALSA, Jack,
+    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/
+
+    RtAudio: realtime audio i/o C++ classes
+    Copyright (c) 2001-2009 Gary P. Scavone
+
+    Permission is hereby granted, free of charge, to any person
+    obtaining a copy of this software and associated documentation files
+    (the "Software"), to deal in the Software without restriction,
+    including without limitation the rights to use, copy, modify, merge,
+    publish, distribute, sublicense, and/or sell copies of the Software,
+    and to permit persons to whom the Software is furnished to do so,
+    subject to the following conditions:
+
+    The above copyright notice and this permission notice shall be
+    included in all copies or substantial portions of the Software.
+
+    Any person wishing to distribute modifications to the Software is
+    asked to send the modifications to the original developer so that
+    they can be incorporated into the canonical version.  This is,
+    however, not a binding provision of this license.
+
+    THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+    EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+    MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+    IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+    ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+    CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+    WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+*/
+/************************************************************************/
+
+/*!
+  \file RtAudio.h
+ */
+
+// RtAudio: Version 4.0.6
+
+#ifndef __RTAUDIO_H
+#define __RTAUDIO_H
+
+#include <string>
+#include <vector>
+#include "RtError.h"
+
+/*! \typedef typedef unsigned long RtAudioFormat;
+    \brief RtAudio data format type.
+
+    Support for signed integers and floats.  Audio data fed to/from an
+    RtAudio stream is assumed to ALWAYS be in host byte order.  The
+    internal routines will automatically take care of any necessary
+    byte-swapping between the host format and the soundcard.  Thus,
+    endian-ness is not a concern in the following format definitions.
+
+    - \e RTAUDIO_SINT8:   8-bit signed integer.
+    - \e RTAUDIO_SINT16:  16-bit signed integer.
+    - \e RTAUDIO_SINT24:  Upper 3 bytes of 32-bit signed integer.
+    - \e RTAUDIO_SINT32:  32-bit signed integer.
+    - \e RTAUDIO_FLOAT32: Normalized between plus/minus 1.0.
+    - \e RTAUDIO_FLOAT64: Normalized between plus/minus 1.0.
+*/
+typedef unsigned long RtAudioFormat;
+static const RtAudioFormat RTAUDIO_SINT8 = 0x1;    // 8-bit signed integer.
+static const RtAudioFormat RTAUDIO_SINT16 = 0x2;   // 16-bit signed integer.
+static const RtAudioFormat RTAUDIO_SINT24 = 0x4;   // Lower 3 bytes of 32-bit signed integer.
+static const RtAudioFormat RTAUDIO_SINT32 = 0x8;   // 32-bit signed integer.
+static const RtAudioFormat RTAUDIO_FLOAT32 = 0x10; // Normalized between plus/minus 1.0.
+static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/minus 1.0.
+
+/*! \typedef typedef unsigned long RtAudioStreamFlags;
+    \brief RtAudio stream option flags.
+
+    The following flags can be OR'ed together to allow a client to
+    make changes to the default stream behavior:
+
+    - \e RTAUDIO_NONINTERLEAVED:   Use non-interleaved buffers (default = interleaved).
+    - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency.
+    - \e RTAUDIO_HOG_DEVICE:       Attempt grab device for exclusive use.
+
+    By default, RtAudio streams pass and receive audio data from the
+    client in an interleaved format.  By passing the
+    RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio
+    data will instead be presented in non-interleaved buffers.  In
+    this case, each buffer argument in the RtAudioCallback function
+    will point to a single array of data, with \c nFrames samples for
+    each channel concatenated back-to-back.  For example, the first
+    sample of data for the second channel would be located at index \c
+    nFrames (assuming the \c buffer pointer was recast to the correct
+    data type for the stream).
+
+    Certain audio APIs offer a number of parameters that influence the
+    I/O latency of a stream.  By default, RtAudio will attempt to set
+    these parameters internally for robust (glitch-free) performance
+    (though some APIs, like Windows Direct Sound, make this difficult).
+    By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream()
+    function, internal stream settings will be influenced in an attempt
+    to minimize stream latency, though possibly at the expense of stream
+    performance.
+
+    If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to
+    open the input and/or output stream device(s) for exclusive use.
+    Note that this is not possible with all supported audio APIs.
+
+    If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt 
+    to select realtime scheduling (round-robin) for the callback thread.
+*/
+typedef unsigned int RtAudioStreamFlags;
+static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1;    // Use non-interleaved buffers (default = interleaved).
+static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2;  // Attempt to set stream parameters for lowest possible latency.
+static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4;        // Attempt grab device and prevent use by others.
+static const RtAudioStreamFlags RTAUDIO_SCHEDULE_REALTIME = 0x8; // Try to select realtime scheduling for callback thread.
+
+/*! \typedef typedef unsigned long RtAudioStreamStatus;
+    \brief RtAudio stream status (over- or underflow) flags.
+
+    Notification of a stream over- or underflow is indicated by a
+    non-zero stream \c status argument in the RtAudioCallback function.
+    The stream status can be one of the following two options,
+    depending on whether the stream is open for output and/or input:
+
+    - \e RTAUDIO_INPUT_OVERFLOW:   Input data was discarded because of an overflow condition at the driver.
+    - \e RTAUDIO_OUTPUT_UNDERFLOW: The output buffer ran low, likely producing a break in the output sound.
+*/
+typedef unsigned int RtAudioStreamStatus;
+static const RtAudioStreamStatus RTAUDIO_INPUT_OVERFLOW = 0x1;    // Input data was discarded because of an overflow condition at the driver.
+static const RtAudioStreamStatus RTAUDIO_OUTPUT_UNDERFLOW = 0x2;  // The output buffer ran low, likely causing a gap in the output sound.
+
+//! RtAudio callback function prototype.
+/*!
+   All RtAudio clients must create a function of type RtAudioCallback
+   to read and/or write data from/to the audio stream.  When the
+   underlying audio system is ready for new input or output data, this
+   function will be invoked.
+
+   \param outputBuffer For output (or duplex) streams, the client
+          should write \c nFrames of audio sample frames into this
+          buffer.  This argument should be recast to the datatype
+          specified when the stream was opened.  For input-only
+          streams, this argument will be NULL.
+
+   \param inputBuffer For input (or duplex) streams, this buffer will
+          hold \c nFrames of input audio sample frames.  This
+          argument should be recast to the datatype specified when the
+          stream was opened.  For output-only streams, this argument
+          will be NULL.
+
+   \param nFrames The number of sample frames of input or output
+          data in the buffers.  The actual buffer size in bytes is
+          dependent on the data type and number of channels in use.
+
+   \param streamTime The number of seconds that have elapsed since the
+          stream was started.
+
+   \param status If non-zero, this argument indicates a data overflow
+          or underflow condition for the stream.  The particular
+          condition can be determined by comparison with the
+          RtAudioStreamStatus flags.
+
+   \param userData A pointer to optional data provided by the client
+          when opening the stream (default = NULL).
+
+   To continue normal stream operation, the RtAudioCallback function
+   should return a value of zero.  To stop the stream and drain the
+   output buffer, the function should return a value of one.  To abort
+   the stream immediately, the client should return a value of two.
+ */
+typedef int (*RtAudioCallback)( void *outputBuffer, void *inputBuffer,
+                                unsigned int nFrames,
+                                double streamTime,
+                                RtAudioStreamStatus status,
+                                void *userData );
+
+
+// **************************************************************** //
+//
+// RtAudio class declaration.
+//
+// RtAudio is a "controller" used to select an available audio i/o
+// interface.  It presents a common API for the user to call but all
+// functionality is implemented by the class RtApi and its
+// subclasses.  RtAudio creates an instance of an RtApi subclass
+// based on the user's API choice.  If no choice is made, RtAudio
+// attempts to make a "logical" API selection.
+//
+// **************************************************************** //
+
+class RtApi;
+
+class RtAudio
+{
+ public:
+
+  //! Audio API specifier arguments.
+  enum Api {
+    UNSPECIFIED,    /*!< Search for a working compiled API. */
+    LINUX_ALSA,     /*!< The Advanced Linux Sound Architecture API. */
+    LINUX_OSS,      /*!< The Linux Open Sound System API. */
+    UNIX_JACK,      /*!< The Jack Low-Latency Audio Server API. */
+    MACOSX_CORE,    /*!< Macintosh OS-X Core Audio API. */
+    WINDOWS_ASIO,   /*!< The Steinberg Audio Stream I/O API. */
+    WINDOWS_DS,     /*!< The Microsoft Direct Sound API. */
+    RTAUDIO_DUMMY   /*!< A compilable but non-functional API. */
+  };
+
+  //! The public device information structure for returning queried values.
+  struct DeviceInfo {
+    bool probed;                  /*!< true if the device capabilities were successfully probed. */
+    std::string name;             /*!< Character string device identifier. */
+    unsigned int outputChannels;  /*!< Maximum output channels supported by device. */
+    unsigned int inputChannels;   /*!< Maximum input channels supported by device. */
+    unsigned int duplexChannels;  /*!< Maximum simultaneous input/output channels supported by device. */
+    bool isDefaultOutput;         /*!< true if this is the default output device. */
+    bool isDefaultInput;          /*!< true if this is the default input device. */
+    std::vector<unsigned int> sampleRates; /*!< Supported sample rates (queried from list of standard rates). */
+    RtAudioFormat nativeFormats;  /*!< Bit mask of supported data formats. */
+
+    // Default constructor.
+    DeviceInfo()
+      :probed(false), outputChannels(0), inputChannels(0), duplexChannels(0),
+       isDefaultOutput(false), isDefaultInput(false), nativeFormats(0) {}
+  };
+
+  //! The structure for specifying input or ouput stream parameters.
+  struct StreamParameters {
+    unsigned int deviceId;     /*!< Device index (0 to getDeviceCount() - 1). */
+    unsigned int nChannels;    /*!< Number of channels. */
+    unsigned int firstChannel; /*!< First channel index on device (default = 0). */
+
+    // Default constructor.
+    StreamParameters()
+      : deviceId(0), nChannels(0), firstChannel(0) {}
+  };
+
+  //! The structure for specifying stream options.
+  /*!
+    The following flags can be OR'ed together to allow a client to
+    make changes to the default stream behavior:
+
+    - \e RTAUDIO_NONINTERLEAVED:    Use non-interleaved buffers (default = interleaved).
+    - \e RTAUDIO_MINIMIZE_LATENCY:  Attempt to set stream parameters for lowest possible latency.
+    - \e RTAUDIO_HOG_DEVICE:        Attempt grab device for exclusive use.
+    - \e RTAUDIO_SCHEDULE_REALTIME: Attempt to select realtime scheduling for callback thread.
+
+    By default, RtAudio streams pass and receive audio data from the
+    client in an interleaved format.  By passing the
+    RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio
+    data will instead be presented in non-interleaved buffers.  In
+    this case, each buffer argument in the RtAudioCallback function
+    will point to a single array of data, with \c nFrames samples for
+    each channel concatenated back-to-back.  For example, the first
+    sample of data for the second channel would be located at index \c
+    nFrames (assuming the \c buffer pointer was recast to the correct
+    data type for the stream).
+
+    Certain audio APIs offer a number of parameters that influence the
+    I/O latency of a stream.  By default, RtAudio will attempt to set
+    these parameters internally for robust (glitch-free) performance
+    (though some APIs, like Windows Direct Sound, make this difficult).
+    By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream()
+    function, internal stream settings will be influenced in an attempt
+    to minimize stream latency, though possibly at the expense of stream
+    performance.
+
+    If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to
+    open the input and/or output stream device(s) for exclusive use.
+    Note that this is not possible with all supported audio APIs.
+
+    If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt 
+    to select realtime scheduling (round-robin) for the callback thread.
+    The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME
+    flag is set. It defines the thread's realtime priority. 
+
+    The \c numberOfBuffers parameter can be used to control stream
+    latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs
+    only.  A value of two is usually the smallest allowed.  Larger
+    numbers can potentially result in more robust stream performance,
+    though likely at the cost of stream latency.  The value set by the
+    user is replaced during execution of the RtAudio::openStream()
+    function by the value actually used by the system.
+
+    The \c streamName parameter can be used to set the client name
+    when using the Jack API.  By default, the client name is set to
+    RtApiJack.  However, if you wish to create multiple instances of
+    RtAudio with Jack, each instance must have a unique client name.
+  */
+  struct StreamOptions {
+    RtAudioStreamFlags flags;      /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE). */
+    unsigned int numberOfBuffers;  /*!< Number of stream buffers. */
+    std::string streamName;        /*!< A stream name (currently used only in Jack). */
+    int priority;                  /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */
+
+    // Default constructor.
+    StreamOptions()
+    : flags(0), numberOfBuffers(0), priority(0) {}
+  };
+
+  //! A static function to determine the available compiled audio APIs.
+  /*!
+    The values returned in the std::vector can be compared against
+    the enumerated list values.  Note that there can be more than one
+    API compiled for certain operating systems.
+  */
+  static void getCompiledApi( std::vector<RtAudio::Api> &apis ) throw();
+
+  //! The class constructor.
+  /*!
+    The constructor performs minor initialization tasks.  No exceptions
+    can be thrown.
+
+    If no API argument is specified and multiple API support has been
+    compiled, the default order of use is JACK, ALSA, OSS (Linux
+    systems) and ASIO, DS (Windows systems).
+  */
+  RtAudio( RtAudio::Api api=UNSPECIFIED ) throw();
+
+  //! The destructor.
+  /*!
+    If a stream is running or open, it will be stopped and closed
+    automatically.
+  */
+  ~RtAudio() throw();
+
+  //! Returns the audio API specifier for the current instance of RtAudio.
+  RtAudio::Api getCurrentApi( void ) throw();
+
+  //! A public function that queries for the number of audio devices available.
+  /*!
+    This function performs a system query of available devices each time it
+    is called, thus supporting devices connected \e after instantiation. If
+    a system error occurs during processing, a warning will be issued. 
+  */
+  unsigned int getDeviceCount( void ) throw();
+
+  //! Return an RtAudio::DeviceInfo structure for a specified device number.
+  /*!
+
+    Any device integer between 0 and getDeviceCount() - 1 is valid.
+    If an invalid argument is provided, an RtError (type = INVALID_USE)
+    will be thrown.  If a device is busy or otherwise unavailable, the
+    structure member "probed" will have a value of "false" and all
+    other members are undefined.  If the specified device is the
+    current default input or output device, the corresponding
+    "isDefault" member will have a value of "true".
+  */
+  RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+
+  //! A function that returns the index of the default output device.
+  /*!
+    If the underlying audio API does not provide a "default
+    device", or if no devices are available, the return value will be
+    0.  Note that this is a valid device identifier and it is the
+    client's responsibility to verify that a device is available
+    before attempting to open a stream.
+  */
+  unsigned int getDefaultOutputDevice( void ) throw();
+
+  //! A function that returns the index of the default input device.
+  /*!
+    If the underlying audio API does not provide a "default
+    device", or if no devices are available, the return value will be
+    0.  Note that this is a valid device identifier and it is the
+    client's responsibility to verify that a device is available
+    before attempting to open a stream.
+  */
+  unsigned int getDefaultInputDevice( void ) throw();
+
+  //! A public function for opening a stream with the specified parameters.
+  /*!
+    An RtError (type = SYSTEM_ERROR) is thrown if a stream cannot be
+    opened with the specified parameters or an error occurs during
+    processing.  An RtError (type = INVALID_USE) is thrown if any
+    invalid device ID or channel number parameters are specified.
+
+    \param outputParameters Specifies output stream parameters to use
+           when opening a stream, including a device ID, number of channels,
+           and starting channel number.  For input-only streams, this
+           argument should be NULL.  The device ID is an index value between
+           0 and getDeviceCount() - 1.
+    \param inputParameters Specifies input stream parameters to use
+           when opening a stream, including a device ID, number of channels,
+           and starting channel number.  For output-only streams, this
+           argument should be NULL.  The device ID is an index value between
+           0 and getDeviceCount() - 1.
+    \param format An RtAudioFormat specifying the desired sample data format.
+    \param sampleRate The desired sample rate (sample frames per second).
+    \param *bufferFrames A pointer to a value indicating the desired
+           internal buffer size in sample frames.  The actual value
+           used by the device is returned via the same pointer.  A
+           value of zero can be specified, in which case the lowest
+           allowable value is determined.
+    \param callback A client-defined function that will be invoked
+           when input data is available and/or output data is needed.
+    \param userData An optional pointer to data that can be accessed
+           from within the callback function.
+    \param options An optional pointer to a structure containing various
+           global stream options, including a list of OR'ed RtAudioStreamFlags
+           and a suggested number of stream buffers that can be used to 
+           control stream latency.  More buffers typically result in more
+           robust performance, though at a cost of greater latency.  If a
+           value of zero is specified, a system-specific median value is
+           chosen.  If the RTAUDIO_MINIMIZE_LATENCY flag bit is set, the
+           lowest allowable value is used.  The actual value used is
+           returned via the structure argument.  The parameter is API dependent.
+  */
+  void openStream( RtAudio::StreamParameters *outputParameters,
+                   RtAudio::StreamParameters *inputParameters,
+                   RtAudioFormat format, unsigned int sampleRate,
+                   unsigned int *bufferFrames, RtAudioCallback callback,
+                   void *userData = NULL, RtAudio::StreamOptions *options = NULL );
+
+  //! A function that closes a stream and frees any associated stream memory.
+  /*!
+    If a stream is not open, this function issues a warning and
+    returns (no exception is thrown).
+  */
+  void closeStream( void ) throw();
+
+  //! A function that starts a stream.
+  /*!
+    An RtError (type = SYSTEM_ERROR) is thrown if an error occurs
+    during processing.  An RtError (type = INVALID_USE) is thrown if a
+    stream is not open.  A warning is issued if the stream is already
+    running.
+  */
+  void startStream( void );
+
+  //! Stop a stream, allowing any samples remaining in the output queue to be played.
+  /*!
+    An RtError (type = SYSTEM_ERROR) is thrown if an error occurs
+    during processing.  An RtError (type = INVALID_USE) is thrown if a
+    stream is not open.  A warning is issued if the stream is already
+    stopped.
+  */
+  void stopStream( void );
+
+  //! Stop a stream, discarding any samples remaining in the input/output queue.
+  /*!
+    An RtError (type = SYSTEM_ERROR) is thrown if an error occurs
+    during processing.  An RtError (type = INVALID_USE) is thrown if a
+    stream is not open.  A warning is issued if the stream is already
+    stopped.
+  */
+  void abortStream( void );
+
+  //! Returns true if a stream is open and false if not.
+  bool isStreamOpen( void ) const throw();
+
+  //! Returns true if the stream is running and false if it is stopped or not open.
+  bool isStreamRunning( void ) const throw();
+
+  //! Returns the number of elapsed seconds since the stream was started.
+  /*!
+    If a stream is not open, an RtError (type = INVALID_USE) will be thrown.
+  */
+  double getStreamTime( void );
+
+  //! Returns the internal stream latency in sample frames.
+  /*!
+    The stream latency refers to delay in audio input and/or output
+    caused by internal buffering by the audio system and/or hardware.
+    For duplex streams, the returned value will represent the sum of
+    the input and output latencies.  If a stream is not open, an
+    RtError (type = INVALID_USE) will be thrown.  If the API does not
+    report latency, the return value will be zero.
+  */
+  long getStreamLatency( void );
+
+ //! Returns actual sample rate in use by the stream.
+ /*!
+   On some systems, the sample rate used may be slightly different
+   than that specified in the stream parameters.  If a stream is not
+   open, an RtError (type = INVALID_USE) will be thrown.
+ */
+  unsigned int getStreamSampleRate( void );
+
+  //! Specify whether warning messages should be printed to stderr.
+  void showWarnings( bool value = true ) throw();
+
+ protected:
+
+  void openRtApi( RtAudio::Api api );
+  RtApi *rtapi_;
+};
+
+// Operating system dependent thread functionality.
+#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__)
+  #include <windows.h>
+  #include <process.h>
+
+  typedef unsigned long ThreadHandle;
+  typedef CRITICAL_SECTION StreamMutex;
+
+#elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
+  // Using pthread library for various flavors of unix.
+  #include <pthread.h>
+
+  typedef pthread_t ThreadHandle;
+  typedef pthread_mutex_t StreamMutex;
+
+#else // Setup for "dummy" behavior
+
+  #define __RTAUDIO_DUMMY__
+  typedef int ThreadHandle;
+  typedef int StreamMutex;
+
+#endif
+
+// This global structure type is used to pass callback information
+// between the private RtAudio stream structure and global callback
+// handling functions.
+struct CallbackInfo {
+  void *object;    // Used as a "this" pointer.
+  ThreadHandle thread;
+  void *callback;
+  void *userData;
+  void *apiInfo;   // void pointer for API specific callback information
+  bool isRunning;
+
+  // Default constructor.
+  CallbackInfo()
+    :object(0), callback(0), userData(0), apiInfo(0), isRunning(false) {}
+};
+
+// **************************************************************** //
+//
+// RtApi class declaration.
+//
+// Subclasses of RtApi contain all API- and OS-specific code necessary
+// to fully implement the RtAudio API.
+//
+// Note that RtApi is an abstract base class and cannot be
+// explicitly instantiated.  The class RtAudio will create an
+// instance of an RtApi subclass (RtApiOss, RtApiAlsa,
+// RtApiJack, RtApiCore, RtApiAl, RtApiDs, or RtApiAsio).
+//
+// **************************************************************** //
+
+#if defined( HAVE_GETTIMEOFDAY )
+  #include <sys/time.h>
+#endif
+
+#include <sstream>
+
+class RtApi
+{
+public:
+
+  RtApi();
+  virtual ~RtApi();
+  virtual RtAudio::Api getCurrentApi( void ) = 0;
+  virtual unsigned int getDeviceCount( void ) = 0;
+  virtual RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) = 0;
+  virtual unsigned int getDefaultInputDevice( void );
+  virtual unsigned int getDefaultOutputDevice( void );
+  void openStream( RtAudio::StreamParameters *outputParameters,
+                   RtAudio::StreamParameters *inputParameters,
+                   RtAudioFormat format, unsigned int sampleRate,
+                   unsigned int *bufferFrames, RtAudioCallback callback,
+                   void *userData, RtAudio::StreamOptions *options );
+  virtual void closeStream( void );
+  virtual void startStream( void ) = 0;
+  virtual void stopStream( void ) = 0;
+  virtual void abortStream( void ) = 0;
+  long getStreamLatency( void );
+  unsigned int getStreamSampleRate( void );
+  virtual double getStreamTime( void );
+  bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; };
+  bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; };
+  void showWarnings( bool value ) { showWarnings_ = value; };
+
+
+protected:
+
+  static const unsigned int MAX_SAMPLE_RATES;
+  static const unsigned int SAMPLE_RATES[];
+
+  enum { FAILURE, SUCCESS };
+
+  enum StreamState {
+    STREAM_STOPPED,
+    STREAM_RUNNING,
+    STREAM_CLOSED = -50
+  };
+
+  enum StreamMode {
+    OUTPUT,
+    INPUT,
+    DUPLEX,
+    UNINITIALIZED = -75
+  };
+
+  // A protected structure used for buffer conversion.
+  struct ConvertInfo {
+    int channels;
+    int inJump, outJump;
+    RtAudioFormat inFormat, outFormat;
+    std::vector<int> inOffset;
+    std::vector<int> outOffset;
+  };
+
+  // A protected structure for audio streams.
+  struct RtApiStream {
+    unsigned int device[2];    // Playback and record, respectively.
+    void *apiHandle;           // void pointer for API specific stream handle information
+    StreamMode mode;           // OUTPUT, INPUT, or DUPLEX.
+    StreamState state;         // STOPPED, RUNNING, or CLOSED
+    char *userBuffer[2];       // Playback and record, respectively.
+    char *deviceBuffer;
+    bool doConvertBuffer[2];   // Playback and record, respectively.
+    bool userInterleaved;
+    bool deviceInterleaved[2]; // Playback and record, respectively.
+    bool doByteSwap[2];        // Playback and record, respectively.
+    unsigned int sampleRate;
+    unsigned int bufferSize;
+    unsigned int nBuffers;
+    unsigned int nUserChannels[2];    // Playback and record, respectively.
+    unsigned int nDeviceChannels[2];  // Playback and record channels, respectively.
+    unsigned int channelOffset[2];    // Playback and record, respectively.
+    unsigned long latency[2];         // Playback and record, respectively.
+    RtAudioFormat userFormat;
+    RtAudioFormat deviceFormat[2];    // Playback and record, respectively.
+    StreamMutex mutex;
+    CallbackInfo callbackInfo;
+    ConvertInfo convertInfo[2];
+    double streamTime;         // Number of elapsed seconds since the stream started.
+
+#if defined(HAVE_GETTIMEOFDAY)
+    struct timeval lastTickTimestamp;
+#endif
+
+    RtApiStream()
+      :apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; }
+  };
+
+  typedef signed short Int16;
+  typedef signed int Int32;
+  typedef float Float32;
+  typedef double Float64;
+
+  std::ostringstream errorStream_;
+  std::string errorText_;
+  bool showWarnings_;
+  RtApiStream stream_;
+
+  /*!
+    Protected, api-specific method that attempts to open a device
+    with the given parameters.  This function MUST be implemented by
+    all subclasses.  If an error is encountered during the probe, a
+    "warning" message is reported and FAILURE is returned. A
+    successful probe is indicated by a return value of SUCCESS.
+  */
+  virtual bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, 
+                                unsigned int firstChannel, unsigned int sampleRate,
+                                RtAudioFormat format, unsigned int *bufferSize,
+                                RtAudio::StreamOptions *options );
+
+  //! A protected function used to increment the stream time.
+  void tickStreamTime( void );
+
+  //! Protected common method to clear an RtApiStream structure.
+  void clearStreamInfo();
+
+  /*!
+    Protected common method that throws an RtError (type =
+    INVALID_USE) if a stream is not open.
+  */
+  void verifyStream( void );
+
+  //! Protected common error method to allow global control over error handling.
+  void error( RtError::Type type );
+
+  /*!
+    Protected method used to perform format, channel number, and/or interleaving
+    conversions between the user and device buffers.
+  */
+  void convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info );
+
+  //! Protected common method used to perform byte-swapping on buffers.
+  void byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format );
+
+  //! Protected common method that returns the number of bytes for a given format.
+  unsigned int formatBytes( RtAudioFormat format );
+
+  //! Protected common method that sets up the parameters for buffer conversion.
+  void setConvertInfo( StreamMode mode, unsigned int firstChannel );
+};
+
+// **************************************************************** //
+//
+// Inline RtAudio definitions.
+//
+// **************************************************************** //
+
+inline RtAudio::Api RtAudio :: getCurrentApi( void ) throw() { return rtapi_->getCurrentApi(); }
+inline unsigned int RtAudio :: getDeviceCount( void ) throw() { return rtapi_->getDeviceCount(); }
+inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); }
+inline unsigned int RtAudio :: getDefaultInputDevice( void ) throw() { return rtapi_->getDefaultInputDevice(); }
+inline unsigned int RtAudio :: getDefaultOutputDevice( void ) throw() { return rtapi_->getDefaultOutputDevice(); }
+inline void RtAudio :: closeStream( void ) throw() { return rtapi_->closeStream(); }
+inline void RtAudio :: startStream( void ) { return rtapi_->startStream(); }
+inline void RtAudio :: stopStream( void )  { return rtapi_->stopStream(); }
+inline void RtAudio :: abortStream( void ) { return rtapi_->abortStream(); }
+inline bool RtAudio :: isStreamOpen( void ) const throw() { return rtapi_->isStreamOpen(); }
+inline bool RtAudio :: isStreamRunning( void ) const throw() { return rtapi_->isStreamRunning(); }
+inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); }
+inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); };
+inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); }
+inline void RtAudio :: showWarnings( bool value ) throw() { rtapi_->showWarnings( value ); }
+
+// RtApi Subclass prototypes.
+
+#if defined(__MACOSX_CORE__)
+
+#include <CoreAudio/AudioHardware.h>
+
+class RtApiCore: public RtApi
+{
+public:
+
+  RtApiCore();
+  ~RtApiCore();
+  RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; };
+  unsigned int getDeviceCount( void );
+  RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+  unsigned int getDefaultOutputDevice( void );
+  unsigned int getDefaultInputDevice( void );
+  void closeStream( void );
+  void startStream( void );
+  void stopStream( void );
+  void abortStream( void );
+  long getStreamLatency( void );
+
+  // This function is intended for internal use only.  It must be
+  // public because it is called by the internal callback handler,
+  // which is not a member of RtAudio.  External use of this function
+  // will most likely produce highly undesireable results!
+  bool callbackEvent( AudioDeviceID deviceId,
+                      const AudioBufferList *inBufferList,
+                      const AudioBufferList *outBufferList );
+
+  private:
+
+  bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, 
+                        unsigned int firstChannel, unsigned int sampleRate,
+                        RtAudioFormat format, unsigned int *bufferSize,
+                        RtAudio::StreamOptions *options );
+  static const char* getErrorCode( OSStatus code );
+};
+
+#endif
+
+#if defined(__UNIX_JACK__)
+
+class RtApiJack: public RtApi
+{
+public:
+
+  RtApiJack();
+  ~RtApiJack();
+  RtAudio::Api getCurrentApi( void ) { return RtAudio::UNIX_JACK; };
+  unsigned int getDeviceCount( void );
+  RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+  void closeStream( void );
+  void startStream( void );
+  void stopStream( void );
+  void abortStream( void );
+  long getStreamLatency( void );
+
+  // This function is intended for internal use only.  It must be
+  // public because it is called by the internal callback handler,
+  // which is not a member of RtAudio.  External use of this function
+  // will most likely produce highly undesireable results!
+  bool callbackEvent( unsigned long nframes );
+
+  private:
+
+  bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, 
+                        unsigned int firstChannel, unsigned int sampleRate,
+                        RtAudioFormat format, unsigned int *bufferSize,
+                        RtAudio::StreamOptions *options );
+};
+
+#endif
+
+#if defined(__WINDOWS_ASIO__)
+
+class RtApiAsio: public RtApi
+{
+public:
+
+  RtApiAsio();
+  ~RtApiAsio();
+  RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_ASIO; };
+  unsigned int getDeviceCount( void );
+  RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+  void closeStream( void );
+  void startStream( void );
+  void stopStream( void );
+  void abortStream( void );
+  long getStreamLatency( void );
+
+  // This function is intended for internal use only.  It must be
+  // public because it is called by the internal callback handler,
+  // which is not a member of RtAudio.  External use of this function
+  // will most likely produce highly undesireable results!
+  bool callbackEvent( long bufferIndex );
+
+  private:
+
+  std::vector<RtAudio::DeviceInfo> devices_;
+  void saveDeviceInfo( void );
+  bool coInitialized_;
+  bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, 
+                        unsigned int firstChannel, unsigned int sampleRate,
+                        RtAudioFormat format, unsigned int *bufferSize,
+                        RtAudio::StreamOptions *options );
+};
+
+#endif
+
+#if defined(__WINDOWS_DS__)
+
+class RtApiDs: public RtApi
+{
+public:
+
+  RtApiDs();
+  ~RtApiDs();
+  RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; };
+  unsigned int getDeviceCount( void );
+  unsigned int getDefaultOutputDevice( void );
+  unsigned int getDefaultInputDevice( void );
+  RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+  void closeStream( void );
+  void startStream( void );
+  void stopStream( void );
+  void abortStream( void );
+  long getStreamLatency( void );
+
+  // This function is intended for internal use only.  It must be
+  // public because it is called by the internal callback handler,
+  // which is not a member of RtAudio.  External use of this function
+  // will most likely produce highly undesireable results!
+  void callbackEvent( void );
+
+  private:
+
+  bool coInitialized_;
+  bool buffersRolling;
+  long duplexPrerollBytes;
+  bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, 
+                        unsigned int firstChannel, unsigned int sampleRate,
+                        RtAudioFormat format, unsigned int *bufferSize,
+                        RtAudio::StreamOptions *options );
+};
+
+#endif
+
+#if defined(__LINUX_ALSA__)
+
+class RtApiAlsa: public RtApi
+{
+public:
+
+  RtApiAlsa();
+  ~RtApiAlsa();
+  RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; };
+  unsigned int getDeviceCount( void );
+  RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+  void closeStream( void );
+  void startStream( void );
+  void stopStream( void );
+  void abortStream( void );
+
+  // This function is intended for internal use only.  It must be
+  // public because it is called by the internal callback handler,
+  // which is not a member of RtAudio.  External use of this function
+  // will most likely produce highly undesireable results!
+  void callbackEvent( void );
+
+  private:
+
+  std::vector<RtAudio::DeviceInfo> devices_;
+  void saveDeviceInfo( void );
+  bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, 
+                        unsigned int firstChannel, unsigned int sampleRate,
+                        RtAudioFormat format, unsigned int *bufferSize,
+                        RtAudio::StreamOptions *options );
+};
+
+#endif
+
+#if defined(__LINUX_OSS__)
+
+class RtApiOss: public RtApi
+{
+public:
+
+  RtApiOss();
+  ~RtApiOss();
+  RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; };
+  unsigned int getDeviceCount( void );
+  RtAudio::DeviceInfo getDeviceInfo( unsigned int device );
+  void closeStream( void );
+  void startStream( void );
+  void stopStream( void );
+  void abortStream( void );
+
+  // This function is intended for internal use only.  It must be
+  // public because it is called by the internal callback handler,
+  // which is not a member of RtAudio.  External use of this function
+  // will most likely produce highly undesireable results!
+  void callbackEvent( void );
+
+  private:
+
+  bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, 
+                        unsigned int firstChannel, unsigned int sampleRate,
+                        RtAudioFormat format, unsigned int *bufferSize,
+                        RtAudio::StreamOptions *options );
+};
+
+#endif
+
+#if defined(__RTAUDIO_DUMMY__)
+
+class RtApiDummy: public RtApi
+{
+public:
+
+  RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtError::WARNING ); };
+  RtAudio::Api getCurrentApi( void ) { return RtAudio::RTAUDIO_DUMMY; };
+  unsigned int getDeviceCount( void ) { return 0; };
+  RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) { RtAudio::DeviceInfo info; return info; };
+  void closeStream( void ) {};
+  void startStream( void ) {};
+  void stopStream( void ) {};
+  void abortStream( void ) {};
+
+  private:
+
+  bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, 
+                        unsigned int firstChannel, unsigned int sampleRate,
+                        RtAudioFormat format, unsigned int *bufferSize,
+                        RtAudio::StreamOptions *options ) { return false; };
+};
+
+#endif
+
+#endif
+
+// Indentation settings for Vim and Emacs
+//
+// Local Variables:
+// c-basic-offset: 2
+// indent-tabs-mode: nil
+// End:
+//
+// vim: et sts=2 sw=2
+#endif
diff --git a/drivers/rtaudio/RtError.h b/drivers/rtaudio/RtError.h
new file mode 100644
index 0000000000..ac5c41498a
--- /dev/null
+++ b/drivers/rtaudio/RtError.h
@@ -0,0 +1,60 @@
+/************************************************************************/
+/*! \class RtError
+    \brief Exception handling class for RtAudio & RtMidi.
+
+    The RtError class is quite simple but it does allow errors to be
+    "caught" by RtError::Type. See the RtAudio and RtMidi
+    documentation to know which methods can throw an RtError.
+
+*/
+/************************************************************************/
+
+#ifndef RTERROR_H
+#define RTERROR_H
+
+#include <exception>
+#include <iostream>
+#include <string>
+
+class RtError : public std::exception
+{
+ public:
+  //! Defined RtError types.
+  enum Type {
+    WARNING,           /*!< A non-critical error. */
+    DEBUG_WARNING,     /*!< A non-critical error which might be useful for debugging. */
+    UNSPECIFIED,       /*!< The default, unspecified error type. */
+    NO_DEVICES_FOUND,  /*!< No devices found on system. */
+    INVALID_DEVICE,    /*!< An invalid device ID was specified. */
+    MEMORY_ERROR,      /*!< An error occured during memory allocation. */
+    INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */
+    INVALID_USE,       /*!< The function was called incorrectly. */
+    DRIVER_ERROR,      /*!< A system driver error occured. */
+    SYSTEM_ERROR,      /*!< A system error occured. */
+    THREAD_ERROR       /*!< A thread error occured. */
+  };
+
+  //! The constructor.
+  RtError( const std::string& message, Type type = RtError::UNSPECIFIED ) throw() : message_(message), type_(type) {}
+ 
+  //! The destructor.
+  virtual ~RtError( void ) throw() {}
+
+  //! Prints thrown error message to stderr.
+  virtual void printMessage( void ) throw() { std::cerr << '\n' << message_ << "\n\n"; }
+
+  //! Returns the thrown error message type.
+  virtual const Type& getType(void) throw() { return type_; }
+
+  //! Returns the thrown error message string.
+  virtual const std::string& getMessage(void) throw() { return message_; }
+
+  //! Returns the thrown error message as a c-style string.
+  virtual const char* what( void ) const throw() { return message_.c_str(); }
+
+ protected:
+  std::string message_;
+  Type type_;
+};
+
+#endif
diff --git a/drivers/rtaudio/SCsub b/drivers/rtaudio/SCsub
new file mode 100644
index 0000000000..6699efef75
--- /dev/null
+++ b/drivers/rtaudio/SCsub
@@ -0,0 +1,4 @@
+Import('env')
+Export('env');
+
+env.add_source_files(env.drivers_sources,"*.cpp")
diff --git a/drivers/rtaudio/audio_driver_rtaudio.cpp b/drivers/rtaudio/audio_driver_rtaudio.cpp
new file mode 100644
index 0000000000..ac8f502178
--- /dev/null
+++ b/drivers/rtaudio/audio_driver_rtaudio.cpp
@@ -0,0 +1,188 @@
+/*************************************************/
+/*  audio_driver_rtaudio.cpp                     */
+/*************************************************/
+/*            This file is part of:              */
+/*                GODOT ENGINE                   */
+/*************************************************/
+/*       Source code within this file is:        */
+/*  (c) 2007-2010 Juan Linietsky, Ariel Manzur   */
+/*             All Rights Reserved.              */
+/*************************************************/
+
+#include "audio_driver_rtaudio.h"
+#include "globals.h"
+#include "os/os.h"
+#ifdef RTAUDIO_ENABLED
+
+const char* AudioDriverRtAudio::get_name() const {
+
+#ifdef OSX_ENABLED
+	return "RtAudio-OSX";
+#elif defined(UNIX_ENABLED)
+	return "RtAudio-ALSA";
+#elif defined(WINDOWS_ENABLED)
+	return "RtAudio-DirectSound";
+#else
+	return "RtAudio-None";
+#endif
+
+}
+
+// Two-channel sawtooth wave generator.
+int AudioDriverRtAudio::callback( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
+	double streamTime, RtAudioStreamStatus status, void *userData ) {
+
+	if (status)
+		print_line("lost?");
+	int32_t *buffer = (int32_t *) outputBuffer;
+
+	AudioDriverRtAudio *self = (AudioDriverRtAudio*)userData;
+
+	if (self->mutex->try_lock()!=OK) {
+
+
+		// what should i do..
+		for(unsigned int i=0;i<nBufferFrames;i++)
+			buffer[i]=0;
+
+		return 0;
+	}
+
+	self->audio_server_process(nBufferFrames,buffer);
+
+	self->mutex->unlock();;
+
+	return 0;
+}
+
+Error AudioDriverRtAudio::init() {
+
+	active=false;
+	mutex=NULL;
+	dac = memnew( RtAudio );
+
+	ERR_EXPLAIN("Cannot initialize RtAudio audio driver: No devices present.")
+	ERR_FAIL_COND_V( dac->getDeviceCount() < 1, ERR_UNAVAILABLE );
+
+	String channels = GLOBAL_DEF("audio/output","stereo");
+
+	if (channels=="5.1")
+		output_format=OUTPUT_5_1;
+	else if (channels=="quad")
+		output_format=OUTPUT_QUAD;
+	else if (channels=="mono")
+		output_format=OUTPUT_MONO;
+	else
+		output_format=OUTPUT_STEREO;
+
+
+	RtAudio::StreamParameters parameters;
+	parameters.deviceId = dac->getDefaultOutputDevice();
+	RtAudio::StreamOptions options;
+//	options.
+//	RtAudioStreamFlags flags;      /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE). *///
+//	unsigned int numberOfBuffers;  /*!< Number of stream buffers. */
+//	std::string streamName;        /*!< A stream name (currently used only in Jack). */
+//	int priority;                  /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */
+
+
+	parameters.firstChannel = 0;
+	mix_rate = GLOBAL_DEF("audio/mix_rate",44100);
+
+	int latency = GLOBAL_DEF("audio/output_latency",25);
+	unsigned int buffer_size = nearest_power_of_2( latency * mix_rate / 1000 );
+	if (OS::get_singleton()->is_stdout_verbose()) {
+		print_line("audio buffer size: "+itos(buffer_size));
+	}
+
+//	bool success=false;
+
+	while( true) {
+
+		switch(output_format) {
+
+			case OUTPUT_MONO: parameters.nChannels = 1; break;
+			case OUTPUT_STEREO: parameters.nChannels = 2; break;
+			case OUTPUT_QUAD: parameters.nChannels = 4; break;
+			case OUTPUT_5_1: parameters.nChannels = 6; break;
+		};
+
+
+		try {
+			dac->openStream( &parameters, NULL, RTAUDIO_SINT32,
+			    mix_rate, &buffer_size, &callback, this,&options );
+			mutex = Mutex::create(true);
+			active=true;
+
+			break;
+		} catch ( RtError& e ) {
+			// try with less channels
+
+			ERR_PRINT("Unable to open audio, retrying with fewer channels..");
+
+			switch(output_format) {
+
+				case OUTPUT_MONO: ERR_EXPLAIN("Unable to open audio."); ERR_FAIL_V( ERR_UNAVAILABLE ); break;
+				case OUTPUT_STEREO: output_format=OUTPUT_MONO; break;
+				case OUTPUT_QUAD: output_format=OUTPUT_STEREO; break;
+				case OUTPUT_5_1: output_format=OUTPUT_QUAD; break;
+			};
+		}
+	}
+
+
+	return OK;
+}
+
+
+int AudioDriverRtAudio::get_mix_rate() const {
+
+	return mix_rate;
+}
+
+AudioDriverSW::OutputFormat AudioDriverRtAudio::get_output_format() const {
+
+	return output_format;
+}
+
+void AudioDriverRtAudio::start() {
+
+	if (active)
+		dac->startStream();
+}
+
+void AudioDriverRtAudio::lock() {
+
+	if (mutex)
+		mutex->lock();
+}
+
+void AudioDriverRtAudio::unlock() {
+
+	if (mutex)
+		mutex->unlock();
+}
+
+void AudioDriverRtAudio::finish() {
+
+
+	 if ( active && dac->isStreamOpen() )
+		 dac->closeStream();
+	 if (mutex)
+		 memdelete(mutex);
+	 if (dac)
+		 memdelete(dac);
+}
+
+
+
+AudioDriverRtAudio::AudioDriverRtAudio()
+{
+	mutex=NULL;
+	mix_rate=44100;
+	output_format=OUTPUT_STEREO;
+}
+
+
+
+#endif
diff --git a/drivers/rtaudio/audio_driver_rtaudio.h b/drivers/rtaudio/audio_driver_rtaudio.h
new file mode 100644
index 0000000000..a16470d701
--- /dev/null
+++ b/drivers/rtaudio/audio_driver_rtaudio.h
@@ -0,0 +1,48 @@
+/*************************************************/
+/*  audio_driver_rtaudio.h                       */
+/*************************************************/
+/*            This file is part of:              */
+/*                GODOT ENGINE                   */
+/*************************************************/
+/*       Source code within this file is:        */
+/*  (c) 2007-2010 Juan Linietsky, Ariel Manzur   */
+/*             All Rights Reserved.              */
+/*************************************************/
+
+#ifndef AUDIO_DRIVER_RTAUDIO_H
+#define AUDIO_DRIVER_RTAUDIO_H
+
+#ifdef RTAUDIO_ENABLED
+
+#include "servers/audio/audio_server_sw.h"
+#include "drivers/rtaudio/RtAudio.h"
+
+class AudioDriverRtAudio : public AudioDriverSW {
+
+
+	static int callback( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
+	 double streamTime, RtAudioStreamStatus status, void *userData );
+	OutputFormat output_format;
+	Mutex *mutex;
+	RtAudio *dac;
+	int mix_rate;
+	bool active;
+public:
+
+
+	virtual const char* get_name() const;
+
+	virtual Error init();
+	virtual void start();
+	virtual int get_mix_rate() const ;
+	virtual OutputFormat get_output_format() const;
+	virtual void lock();
+	virtual void unlock();
+	virtual void finish();
+
+	AudioDriverRtAudio();
+
+};
+
+#endif // AUDIO_DRIVER_RTAUDIO_H
+#endif