--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/DPF-Prymula-audioplugins/dpf/distrho/src/jackbridge/rtaudio/RtAudio.cpp	Mon Oct 16 21:53:34 2023 +0200
@@ -0,0 +1,10908 @@
+/************************************************************************/
+/*! \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, ASIO and WASAPI) operating systems.
+
+    RtAudio GitHub site: https://github.com/thestk/rtaudio
+    RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
+
+    RtAudio: realtime audio i/o C++ classes
+    Copyright (c) 2001-2019 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 5.1.0
+
+#include "RtAudio.h"
+#include <iostream>
+#include <cstdlib>
+#include <cstring>
+#include <climits>
+#include <cmath>
+#include <algorithm>
+
+// 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__) || defined(__WINDOWS_WASAPI__)
+  #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
+  #define MUTEX_DESTROY(A)    DeleteCriticalSection(A)
+  #define MUTEX_LOCK(A)       EnterCriticalSection(A)
+  #define MUTEX_UNLOCK(A)     LeaveCriticalSection(A)
+
+  #include "tchar.h"
+
+  template<typename T> inline
+  std::string convertCharPointerToStdString(const T *text);
+
+  template<> inline
+  std::string convertCharPointerToStdString(const char *text)
+  {
+    return std::string(text);
+  }
+
+  template<> inline
+  std::string convertCharPointerToStdString(const wchar_t *text)
+  {
+    int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL);
+    std::string s( length-1, '\0' );
+    WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL);
+    return s;
+  }
+
+#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || 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.
+//
+// *************************************************** //
+
+std::string RtAudio :: getVersion( void )
+{
+  return RTAUDIO_VERSION;
+}
+
+// Define API names and display names.
+// Must be in same order as API enum.
+extern "C" {
+const char* rtaudio_api_names[][2] = {
+  { "unspecified" , "Unknown" },
+  { "alsa"        , "ALSA" },
+  { "pulse"       , "Pulse" },
+  { "oss"         , "OpenSoundSystem" },
+  { "jack"        , "Jack" },
+  { "core"        , "CoreAudio" },
+  { "wasapi"      , "WASAPI" },
+  { "asio"        , "ASIO" },
+  { "ds"          , "DirectSound" },
+  { "dummy"       , "Dummy" },
+};
+const unsigned int rtaudio_num_api_names = 
+  sizeof(rtaudio_api_names)/sizeof(rtaudio_api_names[0]);
+
+// The order here will control the order of RtAudio's API search in
+// the constructor.
+extern "C" const RtAudio::Api rtaudio_compiled_apis[] = {
+#if defined(__UNIX_JACK__)
+  RtAudio::UNIX_JACK,
+#endif
+#if defined(__LINUX_PULSE__)
+  RtAudio::LINUX_PULSE,
+#endif
+#if defined(__LINUX_ALSA__)
+  RtAudio::LINUX_ALSA,
+#endif
+#if defined(__LINUX_OSS__)
+  RtAudio::LINUX_OSS,
+#endif
+#if defined(__WINDOWS_ASIO__)
+  RtAudio::WINDOWS_ASIO,
+#endif
+#if defined(__WINDOWS_WASAPI__)
+  RtAudio::WINDOWS_WASAPI,
+#endif
+#if defined(__WINDOWS_DS__)
+  RtAudio::WINDOWS_DS,
+#endif
+#if defined(__MACOSX_CORE__)
+  RtAudio::MACOSX_CORE,
+#endif
+#if defined(__RTAUDIO_DUMMY__)
+  RtAudio::RTAUDIO_DUMMY,
+#endif
+  RtAudio::UNSPECIFIED,
+};
+extern "C" const unsigned int rtaudio_num_compiled_apis =
+  sizeof(rtaudio_compiled_apis)/sizeof(rtaudio_compiled_apis[0])-1;
+}
+
+// This is a compile-time check that rtaudio_num_api_names == RtAudio::NUM_APIS.
+// If the build breaks here, check that they match.
+template<bool b> class StaticAssert { private: StaticAssert() {} };
+template<> class StaticAssert<true>{ public: StaticAssert() {} };
+class StaticAssertions { StaticAssertions() {
+  StaticAssert<rtaudio_num_api_names == RtAudio::NUM_APIS>();
+}};
+
+void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis )
+{
+  apis = std::vector<RtAudio::Api>(rtaudio_compiled_apis,
+                                   rtaudio_compiled_apis + rtaudio_num_compiled_apis);
+}
+
+std::string RtAudio :: getApiName( RtAudio::Api api )
+{
+  if (api < 0 || api >= RtAudio::NUM_APIS)
+    return "";
+  return rtaudio_api_names[api][0];
+}
+
+std::string RtAudio :: getApiDisplayName( RtAudio::Api api )
+{
+  if (api < 0 || api >= RtAudio::NUM_APIS)
+    return "Unknown";
+  return rtaudio_api_names[api][1];
+}
+
+RtAudio::Api RtAudio :: getCompiledApiByName( const std::string &name )
+{
+  unsigned int i=0;
+  for (i = 0; i < rtaudio_num_compiled_apis; ++i)
+    if (name == rtaudio_api_names[rtaudio_compiled_apis[i]][0])
+      return rtaudio_compiled_apis[i];
+  return RtAudio::UNSPECIFIED;
+}
+
+void RtAudio :: openRtApi( RtAudio::Api api )
+{
+  if ( rtapi_ )
+    delete rtapi_;
+  rtapi_ = 0;
+
+#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_PULSE__)
+  if ( api == LINUX_PULSE )
+    rtapi_ = new RtApiPulse();
+#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_WASAPI__)
+  if ( api == WINDOWS_WASAPI )
+    rtapi_ = new RtApiWasapi();
+#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 )
+{
+  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_ && 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 thow an error.
+  std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";
+  throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );
+}
+
+RtAudio :: ~RtAudio()
+{
+  if ( rtapi_ )
+    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,
+                            RtAudioErrorCallback errorCallback )
+{
+  return rtapi_->openStream( outputParameters, inputParameters, format,
+                             sampleRate, bufferFrames, callback,
+                             userData, options, errorCallback );
+}
+
+// *************************************************** //
+//
+// 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;
+  firstErrorOccurred_ = false;
+}
+
+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,
+                          RtAudioErrorCallback errorCallback )
+{
+  if ( stream_.state != STREAM_CLOSED ) {
+    errorText_ = "RtApi::openStream: a stream is already open!";
+    error( RtAudioError::INVALID_USE );
+    return;
+  }
+
+  // Clear stream information potentially left from a previously open stream.
+  clearStreamInfo();
+
+  if ( oParams && oParams->nChannels < 1 ) {
+    errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
+    error( RtAudioError::INVALID_USE );
+    return;
+  }
+
+  if ( iParams && iParams->nChannels < 1 ) {
+    errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
+    error( RtAudioError::INVALID_USE );
+    return;
+  }
+
+  if ( oParams == NULL && iParams == NULL ) {
+    errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
+    error( RtAudioError::INVALID_USE );
+    return;
+  }
+
+  if ( formatBytes(format) == 0 ) {
+    errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
+    error( RtAudioError::INVALID_USE );
+    return;
+  }
+
+  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( RtAudioError::INVALID_USE );
+      return;
+    }
+  }
+
+  unsigned int iChannels = 0;
+  if ( iParams ) {
+    iChannels = iParams->nChannels;
+    if ( iParams->deviceId >= nDevices ) {
+      errorText_ = "RtApi::openStream: input device parameter value is invalid.";
+      error( RtAudioError::INVALID_USE );
+      return;
+    }
+  }
+
+  bool result;
+
+  if ( oChannels > 0 ) {
+
+    result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
+                              sampleRate, format, bufferFrames, options );
+    if ( result == false ) {
+      error( RtAudioError::SYSTEM_ERROR );
+      return;
+    }
+  }
+
+  if ( iChannels > 0 ) {
+
+    result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
+                              sampleRate, format, bufferFrames, options );
+    if ( result == false ) {
+      if ( oChannels > 0 ) closeStream();
+      error( RtAudioError::SYSTEM_ERROR );
+      return;
+    }
+  }
+
+  stream_.callbackInfo.callback = (void *) callback;
+  stream_.callbackInfo.userData = userData;
+  stream_.callbackInfo.errorCallback = (void *) errorCallback;
+
+  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
+}
+
+void RtApi :: setStreamTime( double time )
+{
+  verifyStream();
+
+  if ( time >= 0.0 )
+    stream_.streamTime = time;
+#if defined( HAVE_GETTIMEOFDAY )
+  gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
+}
+
+unsigned int RtApi :: getStreamSampleRate( void )
+{
+ verifyStream();
+
+ return stream_.sampleRate;
+}
+
+
+// *************************************************** //
+//
+// OS/API-specific methods.
+//
+// *************************************************** //
+
+#if defined(__MACOSX_CORE__)
+
+#include <unistd.h>
+
+// 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.
+
+// 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()
+{
+#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )
+  // This is a largely undocumented but absolutely necessary
+  // requirement starting with OS-X 10.6.  If not called, queries and
+  // updates to various audio device properties are not handled
+  // correctly.
+  CFRunLoopRef theRunLoop = NULL;
+  AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,
+                                          kAudioObjectPropertyScopeGlobal,
+                                          kAudioObjectPropertyElementMaster };
+  OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
+  if ( result != noErr ) {
+    errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";
+    error( RtAudioError::WARNING );
+  }
+#endif
+}
+
+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;
+  AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+  OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );
+  if ( result != noErr ) {
+    errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
+    error( RtAudioError::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 );
+  AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+  OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
+  if ( result != noErr ) {
+    errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
+    error( RtAudioError::WARNING );
+    return 0;
+  }
+
+  dataSize *= nDevices;
+  AudioDeviceID deviceList[ nDevices ];
+  property.mSelector = kAudioHardwarePropertyDevices;
+  result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
+  if ( result != noErr ) {
+    errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
+    error( RtAudioError::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( RtAudioError::WARNING );
+  return 0;
+}
+
+unsigned int RtApiCore :: getDefaultOutputDevice( void )
+{
+  unsigned int nDevices = getDeviceCount();
+  if ( nDevices <= 1 ) return 0;
+
+  AudioDeviceID id;
+  UInt32 dataSize = sizeof( AudioDeviceID );
+  AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+  OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
+  if ( result != noErr ) {
+    errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
+    error( RtAudioError::WARNING );
+    return 0;
+  }
+
+  dataSize = sizeof( AudioDeviceID ) * nDevices;
+  AudioDeviceID deviceList[ nDevices ];
+  property.mSelector = kAudioHardwarePropertyDevices;
+  result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
+  if ( result != noErr ) {
+    errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
+    error( RtAudioError::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( RtAudioError::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( RtAudioError::INVALID_USE );
+    return info;
+  }
+
+  if ( device >= nDevices ) {
+    errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
+    error( RtAudioError::INVALID_USE );
+    return info;
+  }
+
+  AudioDeviceID deviceList[ nDevices ];
+  UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
+  AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+                                          kAudioObjectPropertyScopeGlobal,
+                                          kAudioObjectPropertyElementMaster };
+  OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
+                                                0, NULL, &dataSize, (void *) &deviceList );
+  if ( result != noErr ) {
+    errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
+    error( RtAudioError::WARNING );
+    return info;
+  }
+
+  AudioDeviceID id = deviceList[ device ];
+
+  // Get the device name.
+  info.name.erase();
+  CFStringRef cfname;
+  dataSize = sizeof( CFStringRef );
+  property.mSelector = kAudioObjectPropertyManufacturer;
+  result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
+  if ( result != noErr ) {
+    errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::WARNING );
+    return info;
+  }
+
+  //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
+  int length = CFStringGetLength(cfname);
+  char *mname = (char *)malloc(length * 3 + 1);
+#if defined( UNICODE ) || defined( _UNICODE )
+  CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8);
+#else
+  CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());
+#endif
+  info.name.append( (const char *)mname, strlen(mname) );
+  info.name.append( ": " );
+  CFRelease( cfname );
+  free(mname);
+
+  property.mSelector = kAudioObjectPropertyName;
+  result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
+  if ( result != noErr ) {
+    errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::WARNING );
+    return info;
+  }
+
+  //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
+  length = CFStringGetLength(cfname);
+  char *name = (char *)malloc(length * 3 + 1);
+#if defined( UNICODE ) || defined( _UNICODE )
+  CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8);
+#else
+  CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());
+#endif
+  info.name.append( (const char *)name, strlen(name) );
+  CFRelease( cfname );
+  free(name);
+
+  // Get the output stream "configuration".
+  AudioBufferList	*bufferList = nil;
+  property.mSelector = kAudioDevicePropertyStreamConfiguration;
+  property.mScope = kAudioDevicePropertyScopeOutput;
+  //  property.mElement = kAudioObjectPropertyElementWildcard;
+  dataSize = 0;
+  result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+  if ( result != noErr || dataSize == 0 ) {
+    errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::WARNING );
+    return info;
+  }
+
+  // Allocate the AudioBufferList.
+  bufferList = (AudioBufferList *) malloc( dataSize );
+  if ( bufferList == NULL ) {
+    errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
+    error( RtAudioError::WARNING );
+    return info;
+  }
+
+  result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+  if ( result != noErr || dataSize == 0 ) {
+    free( bufferList );
+    errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::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".
+  property.mScope = kAudioDevicePropertyScopeInput;
+  result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+  if ( result != noErr || dataSize == 0 ) {
+    errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::WARNING );
+    return info;
+  }
+
+  // Allocate the AudioBufferList.
+  bufferList = (AudioBufferList *) malloc( dataSize );
+  if ( bufferList == NULL ) {
+    errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
+    error( RtAudioError::WARNING );
+    return info;
+  }
+
+  result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+  if (result != noErr || dataSize == 0) {
+    free( bufferList );
+    errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::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.
+  property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
+  if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;
+  result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+  if ( result != kAudioHardwareNoError || dataSize == 0 ) {
+    errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::WARNING );
+    return info;
+  }
+
+  UInt32 nRanges = dataSize / sizeof( AudioValueRange );
+  AudioValueRange rangeList[ nRanges ];
+  result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );
+  if ( result != kAudioHardwareNoError ) {
+    errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::WARNING );
+    return info;
+  }
+
+  // The sample rate reporting mechanism is a bit of a mystery.  It
+  // seems that it can either return individual rates or a range of
+  // rates.  I assume that if the min / max range values are the same,
+  // then that represents a single supported rate and if the min / max
+  // range values are different, the device supports an arbitrary
+  // range of values (though there might be multiple ranges, so we'll
+  // use the most conservative range).
+  Float64 minimumRate = 1.0, maximumRate = 10000000000.0;
+  bool haveValueRange = false;
+  info.sampleRates.clear();
+  for ( UInt32 i=0; i<nRanges; i++ ) {
+    if ( rangeList[i].mMinimum == rangeList[i].mMaximum ) {
+      unsigned int tmpSr = (unsigned int) rangeList[i].mMinimum;
+      info.sampleRates.push_back( tmpSr );
+
+      if ( !info.preferredSampleRate || ( tmpSr <= 48000 && tmpSr > info.preferredSampleRate ) )
+        info.preferredSampleRate = tmpSr;
+
+    } else {
+      haveValueRange = true;
+      if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;
+      if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;
+    }
+  }
+
+  if ( haveValueRange ) {
+    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.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
+          info.preferredSampleRate = SAMPLE_RATES[k];
+      }
+    }
+  }
+
+  // Sort and remove any redundant values
+  std::sort( info.sampleRates.begin(), info.sampleRates.end() );
+  info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );
+
+  if ( info.sampleRates.size() == 0 ) {
+    errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::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 ( info.outputChannels > 0 )
+    if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
+  if ( info.inputChannels > 0 )
+    if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
+
+  info.probed = true;
+  return info;
+}
+
+static 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;
+}
+
+static OSStatus xrunListener( AudioObjectID /*inDevice*/,
+                              UInt32 nAddresses,
+                              const AudioObjectPropertyAddress properties[],
+                              void* handlePointer )
+{
+  CoreHandle *handle = (CoreHandle *) handlePointer;
+  for ( UInt32 i=0; i<nAddresses; i++ ) {
+    if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) {
+      if ( properties[i].mScope == kAudioDevicePropertyScopeInput )
+        handle->xrun[1] = true;
+      else
+        handle->xrun[0] = true;
+    }
+  }
+
+  return kAudioHardwareNoError;
+}
+
+static OSStatus rateListener( AudioObjectID inDevice,
+                              UInt32 /*nAddresses*/,
+                              const AudioObjectPropertyAddress /*properties*/[],
+                              void* ratePointer )
+{
+  Float64 *rate = (Float64 *) ratePointer;
+  UInt32 dataSize = sizeof( Float64 );
+  AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,
+                                          kAudioObjectPropertyScopeGlobal,
+                                          kAudioObjectPropertyElementMaster };
+  AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate );
+  return kAudioHardwareNoError;
+}
+
+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;
+  AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+                                          kAudioObjectPropertyScopeGlobal,
+                                          kAudioObjectPropertyElementMaster };
+  OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
+                                                0, NULL, &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;
+    property.mScope = kAudioDevicePropertyScopeInput;
+  }
+  else
+    property.mScope = kAudioDevicePropertyScopeOutput;
+
+  // Get the stream "configuration".
+  AudioBufferList	*bufferList = nil;
+  dataSize = 0;
+  property.mSelector = kAudioDevicePropertyStreamConfiguration;
+  result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+  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 = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+  if (result != noErr || dataSize == 0) {
+    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 );
+  property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
+  result = AudioObjectGetPropertyData( id, &property, 0, NULL, &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 );
+  property.mSelector = kAudioDevicePropertyBufferFrameSize;
+  result = AudioObjectSetPropertyData( id, &property, 0, NULL, 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;
+
+  // Try to set "hog" mode ... it's not clear to me this is working.
+  if ( options && options->flags & RTAUDIO_HOG_DEVICE ) {
+    pid_t hog_pid;
+    dataSize = sizeof( hog_pid );
+    property.mSelector = kAudioDevicePropertyHogMode;
+    result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid );
+    if ( result != noErr ) {
+      errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+
+    if ( hog_pid != getpid() ) {
+      hog_pid = getpid();
+      result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid );
+      if ( result != noErr ) {
+        errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
+        errorText_ = errorStream_.str();
+        return FAILURE;
+      }
+    }
+  }
+
+  // Check and if necessary, change the sample rate for the device.
+  Float64 nominalRate;
+  dataSize = sizeof( Float64 );
+  property.mSelector = kAudioDevicePropertyNominalSampleRate;
+  result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );
+  if ( result != noErr ) {
+    errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  // Only change the sample rate if off by more than 1 Hz.
+  if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {
+
+    // Set a property listener for the sample rate change
+    Float64 reportedRate = 0.0;
+    AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+    result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
+    if ( result != noErr ) {
+      errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+
+    nominalRate = (Float64) sampleRate;
+    result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );
+    if ( result != noErr ) {
+      AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
+      errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+
+    // Now wait until the reported nominal rate is what we just set.
+    UInt32 microCounter = 0;
+    while ( reportedRate != nominalRate ) {
+      microCounter += 5000;
+      if ( microCounter > 5000000 ) break;
+      usleep( 5000 );
+    }
+
+    // Remove the property listener.
+    AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
+
+    if ( microCounter > 5000000 ) {
+      errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+  }
+
+  // Now set the stream format for all streams.  Also, check the
+  // physical format of the device and change that if necessary.
+  AudioStreamBasicDescription	description;
+  dataSize = sizeof( AudioStreamBasicDescription );
+  property.mSelector = kAudioStreamPropertyVirtualFormat;
+  result = AudioObjectGetPropertyData( id, &property, 0, NULL, &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.
+  bool updateFormat = false;
+  if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) {
+    description.mSampleRate = (Float64) sampleRate;
+    updateFormat = true;
+  }
+
+  if ( description.mFormatID != kAudioFormatLinearPCM ) {
+    description.mFormatID = kAudioFormatLinearPCM;
+    updateFormat = true;
+  }
+
+  if ( updateFormat ) {
+    result = AudioObjectSetPropertyData( id, &property, 0, NULL, 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.
+  property.mSelector = kAudioStreamPropertyPhysicalFormat;
+  result = AudioObjectGetPropertyData( id, &property, 0, NULL,  &dataSize, &description );
+  if ( result != noErr ) {
+    errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+
+  //std::cout << "Current physical stream format:" << std::endl;
+  //std::cout << "   mBitsPerChan = " << description.mBitsPerChannel << std::endl;
+  //std::cout << "   aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
+  //std::cout << "   bytesPerFrame = " << description.mBytesPerFrame << std::endl;
+  //std::cout << "   sample rate = " << description.mSampleRate << std::endl;
+
+  if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) {
+    description.mFormatID = kAudioFormatLinearPCM;
+    //description.mSampleRate = (Float64) sampleRate;
+    AudioStreamBasicDescription	testDescription = description;
+    UInt32 formatFlags;
+
+    // We'll try higher bit rates first and then work our way down.
+    std::vector< std::pair<UInt32, UInt32>  > physicalFormats;
+    formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;
+    physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
+    formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
+    physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
+    physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) );   // 24-bit packed
+    formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh );
+    physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low
+    formatFlags |= kAudioFormatFlagIsAlignedHigh;
+    physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high
+    formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
+    physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) );
+    physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) );
+
+    bool setPhysicalFormat = false;
+    for( unsigned int i=0; i<physicalFormats.size(); i++ ) {
+      testDescription = description;
+      testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first;
+      testDescription.mFormatFlags = physicalFormats[i].second;
+      if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) )
+        testDescription.mBytesPerFrame =  4 * testDescription.mChannelsPerFrame;
+      else
+        testDescription.mBytesPerFrame =  testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
+      testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
+      result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription );
+      if ( result == noErr ) {
+        setPhysicalFormat = true;
+        //std::cout << "Updated physical stream format:" << std::endl;
+        //std::cout << "   mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;
+        //std::cout << "   aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
+        //std::cout << "   bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;
+        //std::cout << "   sample rate = " << testDescription.mSampleRate << std::endl;
+        break;
+      }
+    }
+
+    if ( !setPhysicalFormat ) {
+      errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+  } // done setting virtual/physical formats.
+
+  // Get the stream / device latency.
+  UInt32 latency;
+  dataSize = sizeof( UInt32 );
+  property.mSelector = kAudioDevicePropertyLatency;
+  if ( AudioObjectHasProperty( id, &property ) == true ) {
+    result = AudioObjectGetPropertyData( id, &property, 0, NULL, &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( RtAudioError::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 );
+  stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) );
+  memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) );
+  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.
+  property.mSelector = kAudioDeviceProcessorOverload;
+  property.mScope = kAudioObjectPropertyScopeGlobal;
+  result = AudioObjectAddPropertyListener( id, &property, xrunListener, (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;
+  }
+
+  stream_.state = STREAM_CLOSED;
+  return FAILURE;
+}
+
+void RtApiCore :: closeStream( void )
+{
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiCore::closeStream(): no open stream to close!";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+    if (handle) {
+      AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+        kAudioObjectPropertyScopeGlobal,
+        kAudioObjectPropertyElementMaster };
+
+      property.mSelector = kAudioDeviceProcessorOverload;
+      property.mScope = kAudioObjectPropertyScopeGlobal;
+      if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) {
+        errorText_ = "RtApiCore::closeStream(): error removing property listener!";
+        error( RtAudioError::WARNING );
+      }
+
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+      if ( stream_.state == STREAM_RUNNING )
+        AudioDeviceStop( handle->id[0], handle->procId[0] );
+      AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
+#else // deprecated behaviour
+      if ( stream_.state == STREAM_RUNNING )
+        AudioDeviceStop( handle->id[0], callbackHandler );
+      AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
+#endif
+    }
+  }
+
+  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+    if (handle) {
+      AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+        kAudioObjectPropertyScopeGlobal,
+        kAudioObjectPropertyElementMaster };
+
+      property.mSelector = kAudioDeviceProcessorOverload;
+      property.mScope = kAudioObjectPropertyScopeGlobal;
+      if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) {
+        errorText_ = "RtApiCore::closeStream(): error removing property listener!";
+        error( RtAudioError::WARNING );
+      }
+
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+      if ( stream_.state == STREAM_RUNNING )
+        AudioDeviceStop( handle->id[1], handle->procId[1] );
+      AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
+#else // deprecated behaviour
+      if ( stream_.state == STREAM_RUNNING )
+        AudioDeviceStop( handle->id[1], callbackHandler );
+      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( RtAudioError::WARNING );
+    return;
+  }
+
+#if defined( HAVE_GETTIMEOFDAY )
+  gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
+
+  OSStatus result = noErr;
+  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+    result = AudioDeviceStart( handle->id[0], handle->procId[0] );
+#else // deprecated behaviour
+    result = AudioDeviceStart( handle->id[0], callbackHandler );
+#endif
+    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] ) ) {
+
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+    result = AudioDeviceStart( handle->id[1], handle->procId[1] );
+#else // deprecated behaviour
+    result = AudioDeviceStart( handle->id[1], callbackHandler );
+#endif
+    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:
+  if ( result == noErr ) return;
+  error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiCore :: stopStream( void )
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  OSStatus result = noErr;
+  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+    if ( handle->drainCounter == 0 ) {
+      handle->drainCounter = 2;
+      pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
+    }
+
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+    result = AudioDeviceStop( handle->id[0], handle->procId[0] );
+#else // deprecated behaviour
+    result = AudioDeviceStop( handle->id[0], callbackHandler );
+#endif
+    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] ) ) {
+
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+    result = AudioDeviceStop( handle->id[1], handle->procId[1] );
+#else  // deprecated behaviour
+    result = AudioDeviceStop( handle->id[1], callbackHandler );
+#endif
+    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:
+  if ( result == noErr ) return;
+  error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiCore :: abortStream( void )
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+  handle->drainCounter = 2;
+
+  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 better to handle it this way because the
+// callbackEvent() function probably should return before the AudioDeviceStop()
+// function is called.
+static void *coreStopStream( void *ptr )
+{
+  CallbackInfo *info = (CallbackInfo *) ptr;
+  RtApiCore *object = (RtApiCore *) info->object;
+
+  object->stopStream();
+  pthread_exit( NULL );
+}
+
+bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
+                                 const AudioBufferList *inBufferList,
+                                 const AudioBufferList *outBufferList )
+{
+  if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
+    error( RtAudioError::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 ) {
+    ThreadHandle threadId;
+
+    stream_.state = STREAM_STOPPING;
+    if ( handle->internalDrain == true )
+      pthread_create( &threadId, NULL, coreStopStream, info );
+    else // external call to stopStream()
+      pthread_cond_signal( &handle->condition );
+    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;
+    }
+
+    int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+                                  stream_.bufferSize, streamTime, status, info->userData );
+    if ( cbReturnValue == 2 ) {
+      stream_.state = STREAM_STOPPING;
+      handle->drainCounter = 2;
+      abortStream();
+      return SUCCESS;
+    }
+    else if ( cbReturnValue == 1 ) {
+      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;
+        }
+      }
+    }
+  }
+
+  // Don't bother draining input
+  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 );
+
+  // Make sure to only tick duplex stream time once if using two devices
+  if ( stream_.mode != DUPLEX || (stream_.mode == DUPLEX && handle->id[0] != handle->id[1] && deviceId == handle->id[0] ) )
+    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>
+#include <cstdio>
+
+// 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; }
+};
+
+#if !defined(__RTAUDIO_DEBUG__)
+static void jackSilentError( const char * ) {};
+#endif
+
+RtApiJack :: RtApiJack()
+    :shouldAutoconnect_(true) {
+  // 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 ); //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, JACK_DEFAULT_AUDIO_TYPE, 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 ); //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( RtAudioError::WARNING );
+    return info;
+  }
+
+  const char **ports;
+  std::string port, previousPort;
+  unsigned int nPorts = 0, nDevices = 0;
+  ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 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 ) {
+    jack_client_close( client );
+    errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
+    error( RtAudioError::INVALID_USE );
+    return info;
+  }
+
+  // Get the current jack server sample rate.
+  info.sampleRates.clear();
+
+  info.preferredSampleRate = jack_get_sample_rate( client );
+  info.sampleRates.push_back( info.preferredSampleRate );
+
+  // 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(), JACK_DEFAULT_AUDIO_TYPE, 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(), JACK_DEFAULT_AUDIO_TYPE, 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( RtAudioError::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;
+}
+
+static 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;
+}
+
+// This function will be called by a spawned thread when the Jack
+// server signals that it is shutting down.  It is necessary to handle
+// it this way because the jackShutdown() function must return before
+// the jack_deactivate() function (in closeStream()) will return.
+static void *jackCloseStream( void *ptr )
+{
+  CallbackInfo *info = (CallbackInfo *) ptr;
+  RtApiJack *object = (RtApiJack *) info->object;
+
+  object->closeStream();
+
+  pthread_exit( NULL );
+}
+static 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;
+
+  ThreadHandle threadId;
+  pthread_create( &threadId, NULL, jackCloseStream, info );
+  std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
+}
+
+static 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 ); //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( RtAudioError::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, JACK_DEFAULT_AUDIO_TYPE, 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;
+  }
+
+  unsigned long flag = JackPortIsInput;
+  if ( mode == INPUT ) flag = JackPortIsOutput;
+
+  if ( ! (options && (options->flags & RTAUDIO_JACK_DONT_CONNECT)) ) {
+    // 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, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, 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(), JACK_DEFAULT_AUDIO_TYPE, flag );
+  if ( ports[ firstChannel ] ) {
+    // Added by Ge Wang
+    jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);
+    // the range (usually the min and max are equal)
+    jack_latency_range_t latrange; latrange.min = latrange.max = 0;
+    // get the latency range
+    jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );
+    // be optimistic, use the min!
+    stream_.latency[mode] = latrange.min;
+    //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 *) &stream_.apiHandle );
+    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 );
+
+  if ( options && options->flags & RTAUDIO_JACK_DONT_CONNECT ) shouldAutoconnect_ = false;
+
+  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( RtAudioError::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( RtAudioError::WARNING );
+    return;
+  }
+
+  #if defined( HAVE_GETTIMEOFDAY )
+  gettimeofday( &stream_.lastTickTimestamp, NULL );
+  #endif
+
+  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 ( shouldAutoconnect_ && (stream_.mode == OUTPUT || stream_.mode == DUPLEX) ) {
+    result = 1;
+    ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), JACK_DEFAULT_AUDIO_TYPE, 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 ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) {
+    result = 1;
+    ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), JACK_DEFAULT_AUDIO_TYPE, 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:
+  if ( result == 0 ) return;
+  error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiJack :: stopStream( void )
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  JackHandle *handle = (JackHandle *) stream_.apiHandle;
+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+    if ( handle->drainCounter == 0 ) {
+      handle->drainCounter = 2;
+      pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
+    }
+  }
+
+  jack_deactivate( handle->client );
+  stream_.state = STREAM_STOPPED;
+}
+
+void RtApiJack :: abortStream( void )
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  JackHandle *handle = (JackHandle *) stream_.apiHandle;
+  handle->drainCounter = 2;
+
+  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.
+static 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 || stream_.state == STREAM_STOPPING ) return SUCCESS;
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
+    error( RtAudioError::WARNING );
+    return FAILURE;
+  }
+  if ( stream_.bufferSize != nframes ) {
+    errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
+    error( RtAudioError::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 ) {
+    ThreadHandle threadId;
+
+    stream_.state = STREAM_STOPPING;
+    if ( handle->internalDrain == true )
+      pthread_create( &threadId, NULL, jackStopStream, info );
+    else
+      pthread_cond_signal( &handle->condition );
+    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;
+    }
+    int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+                                  stream_.bufferSize, streamTime, status, info->userData );
+    if ( cbReturnValue == 2 ) {
+      stream_.state = STREAM_STOPPING;
+      handle->drainCounter = 2;
+      ThreadHandle id;
+      pthread_create( &id, NULL, jackStopStream, info );
+      return SUCCESS;
+    }
+    else if ( cbReturnValue == 1 ) {
+      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 > 1 ) { // 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 );
+      }
+    }
+  }
+
+  // Don't bother draining input
+  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:
+  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>
+
+static AsioDrivers drivers;
+static ASIOCallbacks asioCallbacks;
+static ASIODriverInfo driverInfo;
+static CallbackInfo *asioCallbackInfo;
+static 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 );
+static void sampleRateChanged( ASIOSampleRate sRate );
+static 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( RtAudioError::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( RtAudioError::INVALID_USE );
+    return info;
+  }
+
+  if ( device >= nDevices ) {
+    errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
+    error( RtAudioError::INVALID_USE );
+    return info;
+  }
+
+  // If a stream is already open, we cannot probe other devices.  Thus, use the saved results.
+  if ( stream_.state != STREAM_CLOSED ) {
+    if ( device >= devices_.size() ) {
+      errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
+      error( RtAudioError::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( RtAudioError::WARNING );
+    return info;
+  }
+
+  info.name = driverName;
+
+  if ( !drivers.loadDriver( driverName ) ) {
+    errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::WARNING );
+    return info;
+  }
+
+  result = ASIOInit( &driverInfo );
+  if ( result != ASE_OK ) {
+    errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::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( RtAudioError::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] );
+
+      if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
+        info.preferredSampleRate = 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( RtAudioError::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;
+  else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )
+    info.nativeFormats |= RTAUDIO_SINT24;
+
+  if ( info.outputChannels > 0 )
+    if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
+  if ( info.inputChannels > 0 )
+    if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
+
+  info.probed = true;
+  drivers.removeCurrentDriver();
+  return info;
+}
+
+static 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 )
+{////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
+
+  bool isDuplexInput =  mode == INPUT && stream_.mode == OUTPUT;
+
+  // For ASIO, a duplex stream MUST use the same driver.
+  if ( isDuplexInput && 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;
+  }
+
+  // Only load the driver once for duplex stream.
+  if ( !isDuplexInput ) {
+    // 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();
+
+    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;
+    }
+  }
+
+  // keep them before any "goto error", they are used for error cleanup + goto device boundary checks
+  bool buffersAllocated = false;
+  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+  unsigned int nChannels;
+
+
+  // Check the device channel count.
+  long inputChannels, outputChannels;
+  result = ASIOGetChannels( &inputChannels, &outputChannels );
+  if ( result != ASE_OK ) {
+    errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
+    errorText_ = errorStream_.str();
+    goto error;
+  }
+
+  if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
+       ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
+    errorText_ = errorStream_.str();
+    goto error;
+  }
+  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 ) {
+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
+    errorText_ = errorStream_.str();
+    goto error;
+  }
+
+  // Get the current sample rate
+  ASIOSampleRate currentRate;
+  result = ASIOGetSampleRate( &currentRate );
+  if ( result != ASE_OK ) {
+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
+    errorText_ = errorStream_.str();
+    goto error;
+  }
+
+  // Set the sample rate only if necessary
+  if ( currentRate != sampleRate ) {
+    result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
+    if ( result != ASE_OK ) {
+      errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
+      errorText_ = errorStream_.str();
+      goto error;
+    }
+  }
+
+  // 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 ) {
+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
+    errorText_ = errorStream_.str();
+    goto error;
+  }
+
+  // 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;
+  }
+  else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {
+    stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+    if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;
+  }
+
+  if ( stream_.deviceFormat[mode] == 0 ) {
+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
+    errorText_ = errorStream_.str();
+    goto error;
+  }
+
+  // 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 ) {
+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
+    errorText_ = errorStream_.str();
+    goto error;
+  }
+
+  if ( isDuplexInput ) {
+    // When this is the duplex input (output was opened before), then we have to use the same
+    // buffersize as the output, because it might use the preferred buffer size, which most
+    // likely wasn't passed as input to this. The buffer sizes have to be identically anyway,
+    // So instead of throwing an error, make them equal. The caller uses the reference
+    // to the "bufferSize" param as usual to set up processing buffers.
+
+    *bufferSize = stream_.bufferSize;
+
+  } else {
+    if ( *bufferSize == 0 ) *bufferSize = preferSize;
+    else 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;
+    }
+  }
+
+  /*
+  // we don't use it anymore, see above!
+  // Just left it here for the case...
+  if ( isDuplexInput && stream_.bufferSize != *bufferSize ) {
+    errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
+    goto error;
+  }
+  */
+
+  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.
+  if ( handle == 0 ) {
+    try {
+      handle = new AsioHandle;
+    }
+    catch ( std::bad_alloc& ) {
+      errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
+      goto error;
+    }
+    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.
+  if ( mode == INPUT && stream_.mode == OUTPUT ) {
+    ASIODisposeBuffers();
+    if ( handle->bufferInfos ) free( handle->bufferInfos );
+  }
+
+  // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
+  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;
+  }
+
+  // prepare for callbacks
+  stream_.sampleRate = sampleRate;
+  stream_.device[mode] = device;
+  stream_.mode = isDuplexInput ? DUPLEX : mode;
+
+  // store this class instance before registering callbacks, that are going to use it
+  asioCallbackInfo = &stream_.callbackInfo;
+  stream_.callbackInfo.object = (void *) this;
+
+  // 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 ) {
+    // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges
+    // but only accept the preferred buffer size as parameter for ASIOCreateBuffers (e.g. Creative's ASIO driver).
+    // In that case, let's be naïve and try that instead.
+    *bufferSize = preferSize;
+    stream_.bufferSize = *bufferSize;
+    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;  
+  stream_.state = STREAM_STOPPED;
+
+  // 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 ( isDuplexInput && 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;
+      }
+    }
+  }
+
+  // Determine device latencies
+  long inputLatency, outputLatency;
+  result = ASIOGetLatencies( &inputLatency, &outputLatency );
+  if ( result != ASE_OK ) {
+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::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 ( !isDuplexInput ) {
+    // the cleanup for error in the duplex input, is done by RtApi::openStream
+    // So we clean up for single channel only
+
+    if ( buffersAllocated )
+      ASIODisposeBuffers();
+
+    drivers.removeCurrentDriver();
+
+    if ( handle ) {
+      CloseHandle( handle->condition );
+      if ( handle->bufferInfos )
+        free( handle->bufferInfos );
+
+      delete handle;
+      stream_.apiHandle = 0;
+    }
+
+
+    if ( stream_.userBuffer[mode] ) {
+      free( stream_.userBuffer[mode] );
+      stream_.userBuffer[mode] = 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( RtAudioError::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;
+}
+
+bool stopThreadCalled = false;
+
+void RtApiAsio :: startStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_RUNNING ) {
+    errorText_ = "RtApiAsio::startStream(): the stream is already running!";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  #if defined( HAVE_GETTIMEOFDAY )
+  gettimeofday( &stream_.lastTickTimestamp, NULL );
+  #endif
+
+  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;
+  ResetEvent( handle->condition );
+  stream_.state = STREAM_RUNNING;
+  asioXRun = false;
+
+ unlock:
+  stopThreadCalled = false;
+
+  if ( result == ASE_OK ) return;
+  error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiAsio :: stopStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+    if ( handle->drainCounter == 0 ) {
+      handle->drainCounter = 2;
+      WaitForSingleObject( handle->condition, INFINITE );  // block until signaled
+    }
+  }
+
+  stream_.state = STREAM_STOPPED;
+
+  ASIOError result = ASIOStop();
+  if ( result != ASE_OK ) {
+    errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
+    errorText_ = errorStream_.str();
+  }
+
+  if ( result == ASE_OK ) return;
+  error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiAsio :: abortStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
+    error( RtAudioError::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 = 2;
+  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 ASIOStop()
+// function will return.
+static unsigned __stdcall asioStopStream( void *ptr )
+{
+  CallbackInfo *info = (CallbackInfo *) ptr;
+  RtApiAsio *object = (RtApiAsio *) info->object;
+
+  object->stopStream();
+  _endthreadex( 0 );
+  return 0;
+}
+
+bool RtApiAsio :: callbackEvent( long bufferIndex )
+{
+  if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
+    error( RtAudioError::WARNING );
+    return FAILURE;
+  }
+
+  CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+
+  // Check if we were draining the stream and signal if finished.
+  if ( handle->drainCounter > 3 ) {
+
+    stream_.state = STREAM_STOPPING;
+    if ( handle->internalDrain == false )
+      SetEvent( handle->condition );
+    else { // spawn a thread to stop the stream
+      unsigned threadId;
+      stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
+                                                    &stream_.callbackInfo, 0, &threadId );
+    }
+    return SUCCESS;
+  }
+
+  // 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;
+    }
+    int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+                                     stream_.bufferSize, streamTime, status, info->userData );
+    if ( cbReturnValue == 2 ) {
+      stream_.state = STREAM_STOPPING;
+      handle->drainCounter = 2;
+      unsigned threadId;
+      stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
+                                                    &stream_.callbackInfo, 0, &threadId );
+      return SUCCESS;
+    }
+    else if ( cbReturnValue == 1 ) {
+      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 );
+      }
+
+    }
+  }
+
+  // Don't bother draining input
+  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();
+
+  RtApi::tickStreamTime();
+  return SUCCESS;
+}
+
+static 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 ( RtAudioError &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;
+}
+
+static 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 const 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_WASAPI__) // Windows WASAPI API
+
+// Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014
+// - Introduces support for the Windows WASAPI API
+// - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required
+// - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface
+// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
+
+#ifndef INITGUID
+  #define INITGUID
+#endif
+
+#include <mfapi.h>
+#include <mferror.h>
+#include <mfplay.h>
+#include <mftransform.h>
+#include <wmcodecdsp.h>
+
+#include <audioclient.h>
+#include <avrt.h>
+#include <mmdeviceapi.h>
+#include <functiondiscoverykeys_devpkey.h>
+
+#ifndef MF_E_TRANSFORM_NEED_MORE_INPUT
+  #define MF_E_TRANSFORM_NEED_MORE_INPUT _HRESULT_TYPEDEF_(0xc00d6d72)
+#endif
+
+#ifndef MFSTARTUP_NOSOCKET
+  #define MFSTARTUP_NOSOCKET 0x1
+#endif
+
+#ifdef _MSC_VER
+  #pragma comment( lib, "ksuser" )
+  #pragma comment( lib, "mfplat.lib" )
+  #pragma comment( lib, "mfuuid.lib" )
+  #pragma comment( lib, "wmcodecdspuuid" )
+#endif
+
+//=============================================================================
+
+#define SAFE_RELEASE( objectPtr )\
+if ( objectPtr )\
+{\
+  objectPtr->Release();\
+  objectPtr = NULL;\
+}
+
+typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
+
+//-----------------------------------------------------------------------------
+
+// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
+// Therefore we must perform all necessary conversions to user buffers in order to satisfy these
+// requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
+// provide intermediate storage for read / write synchronization.
+class WasapiBuffer
+{
+public:
+  WasapiBuffer()
+    : buffer_( NULL ),
+      bufferSize_( 0 ),
+      inIndex_( 0 ),
+      outIndex_( 0 ) {}
+
+  ~WasapiBuffer() {
+    free( buffer_ );
+  }
+
+  // sets the length of the internal ring buffer
+  void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
+    free( buffer_ );
+
+    buffer_ = ( char* ) calloc( bufferSize, formatBytes );
+
+    bufferSize_ = bufferSize;
+    inIndex_ = 0;
+    outIndex_ = 0;
+  }
+
+  // attempt to push a buffer into the ring buffer at the current "in" index
+  bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
+  {
+    if ( !buffer ||                 // incoming buffer is NULL
+         bufferSize == 0 ||         // incoming buffer has no data
+         bufferSize > bufferSize_ ) // incoming buffer too large
+    {
+      return false;
+    }
+
+    unsigned int relOutIndex = outIndex_;
+    unsigned int inIndexEnd = inIndex_ + bufferSize;
+    if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
+      relOutIndex += bufferSize_;
+    }
+
+    // the "IN" index CAN BEGIN at the "OUT" index
+    // the "IN" index CANNOT END at the "OUT" index
+    if ( inIndex_ < relOutIndex && inIndexEnd >= relOutIndex ) {
+      return false; // not enough space between "in" index and "out" index
+    }
+
+    // copy buffer from external to internal
+    int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
+    fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
+    int fromInSize = bufferSize - fromZeroSize;
+
+    switch( format )
+      {
+      case RTAUDIO_SINT8:
+        memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
+        memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
+        break;
+      case RTAUDIO_SINT16:
+        memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
+        memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
+        break;
+      case RTAUDIO_SINT24:
+        memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
+        memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
+        break;
+      case RTAUDIO_SINT32:
+        memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
+        memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
+        break;
+      case RTAUDIO_FLOAT32:
+        memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
+        memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
+        break;
+      case RTAUDIO_FLOAT64:
+        memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
+        memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
+        break;
+    }
+
+    // update "in" index
+    inIndex_ += bufferSize;
+    inIndex_ %= bufferSize_;
+
+    return true;
+  }
+
+  // attempt to pull a buffer from the ring buffer from the current "out" index
+  bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
+  {
+    if ( !buffer ||                 // incoming buffer is NULL
+         bufferSize == 0 ||         // incoming buffer has no data
+         bufferSize > bufferSize_ ) // incoming buffer too large
+    {
+      return false;
+    }
+
+    unsigned int relInIndex = inIndex_;
+    unsigned int outIndexEnd = outIndex_ + bufferSize;
+    if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
+      relInIndex += bufferSize_;
+    }
+
+    // the "OUT" index CANNOT BEGIN at the "IN" index
+    // the "OUT" index CAN END at the "IN" index
+    if ( outIndex_ <= relInIndex && outIndexEnd > relInIndex ) {
+      return false; // not enough space between "out" index and "in" index
+    }
+
+    // copy buffer from internal to external
+    int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
+    fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
+    int fromOutSize = bufferSize - fromZeroSize;
+
+    switch( format )
+    {
+      case RTAUDIO_SINT8:
+        memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
+        memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
+        break;
+      case RTAUDIO_SINT16:
+        memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
+        memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
+        break;
+      case RTAUDIO_SINT24:
+        memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
+        memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
+        break;
+      case RTAUDIO_SINT32:
+        memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
+        memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
+        break;
+      case RTAUDIO_FLOAT32:
+        memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
+        memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
+        break;
+      case RTAUDIO_FLOAT64:
+        memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
+        memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
+        break;
+    }
+
+    // update "out" index
+    outIndex_ += bufferSize;
+    outIndex_ %= bufferSize_;
+
+    return true;
+  }
+
+private:
+  char* buffer_;
+  unsigned int bufferSize_;
+  unsigned int inIndex_;
+  unsigned int outIndex_;
+};
+
+//-----------------------------------------------------------------------------
+
+// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
+// between HW and the user. The WasapiResampler class is used to perform this conversion between
+// HwIn->UserIn and UserOut->HwOut during the stream callback loop.
+class WasapiResampler
+{
+public:
+  WasapiResampler( bool isFloat, unsigned int bitsPerSample, unsigned int channelCount,
+                   unsigned int inSampleRate, unsigned int outSampleRate )
+    : _bytesPerSample( bitsPerSample / 8 )
+    , _channelCount( channelCount )
+    , _sampleRatio( ( float ) outSampleRate / inSampleRate )
+    , _transformUnk( NULL )
+    , _transform( NULL )
+    , _mediaType( NULL )
+    , _inputMediaType( NULL )
+    , _outputMediaType( NULL )
+
+    #ifdef __IWMResamplerProps_FWD_DEFINED__
+      , _resamplerProps( NULL )
+    #endif
+  {
+    // 1. Initialization
+
+    MFStartup( MF_VERSION, MFSTARTUP_NOSOCKET );
+
+    // 2. Create Resampler Transform Object
+
+    CoCreateInstance( CLSID_CResamplerMediaObject, NULL, CLSCTX_INPROC_SERVER,
+                      IID_IUnknown, ( void** ) &_transformUnk );
+
+    _transformUnk->QueryInterface( IID_PPV_ARGS( &_transform ) );
+
+    #ifdef __IWMResamplerProps_FWD_DEFINED__
+      _transformUnk->QueryInterface( IID_PPV_ARGS( &_resamplerProps ) );
+      _resamplerProps->SetHalfFilterLength( 60 ); // best conversion quality
+    #endif
+
+    // 3. Specify input / output format
+
+    MFCreateMediaType( &_mediaType );
+    _mediaType->SetGUID( MF_MT_MAJOR_TYPE, MFMediaType_Audio );
+    _mediaType->SetGUID( MF_MT_SUBTYPE, isFloat ? MFAudioFormat_Float : MFAudioFormat_PCM );
+    _mediaType->SetUINT32( MF_MT_AUDIO_NUM_CHANNELS, channelCount );
+    _mediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, inSampleRate );
+    _mediaType->SetUINT32( MF_MT_AUDIO_BLOCK_ALIGNMENT, _bytesPerSample * channelCount );
+    _mediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * inSampleRate );
+    _mediaType->SetUINT32( MF_MT_AUDIO_BITS_PER_SAMPLE, bitsPerSample );
+    _mediaType->SetUINT32( MF_MT_ALL_SAMPLES_INDEPENDENT, TRUE );
+
+    MFCreateMediaType( &_inputMediaType );
+    _mediaType->CopyAllItems( _inputMediaType );
+
+    _transform->SetInputType( 0, _inputMediaType, 0 );
+
+    MFCreateMediaType( &_outputMediaType );
+    _mediaType->CopyAllItems( _outputMediaType );
+
+    _outputMediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, outSampleRate );
+    _outputMediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * outSampleRate );
+
+    _transform->SetOutputType( 0, _outputMediaType, 0 );
+
+    // 4. Send stream start messages to Resampler
+
+    _transform->ProcessMessage( MFT_MESSAGE_COMMAND_FLUSH, 0 );
+    _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_BEGIN_STREAMING, 0 );
+    _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_START_OF_STREAM, 0 );
+  }
+
+  ~WasapiResampler()
+  {
+    // 8. Send stream stop messages to Resampler
+
+    _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_OF_STREAM, 0 );
+    _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_STREAMING, 0 );
+
+    // 9. Cleanup
+
+    MFShutdown();
+
+    SAFE_RELEASE( _transformUnk );
+    SAFE_RELEASE( _transform );
+    SAFE_RELEASE( _mediaType );
+    SAFE_RELEASE( _inputMediaType );
+    SAFE_RELEASE( _outputMediaType );
+
+    #ifdef __IWMResamplerProps_FWD_DEFINED__
+      SAFE_RELEASE( _resamplerProps );
+    #endif
+  }
+
+  void Convert( char* outBuffer, const char* inBuffer, unsigned int inSampleCount, unsigned int& outSampleCount, int maxOutSampleCount = -1 )
+  {
+    unsigned int inputBufferSize = _bytesPerSample * _channelCount * inSampleCount;
+    if ( _sampleRatio == 1 )
+    {
+      // no sample rate conversion required
+      memcpy( outBuffer, inBuffer, inputBufferSize );
+      outSampleCount = inSampleCount;
+      return;
+    }
+
+    unsigned int outputBufferSize = 0;
+    if ( maxOutSampleCount != -1 )
+    {
+      outputBufferSize = _bytesPerSample * _channelCount * maxOutSampleCount;
+    }
+    else
+    {
+      outputBufferSize = ( unsigned int ) ceilf( inputBufferSize * _sampleRatio ) + ( _bytesPerSample * _channelCount );
+    }
+
+    IMFMediaBuffer* rInBuffer;
+    IMFSample* rInSample;
+    BYTE* rInByteBuffer = NULL;
+
+    // 5. Create Sample object from input data
+
+    MFCreateMemoryBuffer( inputBufferSize, &rInBuffer );
+
+    rInBuffer->Lock( &rInByteBuffer, NULL, NULL );
+    memcpy( rInByteBuffer, inBuffer, inputBufferSize );
+    rInBuffer->Unlock();
+    rInByteBuffer = NULL;
+
+    rInBuffer->SetCurrentLength( inputBufferSize );
+
+    MFCreateSample( &rInSample );
+    rInSample->AddBuffer( rInBuffer );
+
+    // 6. Pass input data to Resampler
+
+    _transform->ProcessInput( 0, rInSample, 0 );
+
+    SAFE_RELEASE( rInBuffer );
+    SAFE_RELEASE( rInSample );
+
+    // 7. Perform sample rate conversion
+
+    IMFMediaBuffer* rOutBuffer = NULL;
+    BYTE* rOutByteBuffer = NULL;
+
+    MFT_OUTPUT_DATA_BUFFER rOutDataBuffer;
+    DWORD rStatus;
+    DWORD rBytes = outputBufferSize; // maximum bytes accepted per ProcessOutput
+
+    // 7.1 Create Sample object for output data
+
+    memset( &rOutDataBuffer, 0, sizeof rOutDataBuffer );
+    MFCreateSample( &( rOutDataBuffer.pSample ) );
+    MFCreateMemoryBuffer( rBytes, &rOutBuffer );
+    rOutDataBuffer.pSample->AddBuffer( rOutBuffer );
+    rOutDataBuffer.dwStreamID = 0;
+    rOutDataBuffer.dwStatus = 0;
+    rOutDataBuffer.pEvents = NULL;
+
+    // 7.2 Get output data from Resampler
+
+    if ( _transform->ProcessOutput( 0, 1, &rOutDataBuffer, &rStatus ) == MF_E_TRANSFORM_NEED_MORE_INPUT )
+    {
+      outSampleCount = 0;
+      SAFE_RELEASE( rOutBuffer );
+      SAFE_RELEASE( rOutDataBuffer.pSample );
+      return;
+    }
+
+    // 7.3 Write output data to outBuffer
+
+    SAFE_RELEASE( rOutBuffer );
+    rOutDataBuffer.pSample->ConvertToContiguousBuffer( &rOutBuffer );
+    rOutBuffer->GetCurrentLength( &rBytes );
+
+    rOutBuffer->Lock( &rOutByteBuffer, NULL, NULL );
+    memcpy( outBuffer, rOutByteBuffer, rBytes );
+    rOutBuffer->Unlock();
+    rOutByteBuffer = NULL;
+
+    outSampleCount = rBytes / _bytesPerSample / _channelCount;
+    SAFE_RELEASE( rOutBuffer );
+    SAFE_RELEASE( rOutDataBuffer.pSample );
+  }
+
+private:
+  unsigned int _bytesPerSample;
+  unsigned int _channelCount;
+  float _sampleRatio;
+
+  IUnknown* _transformUnk;
+  IMFTransform* _transform;
+  IMFMediaType* _mediaType;
+  IMFMediaType* _inputMediaType;
+  IMFMediaType* _outputMediaType;
+
+  #ifdef __IWMResamplerProps_FWD_DEFINED__
+    IWMResamplerProps* _resamplerProps;
+  #endif
+};
+
+//-----------------------------------------------------------------------------
+
+// A structure to hold various information related to the WASAPI implementation.
+struct WasapiHandle
+{
+  IAudioClient* captureAudioClient;
+  IAudioClient* renderAudioClient;
+  IAudioCaptureClient* captureClient;
+  IAudioRenderClient* renderClient;
+  HANDLE captureEvent;
+  HANDLE renderEvent;
+
+  WasapiHandle()
+  : captureAudioClient( NULL ),
+    renderAudioClient( NULL ),
+    captureClient( NULL ),
+    renderClient( NULL ),
+    captureEvent( NULL ),
+    renderEvent( NULL ) {}
+};
+
+//=============================================================================
+
+RtApiWasapi::RtApiWasapi()
+  : coInitialized_( false ), deviceEnumerator_( NULL )
+{
+  // WASAPI can run either apartment or multi-threaded
+  HRESULT hr = CoInitialize( NULL );
+  if ( !FAILED( hr ) )
+    coInitialized_ = true;
+
+  // Instantiate device enumerator
+  hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
+                         CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
+                         ( void** ) &deviceEnumerator_ );
+
+  // If this runs on an old Windows, it will fail. Ignore and proceed.
+  if ( FAILED( hr ) )
+    deviceEnumerator_ = NULL;
+}
+
+//-----------------------------------------------------------------------------
+
+RtApiWasapi::~RtApiWasapi()
+{
+  if ( stream_.state != STREAM_CLOSED )
+    closeStream();
+
+  SAFE_RELEASE( deviceEnumerator_ );
+
+  // If this object previously called CoInitialize()
+  if ( coInitialized_ )
+    CoUninitialize();
+}
+
+//=============================================================================
+
+unsigned int RtApiWasapi::getDeviceCount( void )
+{
+  unsigned int captureDeviceCount = 0;
+  unsigned int renderDeviceCount = 0;
+
+  IMMDeviceCollection* captureDevices = NULL;
+  IMMDeviceCollection* renderDevices = NULL;
+
+  if ( !deviceEnumerator_ )
+    return 0;
+
+  // Count capture devices
+  errorText_.clear();
+  HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+  if ( FAILED( hr ) ) {
+    errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
+    goto Exit;
+  }
+
+  hr = captureDevices->GetCount( &captureDeviceCount );
+  if ( FAILED( hr ) ) {
+    errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
+    goto Exit;
+  }
+
+  // Count render devices
+  hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+  if ( FAILED( hr ) ) {
+    errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
+    goto Exit;
+  }
+
+  hr = renderDevices->GetCount( &renderDeviceCount );
+  if ( FAILED( hr ) ) {
+    errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
+    goto Exit;
+  }
+
+Exit:
+  // release all references
+  SAFE_RELEASE( captureDevices );
+  SAFE_RELEASE( renderDevices );
+
+  if ( errorText_.empty() )
+    return captureDeviceCount + renderDeviceCount;
+
+  error( RtAudioError::DRIVER_ERROR );
+  return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
+{
+  RtAudio::DeviceInfo info;
+  unsigned int captureDeviceCount = 0;
+  unsigned int renderDeviceCount = 0;
+  std::string defaultDeviceName;
+  bool isCaptureDevice = false;
+
+  PROPVARIANT deviceNameProp;
+  PROPVARIANT defaultDeviceNameProp;
+
+  IMMDeviceCollection* captureDevices = NULL;
+  IMMDeviceCollection* renderDevices = NULL;
+  IMMDevice* devicePtr = NULL;
+  IMMDevice* defaultDevicePtr = NULL;
+  IAudioClient* audioClient = NULL;
+  IPropertyStore* devicePropStore = NULL;
+  IPropertyStore* defaultDevicePropStore = NULL;
+
+  WAVEFORMATEX* deviceFormat = NULL;
+  WAVEFORMATEX* closestMatchFormat = NULL;
+
+  // probed
+  info.probed = false;
+
+  // Count capture devices
+  errorText_.clear();
+  RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
+  HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+  if ( FAILED( hr ) ) {
+    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
+    goto Exit;
+  }
+
+  hr = captureDevices->GetCount( &captureDeviceCount );
+  if ( FAILED( hr ) ) {
+    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
+    goto Exit;
+  }
+
+  // Count render devices
+  hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+  if ( FAILED( hr ) ) {
+    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
+    goto Exit;
+  }
+
+  hr = renderDevices->GetCount( &renderDeviceCount );
+  if ( FAILED( hr ) ) {
+    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
+    goto Exit;
+  }
+
+  // validate device index
+  if ( device >= captureDeviceCount + renderDeviceCount ) {
+    errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
+    errorType = RtAudioError::INVALID_USE;
+    goto Exit;
+  }
+
+  // determine whether index falls within capture or render devices
+  if ( device >= renderDeviceCount ) {
+    hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
+    if ( FAILED( hr ) ) {
+      errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
+      goto Exit;
+    }
+    isCaptureDevice = true;
+  }
+  else {
+    hr = renderDevices->Item( device, &devicePtr );
+    if ( FAILED( hr ) ) {
+      errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
+      goto Exit;
+    }
+    isCaptureDevice = false;
+  }
+
+  // get default device name
+  if ( isCaptureDevice ) {
+    hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
+    if ( FAILED( hr ) ) {
+      errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
+      goto Exit;
+    }
+  }
+  else {
+    hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
+    if ( FAILED( hr ) ) {
+      errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
+      goto Exit;
+    }
+  }
+
+  hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
+  if ( FAILED( hr ) ) {
+    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
+    goto Exit;
+  }
+  PropVariantInit( &defaultDeviceNameProp );
+
+  hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
+  if ( FAILED( hr ) ) {
+    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
+    goto Exit;
+  }
+
+  defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);
+
+  // name
+  hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
+  if ( FAILED( hr ) ) {
+    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
+    goto Exit;
+  }
+
+  PropVariantInit( &deviceNameProp );
+
+  hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
+  if ( FAILED( hr ) ) {
+    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
+    goto Exit;
+  }
+
+  info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);
+
+  // is default
+  if ( isCaptureDevice ) {
+    info.isDefaultInput = info.name == defaultDeviceName;
+    info.isDefaultOutput = false;
+  }
+  else {
+    info.isDefaultInput = false;
+    info.isDefaultOutput = info.name == defaultDeviceName;
+  }
+
+  // channel count
+  hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
+  if ( FAILED( hr ) ) {
+    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
+    goto Exit;
+  }
+
+  hr = audioClient->GetMixFormat( &deviceFormat );
+  if ( FAILED( hr ) ) {
+    errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
+    goto Exit;
+  }
+
+  if ( isCaptureDevice ) {
+    info.inputChannels = deviceFormat->nChannels;
+    info.outputChannels = 0;
+    info.duplexChannels = 0;
+  }
+  else {
+    info.inputChannels = 0;
+    info.outputChannels = deviceFormat->nChannels;
+    info.duplexChannels = 0;
+  }
+
+  // sample rates
+  info.sampleRates.clear();
+
+  // allow support for all sample rates as we have a built-in sample rate converter
+  for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
+    info.sampleRates.push_back( SAMPLE_RATES[i] );
+  }
+  info.preferredSampleRate = deviceFormat->nSamplesPerSec;
+
+  // native format
+  info.nativeFormats = 0;
+
+  if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
+       ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
+         ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
+  {
+    if ( deviceFormat->wBitsPerSample == 32 ) {
+      info.nativeFormats |= RTAUDIO_FLOAT32;
+    }
+    else if ( deviceFormat->wBitsPerSample == 64 ) {
+      info.nativeFormats |= RTAUDIO_FLOAT64;
+    }
+  }
+  else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
+           ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
+             ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
+  {
+    if ( deviceFormat->wBitsPerSample == 8 ) {
+      info.nativeFormats |= RTAUDIO_SINT8;
+    }
+    else if ( deviceFormat->wBitsPerSample == 16 ) {
+      info.nativeFormats |= RTAUDIO_SINT16;
+    }
+    else if ( deviceFormat->wBitsPerSample == 24 ) {
+      info.nativeFormats |= RTAUDIO_SINT24;
+    }
+    else if ( deviceFormat->wBitsPerSample == 32 ) {
+      info.nativeFormats |= RTAUDIO_SINT32;
+    }
+  }
+
+  // probed
+  info.probed = true;
+
+Exit:
+  // release all references
+  PropVariantClear( &deviceNameProp );
+  PropVariantClear( &defaultDeviceNameProp );
+
+  SAFE_RELEASE( captureDevices );
+  SAFE_RELEASE( renderDevices );
+  SAFE_RELEASE( devicePtr );
+  SAFE_RELEASE( defaultDevicePtr );
+  SAFE_RELEASE( audioClient );
+  SAFE_RELEASE( devicePropStore );
+  SAFE_RELEASE( defaultDevicePropStore );
+
+  CoTaskMemFree( deviceFormat );
+  CoTaskMemFree( closestMatchFormat );
+
+  if ( !errorText_.empty() )
+    error( errorType );
+  return info;
+}
+
+//-----------------------------------------------------------------------------
+
+unsigned int RtApiWasapi::getDefaultOutputDevice( void )
+{
+  for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
+    if ( getDeviceInfo( i ).isDefaultOutput ) {
+      return i;
+    }
+  }
+
+  return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+unsigned int RtApiWasapi::getDefaultInputDevice( void )
+{
+  for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
+    if ( getDeviceInfo( i ).isDefaultInput ) {
+      return i;
+    }
+  }
+
+  return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::closeStream( void )
+{
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  if ( stream_.state != STREAM_STOPPED )
+    stopStream();
+
+  // clean up stream memory
+  SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
+  SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
+
+  SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
+  SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
+
+  if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
+    CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
+
+  if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
+    CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
+
+  delete ( WasapiHandle* ) stream_.apiHandle;
+  stream_.apiHandle = NULL;
+
+  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;
+  }
+
+  // update stream state
+  stream_.state = STREAM_CLOSED;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::startStream( void )
+{
+  verifyStream();
+
+  if ( stream_.state == STREAM_RUNNING ) {
+    errorText_ = "RtApiWasapi::startStream: The stream is already running.";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  #if defined( HAVE_GETTIMEOFDAY )
+  gettimeofday( &stream_.lastTickTimestamp, NULL );
+  #endif
+
+  // update stream state
+  stream_.state = STREAM_RUNNING;
+
+  // create WASAPI stream thread
+  stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
+
+  if ( !stream_.callbackInfo.thread ) {
+    errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
+    error( RtAudioError::THREAD_ERROR );
+  }
+  else {
+    SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
+    ResumeThread( ( void* ) stream_.callbackInfo.thread );
+  }
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::stopStream( void )
+{
+  verifyStream();
+
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  // inform stream thread by setting stream state to STREAM_STOPPING
+  stream_.state = STREAM_STOPPING;
+
+  // wait until stream thread is stopped
+  for (int i=0; i < 2 && stream_.state != STREAM_STOPPED; ++i ) {
+    Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
+  }
+
+  // close thread handle
+  if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
+    errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
+    error( RtAudioError::THREAD_ERROR );
+    return;
+  }
+
+  stream_.callbackInfo.thread = (ThreadHandle) NULL;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::abortStream( void )
+{
+  verifyStream();
+
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  // inform stream thread by setting stream state to STREAM_STOPPING
+  stream_.state = STREAM_STOPPING;
+
+  // wait until stream thread is stopped
+  while ( stream_.state != STREAM_STOPPED ) {
+    Sleep( 1 );
+  }
+
+  // close thread handle
+  if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
+    errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
+    error( RtAudioError::THREAD_ERROR );
+    return;
+  }
+
+  stream_.callbackInfo.thread = (ThreadHandle) NULL;
+}
+
+//-----------------------------------------------------------------------------
+
+bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+                                   unsigned int firstChannel, unsigned int sampleRate,
+                                   RtAudioFormat format, unsigned int* bufferSize,
+                                   RtAudio::StreamOptions* options )
+{
+  bool methodResult = FAILURE;
+  unsigned int captureDeviceCount = 0;
+  unsigned int renderDeviceCount = 0;
+
+  IMMDeviceCollection* captureDevices = NULL;
+  IMMDeviceCollection* renderDevices = NULL;
+  IMMDevice* devicePtr = NULL;
+  WAVEFORMATEX* deviceFormat = NULL;
+  unsigned int bufferBytes;
+  stream_.state = STREAM_STOPPED;
+
+  // create API Handle if not already created
+  if ( !stream_.apiHandle )
+    stream_.apiHandle = ( void* ) new WasapiHandle();
+
+  // Count capture devices
+  errorText_.clear();
+  RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
+  HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+  if ( FAILED( hr ) ) {
+    errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
+    goto Exit;
+  }
+
+  hr = captureDevices->GetCount( &captureDeviceCount );
+  if ( FAILED( hr ) ) {
+    errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
+    goto Exit;
+  }
+
+  // Count render devices
+  hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+  if ( FAILED( hr ) ) {
+    errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
+    goto Exit;
+  }
+
+  hr = renderDevices->GetCount( &renderDeviceCount );
+  if ( FAILED( hr ) ) {
+    errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
+    goto Exit;
+  }
+
+  // validate device index
+  if ( device >= captureDeviceCount + renderDeviceCount ) {
+    errorType = RtAudioError::INVALID_USE;
+    errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
+    goto Exit;
+  }
+
+  // if device index falls within capture devices
+  if ( device >= renderDeviceCount ) {
+    if ( mode != INPUT ) {
+      errorType = RtAudioError::INVALID_USE;
+      errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
+      goto Exit;
+    }
+
+    // retrieve captureAudioClient from devicePtr
+    IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
+
+    hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
+    if ( FAILED( hr ) ) {
+      errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
+      goto Exit;
+    }
+
+    hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+                              NULL, ( void** ) &captureAudioClient );
+    if ( FAILED( hr ) ) {
+      errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device audio client.";
+      goto Exit;
+    }
+
+    hr = captureAudioClient->GetMixFormat( &deviceFormat );
+    if ( FAILED( hr ) ) {
+      errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device mix format.";
+      goto Exit;
+    }
+
+    stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+    captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
+  }
+
+  // if device index falls within render devices and is configured for loopback
+  if ( device < renderDeviceCount && mode == INPUT )
+  {
+    // if renderAudioClient is not initialised, initialise it now
+    IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
+    if ( !renderAudioClient )
+    {
+      probeDeviceOpen( device, OUTPUT, channels, firstChannel, sampleRate, format, bufferSize, options );
+    }
+
+    // retrieve captureAudioClient from devicePtr
+    IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
+
+    hr = renderDevices->Item( device, &devicePtr );
+    if ( FAILED( hr ) ) {
+      errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
+      goto Exit;
+    }
+
+    hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+                              NULL, ( void** ) &captureAudioClient );
+    if ( FAILED( hr ) ) {
+      errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client.";
+      goto Exit;
+    }
+
+    hr = captureAudioClient->GetMixFormat( &deviceFormat );
+    if ( FAILED( hr ) ) {
+      errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format.";
+      goto Exit;
+    }
+
+    stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+    captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
+  }
+
+  // if device index falls within render devices and is configured for output
+  if ( device < renderDeviceCount && mode == OUTPUT )
+  {
+    // if renderAudioClient is already initialised, don't initialise it again
+    IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
+    if ( renderAudioClient )
+    {
+      methodResult = SUCCESS;
+      goto Exit;
+    }
+
+    hr = renderDevices->Item( device, &devicePtr );
+    if ( FAILED( hr ) ) {
+      errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
+      goto Exit;
+    }
+
+    hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+                              NULL, ( void** ) &renderAudioClient );
+    if ( FAILED( hr ) ) {
+      errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client.";
+      goto Exit;
+    }
+
+    hr = renderAudioClient->GetMixFormat( &deviceFormat );
+    if ( FAILED( hr ) ) {
+      errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format.";
+      goto Exit;
+    }
+
+    stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+    renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
+  }
+
+  // fill stream data
+  if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
+       ( stream_.mode == INPUT && mode == OUTPUT ) ) {
+    stream_.mode = DUPLEX;
+  }
+  else {
+    stream_.mode = mode;
+  }
+
+  stream_.device[mode] = device;
+  stream_.doByteSwap[mode] = false;
+  stream_.sampleRate = sampleRate;
+  stream_.bufferSize = *bufferSize;
+  stream_.nBuffers = 1;
+  stream_.nUserChannels[mode] = channels;
+  stream_.channelOffset[mode] = firstChannel;
+  stream_.userFormat = format;
+  stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
+
+  if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
+    stream_.userInterleaved = false;
+  else
+    stream_.userInterleaved = true;
+  stream_.deviceInterleaved[mode] = true;
+
+  // Set flags for buffer conversion.
+  stream_.doConvertBuffer[mode] = false;
+  if ( stream_.userFormat != stream_.deviceFormat[mode] ||
+       stream_.nUserChannels[0] != stream_.nDeviceChannels[0] ||
+       stream_.nUserChannels[1] != stream_.nDeviceChannels[1] )
+    stream_.doConvertBuffer[mode] = true;
+  else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+            stream_.nUserChannels[mode] > 1 )
+    stream_.doConvertBuffer[mode] = true;
+
+  if ( stream_.doConvertBuffer[mode] )
+    setConvertInfo( mode, firstChannel );
+
+  // Allocate necessary internal buffers
+  bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
+
+  stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
+  if ( !stream_.userBuffer[mode] ) {
+    errorType = RtAudioError::MEMORY_ERROR;
+    errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
+    goto Exit;
+  }
+
+  if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
+    stream_.callbackInfo.priority = 15;
+  else
+    stream_.callbackInfo.priority = 0;
+
+  ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
+  ///! TODO: RTAUDIO_HOG_DEVICE       // Exclusive mode
+
+  methodResult = SUCCESS;
+
+Exit:
+  //clean up
+  SAFE_RELEASE( captureDevices );
+  SAFE_RELEASE( renderDevices );
+  SAFE_RELEASE( devicePtr );
+  CoTaskMemFree( deviceFormat );
+
+  // if method failed, close the stream
+  if ( methodResult == FAILURE )
+    closeStream();
+
+  if ( !errorText_.empty() )
+    error( errorType );
+  return methodResult;
+}
+
+//=============================================================================
+
+DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
+{
+  if ( wasapiPtr )
+    ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
+
+  return 0;
+}
+
+DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
+{
+  if ( wasapiPtr )
+    ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
+
+  return 0;
+}
+
+DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
+{
+  if ( wasapiPtr )
+    ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
+
+  return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::wasapiThread()
+{
+  // as this is a new thread, we must CoInitialize it
+  CoInitialize( NULL );
+
+  HRESULT hr;
+
+  IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
+  IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
+  IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
+  IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
+  HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
+  HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
+
+  WAVEFORMATEX* captureFormat = NULL;
+  WAVEFORMATEX* renderFormat = NULL;
+  float captureSrRatio = 0.0f;
+  float renderSrRatio = 0.0f;
+  WasapiBuffer captureBuffer;
+  WasapiBuffer renderBuffer;
+  WasapiResampler* captureResampler = NULL;
+  WasapiResampler* renderResampler = NULL;
+
+  // declare local stream variables
+  RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
+  BYTE* streamBuffer = NULL;
+  DWORD captureFlags = 0;
+  unsigned int bufferFrameCount = 0;
+  unsigned int numFramesPadding = 0;
+  unsigned int convBufferSize = 0;
+  bool loopbackEnabled = stream_.device[INPUT] == stream_.device[OUTPUT];
+  bool callbackPushed = true;
+  bool callbackPulled = false;
+  bool callbackStopped = false;
+  int callbackResult = 0;
+
+  // convBuffer is used to store converted buffers between WASAPI and the user
+  char* convBuffer = NULL;
+  unsigned int convBuffSize = 0;
+  unsigned int deviceBuffSize = 0;
+
+  std::string errorText;
+  RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
+
+  // Attempt to assign "Pro Audio" characteristic to thread
+  HMODULE AvrtDll = LoadLibraryW( L"AVRT.dll" );
+  if ( AvrtDll ) {
+    DWORD taskIndex = 0;
+    TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr =
+      ( TAvSetMmThreadCharacteristicsPtr ) (void(*)()) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
+    AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
+    FreeLibrary( AvrtDll );
+  }
+
+  // start capture stream if applicable
+  if ( captureAudioClient ) {
+    hr = captureAudioClient->GetMixFormat( &captureFormat );
+    if ( FAILED( hr ) ) {
+      errorText = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
+      goto Exit;
+    }
+
+    // init captureResampler
+    captureResampler = new WasapiResampler( stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT64,
+                                            formatBytes( stream_.deviceFormat[INPUT] ) * 8, stream_.nDeviceChannels[INPUT],
+                                            captureFormat->nSamplesPerSec, stream_.sampleRate );
+
+    captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
+
+    if ( !captureClient ) {
+      hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
+                                           loopbackEnabled ? AUDCLNT_STREAMFLAGS_LOOPBACK : AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+                                           0,
+                                           0,
+                                           captureFormat,
+                                           NULL );
+      if ( FAILED( hr ) ) {
+        errorText = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
+        goto Exit;
+      }
+
+      hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
+                                           ( void** ) &captureClient );
+      if ( FAILED( hr ) ) {
+        errorText = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
+        goto Exit;
+      }
+
+      // don't configure captureEvent if in loopback mode
+      if ( !loopbackEnabled )
+      {
+        // configure captureEvent to trigger on every available capture buffer
+        captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
+        if ( !captureEvent ) {
+          errorType = RtAudioError::SYSTEM_ERROR;
+          errorText = "RtApiWasapi::wasapiThread: Unable to create capture event.";
+          goto Exit;
+        }
+
+        hr = captureAudioClient->SetEventHandle( captureEvent );
+        if ( FAILED( hr ) ) {
+          errorText = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
+          goto Exit;
+        }
+
+        ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
+      }
+
+      ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
+
+      // reset the capture stream
+      hr = captureAudioClient->Reset();
+      if ( FAILED( hr ) ) {
+        errorText = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
+        goto Exit;
+      }
+
+      // start the capture stream
+      hr = captureAudioClient->Start();
+      if ( FAILED( hr ) ) {
+        errorText = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
+        goto Exit;
+      }
+    }
+
+    unsigned int inBufferSize = 0;
+    hr = captureAudioClient->GetBufferSize( &inBufferSize );
+    if ( FAILED( hr ) ) {
+      errorText = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
+      goto Exit;
+    }
+
+    // scale outBufferSize according to stream->user sample rate ratio
+    unsigned int outBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
+    inBufferSize *= stream_.nDeviceChannels[INPUT];
+
+    // set captureBuffer size
+    captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
+  }
+
+  // start render stream if applicable
+  if ( renderAudioClient ) {
+    hr = renderAudioClient->GetMixFormat( &renderFormat );
+    if ( FAILED( hr ) ) {
+      errorText = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
+      goto Exit;
+    }
+
+    // init renderResampler
+    renderResampler = new WasapiResampler( stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT64,
+                                           formatBytes( stream_.deviceFormat[OUTPUT] ) * 8, stream_.nDeviceChannels[OUTPUT],
+                                           stream_.sampleRate, renderFormat->nSamplesPerSec );
+
+    renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
+
+    if ( !renderClient ) {
+      hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
+                                          AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+                                          0,
+                                          0,
+                                          renderFormat,
+                                          NULL );
+      if ( FAILED( hr ) ) {
+        errorText = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
+        goto Exit;
+      }
+
+      hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
+                                          ( void** ) &renderClient );
+      if ( FAILED( hr ) ) {
+        errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
+        goto Exit;
+      }
+
+      // configure renderEvent to trigger on every available render buffer
+      renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
+      if ( !renderEvent ) {
+        errorType = RtAudioError::SYSTEM_ERROR;
+        errorText = "RtApiWasapi::wasapiThread: Unable to create render event.";
+        goto Exit;
+      }
+
+      hr = renderAudioClient->SetEventHandle( renderEvent );
+      if ( FAILED( hr ) ) {
+        errorText = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
+        goto Exit;
+      }
+
+      ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
+      ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
+
+      // reset the render stream
+      hr = renderAudioClient->Reset();
+      if ( FAILED( hr ) ) {
+        errorText = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
+        goto Exit;
+      }
+
+      // start the render stream
+      hr = renderAudioClient->Start();
+      if ( FAILED( hr ) ) {
+        errorText = "RtApiWasapi::wasapiThread: Unable to start render stream.";
+        goto Exit;
+      }
+    }
+
+    unsigned int outBufferSize = 0;
+    hr = renderAudioClient->GetBufferSize( &outBufferSize );
+    if ( FAILED( hr ) ) {
+      errorText = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
+      goto Exit;
+    }
+
+    // scale inBufferSize according to user->stream sample rate ratio
+    unsigned int inBufferSize = ( unsigned int ) ceilf( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
+    outBufferSize *= stream_.nDeviceChannels[OUTPUT];
+
+    // set renderBuffer size
+    renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
+  }
+
+  // malloc buffer memory
+  if ( stream_.mode == INPUT )
+  {
+    using namespace std; // for ceilf
+    convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
+    deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
+  }
+  else if ( stream_.mode == OUTPUT )
+  {
+    convBuffSize = ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
+    deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
+  }
+  else if ( stream_.mode == DUPLEX )
+  {
+    convBuffSize = std::max( ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
+                             ( size_t ) ( ceilf( stream_.bufferSize * renderSrRatio ) ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
+    deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
+                               stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
+  }
+
+  convBuffSize *= 2; // allow overflow for *SrRatio remainders
+  convBuffer = ( char* ) calloc( convBuffSize, 1 );
+  stream_.deviceBuffer = ( char* ) calloc( deviceBuffSize, 1 );
+  if ( !convBuffer || !stream_.deviceBuffer ) {
+    errorType = RtAudioError::MEMORY_ERROR;
+    errorText = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
+    goto Exit;
+  }
+
+  // stream process loop
+  while ( stream_.state != STREAM_STOPPING ) {
+    if ( !callbackPulled ) {
+      // Callback Input
+      // ==============
+      // 1. Pull callback buffer from inputBuffer
+      // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
+      //                          Convert callback buffer to user format
+
+      if ( captureAudioClient )
+      {
+        int samplesToPull = ( unsigned int ) floorf( stream_.bufferSize * captureSrRatio );
+
+        convBufferSize = 0;
+        while ( convBufferSize < stream_.bufferSize )
+        {
+          // Pull callback buffer from inputBuffer
+          callbackPulled = captureBuffer.pullBuffer( convBuffer,
+                                                     samplesToPull * stream_.nDeviceChannels[INPUT],
+                                                     stream_.deviceFormat[INPUT] );
+
+          if ( !callbackPulled )
+          {
+            break;
+          }
+
+          // Convert callback buffer to user sample rate
+          unsigned int deviceBufferOffset = convBufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
+          unsigned int convSamples = 0;
+
+          captureResampler->Convert( stream_.deviceBuffer + deviceBufferOffset,
+                                     convBuffer,
+                                     samplesToPull,
+                                     convSamples,
+                                     convBufferSize == 0 ? -1 : stream_.bufferSize - convBufferSize );
+
+          convBufferSize += convSamples;
+          samplesToPull = 1; // now pull one sample at a time until we have stream_.bufferSize samples
+        }
+
+        if ( callbackPulled )
+        {
+          if ( stream_.doConvertBuffer[INPUT] ) {
+            // Convert callback buffer to user format
+            convertBuffer( stream_.userBuffer[INPUT],
+                           stream_.deviceBuffer,
+                           stream_.convertInfo[INPUT] );
+          }
+          else {
+            // no further conversion, simple copy deviceBuffer to userBuffer
+            memcpy( stream_.userBuffer[INPUT],
+                    stream_.deviceBuffer,
+                    stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
+          }
+        }
+      }
+      else {
+        // if there is no capture stream, set callbackPulled flag
+        callbackPulled = true;
+      }
+
+      // Execute Callback
+      // ================
+      // 1. Execute user callback method
+      // 2. Handle return value from callback
+
+      // if callback has not requested the stream to stop
+      if ( callbackPulled && !callbackStopped ) {
+        // Execute user callback method
+        callbackResult = callback( stream_.userBuffer[OUTPUT],
+                                   stream_.userBuffer[INPUT],
+                                   stream_.bufferSize,
+                                   getStreamTime(),
+                                   captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
+                                   stream_.callbackInfo.userData );
+
+        // tick stream time
+        RtApi::tickStreamTime();
+
+        // Handle return value from callback
+        if ( callbackResult == 1 ) {
+          // instantiate a thread to stop this thread
+          HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
+          if ( !threadHandle ) {
+            errorType = RtAudioError::THREAD_ERROR;
+            errorText = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
+            goto Exit;
+          }
+          else if ( !CloseHandle( threadHandle ) ) {
+            errorType = RtAudioError::THREAD_ERROR;
+            errorText = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
+            goto Exit;
+          }
+
+          callbackStopped = true;
+        }
+        else if ( callbackResult == 2 ) {
+          // instantiate a thread to stop this thread
+          HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
+          if ( !threadHandle ) {
+            errorType = RtAudioError::THREAD_ERROR;
+            errorText = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
+            goto Exit;
+          }
+          else if ( !CloseHandle( threadHandle ) ) {
+            errorType = RtAudioError::THREAD_ERROR;
+            errorText = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
+            goto Exit;
+          }
+
+          callbackStopped = true;
+        }
+      }
+    }
+
+    // Callback Output
+    // ===============
+    // 1. Convert callback buffer to stream format
+    // 2. Convert callback buffer to stream sample rate and channel count
+    // 3. Push callback buffer into outputBuffer
+
+    if ( renderAudioClient && callbackPulled )
+    {
+      // if the last call to renderBuffer.PushBuffer() was successful
+      if ( callbackPushed || convBufferSize == 0 )
+      {
+        if ( stream_.doConvertBuffer[OUTPUT] )
+        {
+          // Convert callback buffer to stream format
+          convertBuffer( stream_.deviceBuffer,
+                         stream_.userBuffer[OUTPUT],
+                         stream_.convertInfo[OUTPUT] );
+
+        }
+        else {
+          // no further conversion, simple copy userBuffer to deviceBuffer
+          memcpy( stream_.deviceBuffer,
+                  stream_.userBuffer[OUTPUT],
+                  stream_.bufferSize * stream_.nUserChannels[OUTPUT] * formatBytes( stream_.userFormat ) );
+        }
+
+        // Convert callback buffer to stream sample rate
+        renderResampler->Convert( convBuffer,
+                                  stream_.deviceBuffer,
+                                  stream_.bufferSize,
+                                  convBufferSize );
+      }
+
+      // Push callback buffer into outputBuffer
+      callbackPushed = renderBuffer.pushBuffer( convBuffer,
+                                                convBufferSize * stream_.nDeviceChannels[OUTPUT],
+                                                stream_.deviceFormat[OUTPUT] );
+    }
+    else {
+      // if there is no render stream, set callbackPushed flag
+      callbackPushed = true;
+    }
+
+    // Stream Capture
+    // ==============
+    // 1. Get capture buffer from stream
+    // 2. Push capture buffer into inputBuffer
+    // 3. If 2. was successful: Release capture buffer
+
+    if ( captureAudioClient ) {
+      // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
+      if ( !callbackPulled ) {
+        WaitForSingleObject( loopbackEnabled ? renderEvent : captureEvent, INFINITE );
+      }
+
+      // Get capture buffer from stream
+      hr = captureClient->GetBuffer( &streamBuffer,
+                                     &bufferFrameCount,
+                                     &captureFlags, NULL, NULL );
+      if ( FAILED( hr ) ) {
+        errorText = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
+        goto Exit;
+      }
+
+      if ( bufferFrameCount != 0 ) {
+        // Push capture buffer into inputBuffer
+        if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
+                                       bufferFrameCount * stream_.nDeviceChannels[INPUT],
+                                       stream_.deviceFormat[INPUT] ) )
+        {
+          // Release capture buffer
+          hr = captureClient->ReleaseBuffer( bufferFrameCount );
+          if ( FAILED( hr ) ) {
+            errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
+            goto Exit;
+          }
+        }
+        else
+        {
+          // Inform WASAPI that capture was unsuccessful
+          hr = captureClient->ReleaseBuffer( 0 );
+          if ( FAILED( hr ) ) {
+            errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
+            goto Exit;
+          }
+        }
+      }
+      else
+      {
+        // Inform WASAPI that capture was unsuccessful
+        hr = captureClient->ReleaseBuffer( 0 );
+        if ( FAILED( hr ) ) {
+          errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
+          goto Exit;
+        }
+      }
+    }
+
+    // Stream Render
+    // =============
+    // 1. Get render buffer from stream
+    // 2. Pull next buffer from outputBuffer
+    // 3. If 2. was successful: Fill render buffer with next buffer
+    //                          Release render buffer
+
+    if ( renderAudioClient ) {
+      // if the callback output buffer was not pushed to renderBuffer, wait for next render event
+      if ( callbackPulled && !callbackPushed ) {
+        WaitForSingleObject( renderEvent, INFINITE );
+      }
+
+      // Get render buffer from stream
+      hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
+      if ( FAILED( hr ) ) {
+        errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
+        goto Exit;
+      }
+
+      hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
+      if ( FAILED( hr ) ) {
+        errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
+        goto Exit;
+      }
+
+      bufferFrameCount -= numFramesPadding;
+
+      if ( bufferFrameCount != 0 ) {
+        hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
+        if ( FAILED( hr ) ) {
+          errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
+          goto Exit;
+        }
+
+        // Pull next buffer from outputBuffer
+        // Fill render buffer with next buffer
+        if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
+                                      bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
+                                      stream_.deviceFormat[OUTPUT] ) )
+        {
+          // Release render buffer
+          hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
+          if ( FAILED( hr ) ) {
+            errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
+            goto Exit;
+          }
+        }
+        else
+        {
+          // Inform WASAPI that render was unsuccessful
+          hr = renderClient->ReleaseBuffer( 0, 0 );
+          if ( FAILED( hr ) ) {
+            errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
+            goto Exit;
+          }
+        }
+      }
+      else
+      {
+        // Inform WASAPI that render was unsuccessful
+        hr = renderClient->ReleaseBuffer( 0, 0 );
+        if ( FAILED( hr ) ) {
+          errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
+          goto Exit;
+        }
+      }
+    }
+
+    // if the callback buffer was pushed renderBuffer reset callbackPulled flag
+    if ( callbackPushed ) {
+      // unsetting the callbackPulled flag lets the stream know that
+      // the audio device is ready for another callback output buffer.
+      callbackPulled = false;
+    }
+
+  }
+
+Exit:
+  // clean up
+  CoTaskMemFree( captureFormat );
+  CoTaskMemFree( renderFormat );
+
+  free ( convBuffer );
+  delete renderResampler;
+  delete captureResampler;
+
+  CoUninitialize();
+
+  // update stream state
+  stream_.state = STREAM_STOPPED;
+
+  if ( !errorText.empty() )
+  {
+    errorText_ = errorText;
+    error( errorType );
+  }
+}
+
+//******************** End of __WINDOWS_WASAPI__ *********************//
+#endif
+
+
+#if defined(__WINDOWS_DS__) // Windows DirectSound API
+
+// Modified by Robin Davies, October 2005
+// - Improvements to DirectX pointer chasing. 
+// - 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
+// Changed device query structure for RtAudio 4.0.7, January 2010
+
+#include <windows.h>
+#include <process.h>
+#include <mmsystem.h>
+#include <mmreg.h>
+#include <dsound.h>
+#include <assert.h>
+#include <algorithm>
+
+#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 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; }
+};
+
+// 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 const char* getErrorString( int code );
+
+static unsigned __stdcall callbackHandler( void *ptr );
+
+struct DsDevice {
+  LPGUID id[2];
+  bool validId[2];
+  bool found;
+  std::string name;
+
+  DsDevice()
+  : found(false) { validId[0] = false; validId[1] = false; }
+};
+
+struct DsProbeData {
+  bool isInput;
+  std::vector<struct DsDevice>* dsDevices;
+};
+
+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 ( stream_.state != STREAM_CLOSED ) closeStream();
+  if ( coInitialized_ ) CoUninitialize(); // balanced call.
+}
+
+// The DirectSound default output is always the first device.
+unsigned int RtApiDs :: getDefaultOutputDevice( void )
+{
+  return 0;
+}
+
+// The DirectSound default input is always the first input device,
+// which is the first capture device enumerated.
+unsigned int RtApiDs :: getDefaultInputDevice( void )
+{
+  return 0;
+}
+
+unsigned int RtApiDs :: getDeviceCount( void )
+{
+  // Set query flag for previously found devices to false, so that we
+  // can check for any devices that have disappeared.
+  for ( unsigned int i=0; i<dsDevices.size(); i++ )
+    dsDevices[i].found = false;
+
+  // Query DirectSound devices.
+  struct DsProbeData probeInfo;
+  probeInfo.isInput = false;
+  probeInfo.dsDevices = &dsDevices;
+  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
+  if ( FAILED( result ) ) {
+    errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::WARNING );
+  }
+
+  // Query DirectSoundCapture devices.
+  probeInfo.isInput = true;
+  result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
+  if ( FAILED( result ) ) {
+    errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::WARNING );
+  }
+
+  // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
+  for ( unsigned int i=0; i<dsDevices.size(); ) {
+    if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
+    else i++;
+  }
+
+  return static_cast<unsigned int>(dsDevices.size());
+}
+
+RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
+{
+  RtAudio::DeviceInfo info;
+  info.probed = false;
+
+  if ( dsDevices.size() == 0 ) {
+    // Force a query of all devices
+    getDeviceCount();
+    if ( dsDevices.size() == 0 ) {
+      errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
+      error( RtAudioError::INVALID_USE );
+      return info;
+    }
+  }
+
+  if ( device >= dsDevices.size() ) {
+    errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
+    error( RtAudioError::INVALID_USE );
+    return info;
+  }
+
+  HRESULT result;
+  if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
+
+  LPDIRECTSOUND output;
+  DSCAPS outCaps;
+  result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
+  if ( FAILED( result ) ) {
+    errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::WARNING );
+    goto probeInput;
+  }
+
+  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( RtAudioError::WARNING );
+    goto probeInput;
+  }
+
+  // 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] );
+
+      if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
+        info.preferredSampleRate = 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;
+
+  if ( dsDevices[ device ].validId[1] == false ) {
+    info.name = dsDevices[ device ].name;
+    info.probed = true;
+    return info;
+  }
+
+ probeInput:
+
+  LPDIRECTSOUNDCAPTURE input;
+  result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
+  if ( FAILED( result ) ) {
+    errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::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 (" << dsDevices[ device ].name << ")!";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::WARNING );
+    return info;
+  }
+
+  // Get input channel information.
+  info.inputChannels = inCaps.dwChannels;
+
+  // Get sample rate and format information.
+  std::vector<unsigned int> rates;
+  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 ) rates.push_back( 11025 );
+      if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
+      if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
+      if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
+    }
+    else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
+      if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
+      if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
+      if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
+      if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
+    }
+  }
+  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 ) rates.push_back( 11025 );
+      if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
+      if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
+      if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
+    }
+    else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
+      if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
+      if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
+      if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
+      if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
+    }
+  }
+  else info.inputChannels = 0; // technically, this would be an error
+
+  input->Release();
+
+  if ( info.inputChannels == 0 ) return info;
+
+  // Copy the supported rates to the info structure but avoid duplication.
+  bool found;
+  for ( unsigned int i=0; i<rates.size(); i++ ) {
+    found = false;
+    for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
+      if ( rates[i] == info.sampleRates[j] ) {
+        found = true;
+        break;
+      }
+    }
+    if ( found == false ) info.sampleRates.push_back( rates[i] );
+  }
+  std::sort( info.sampleRates.begin(), info.sampleRates.end() );
+
+  // If device opens for both playback and capture, we determine the channels.
+  if ( info.outputChannels > 0 && info.inputChannels > 0 )
+    info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+
+  if ( device == 0 ) info.isDefaultInput = true;
+
+  // Copy name and return.
+  info.name = dsDevices[ device ].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;
+  }
+
+  size_t nDevices = dsDevices.size();
+  if ( nDevices == 0 ) {
+    // This should not happen because a check is made before this function is called.
+    errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
+    return FAILURE;
+  }
+
+  if ( device >= nDevices ) {
+    // This should not happen because a check is made before this function is called.
+    errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
+    return FAILURE;
+  }
+
+  if ( mode == OUTPUT ) {
+    if ( dsDevices[ device ].validId[0] == false ) {
+      errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+  }
+  else { // mode == INPUT
+    if ( dsDevices[ device ].validId[1] == false ) {
+      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.  In the past, I had
+  // problems when using GetDesktopWindow() but it seems fine now
+  // (January 2010).  I'll leave it commented here.
+  // HWND hWnd = GetForegroundWindow();
+  HWND hWnd = GetDesktopWindow();
+
+  // 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;
+
+  // Check the lower range of the user-specified buffer size and set
+  // (arbitrarily) to a lower bound of 32.
+  if ( *bufferSize < 32 ) *bufferSize = 32;
+
+  // 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, we'll use the value
+  // defined above (32K), but we will grow it to make allowances for
+  // very large software buffer sizes.
+  DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
+  DWORD dsPointerLeadTime = 0;
+
+  void *ohandle = 0, *bhandle = 0;
+  HRESULT result;
+  if ( mode == OUTPUT ) {
+
+    LPDIRECTSOUND output;
+    result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
+    if ( FAILED( result ) ) {
+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].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 (" << dsDevices[ device ].name << ")!";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+
+    // Check channel information.
+    if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
+      errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].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 > dsBufferSize )
+      dsBufferSize *= 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 (" << dsDevices[ device ].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 (" << dsDevices[ device ].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 (" << dsDevices[ device ].name << ")!";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+
+    // Setup the secondary DS buffer description.
+    ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
+    bufferDescription.dwSize = sizeof( DSBUFFERDESC );
+    bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
+                                  DSBCAPS_GLOBALFOCUS |
+                                  DSBCAPS_GETCURRENTPOSITION2 |
+                                  DSBCAPS_LOCHARDWARE );  // Force hardware mixing
+    bufferDescription.dwBufferBytes = dsBufferSize;
+    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 (" << dsDevices[ device ].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 (" << dsDevices[ device ].name << ")!";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+
+    dsBufferSize = dsbcaps.dwBufferBytes;
+
+    // Lock the DS buffer
+    LPVOID audioPtr;
+    DWORD dataLen;
+    result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
+    if ( FAILED( result ) ) {
+      output->Release();
+      buffer->Release();
+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].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 (" << dsDevices[ device ].name << ")!";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+
+    ohandle = (void *) output;
+    bhandle = (void *) buffer;
+  }
+
+  if ( mode == INPUT ) {
+
+    LPDIRECTSOUNDCAPTURE input;
+    result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
+    if ( FAILED( result ) ) {
+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].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 (" << dsDevices[ device ].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 > dsBufferSize )
+      dsBufferSize *= 2;
+
+    // Setup the secondary DS buffer description.
+    DSCBUFFERDESC bufferDescription;
+    ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
+    bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
+    bufferDescription.dwFlags = 0;
+    bufferDescription.dwReserved = 0;
+    bufferDescription.dwBufferBytes = dsBufferSize;
+    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 (" << dsDevices[ device ].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 (" << dsDevices[ device ].name << ")!";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+
+    dsBufferSize = dscbcaps.dwBufferBytes;
+
+    // NOTE: We could have a problem here if this is a duplex stream
+    // and the play and capture hardware buffer sizes are different
+    // (I'm actually not sure if that is a problem or not).
+    // Currently, we are not verifying that.
+
+    // Lock the capture buffer
+    LPVOID audioPtr;
+    DWORD dataLen;
+    result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
+    if ( FAILED( result ) ) {
+      input->Release();
+      buffer->Release();
+      errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].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 (" << dsDevices[ device ].name << ")!";
+      errorText_ = errorStream_.str();
+      return FAILURE;
+    }
+
+    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
+  long 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.
+  if ( stream_.callbackInfo.isRunning == false ) {
+    unsigned threadId;
+    stream_.callbackInfo.isRunning = true;
+    stream_.callbackInfo.object = (void *) this;
+    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;
+  }
+
+  stream_.state = STREAM_CLOSED;
+  return FAILURE;
+}
+
+void RtApiDs :: closeStream()
+{
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiDs::closeStream(): no open stream to close!";
+    error( RtAudioError::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( RtAudioError::WARNING );
+    return;
+  }
+
+  #if defined( HAVE_GETTIMEOFDAY )
+  gettimeofday( &stream_.lastTickTimestamp, NULL );
+  #endif
+
+  DsHandle *handle = (DsHandle *) stream_.apiHandle;
+
+  // Increase scheduler frequency on lesser windows (a side-effect of
+  // increasing timer accuracy).  On greater windows (Win2K or later),
+  // this is already in effect.
+  timeBeginPeriod( 1 ); 
+
+  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 ) {
+
+    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 ) {
+
+    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;
+  ResetEvent( handle->condition );
+  stream_.state = STREAM_RUNNING;
+
+ unlock:
+  if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiDs :: stopStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
+    error( RtAudioError::WARNING );
+    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 = 2;
+      WaitForSingleObject( handle->condition, INFINITE );  // block until signaled
+    }
+
+    stream_.state = STREAM_STOPPED;
+
+    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;
+
+    stream_.state = STREAM_STOPPED;
+
+    if ( stream_.mode != DUPLEX )
+      MUTEX_LOCK( &stream_.mutex );
+
+    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.
+  MUTEX_UNLOCK( &stream_.mutex );
+
+  if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiDs :: abortStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  DsHandle *handle = (DsHandle *) stream_.apiHandle;
+  handle->drainCounter = 2;
+
+  stopStream();
+}
+
+void RtApiDs :: callbackEvent()
+{
+  if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
+    Sleep( 50 ); // sleep 50 milliseconds
+    return;
+  }
+
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
+    error( RtAudioError::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 ) {
+
+    stream_.state = STREAM_STOPPING;
+    if ( handle->internalDrain == false )
+      SetEvent( handle->condition );
+    else
+      stopStream();
+    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;
+    }
+    int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+                                  stream_.bufferSize, streamTime, status, info->userData );
+    if ( cbReturnValue == 2 ) {
+      stream_.state = STREAM_STOPPING;
+      handle->drainCounter = 2;
+      abortStream();
+      return;
+    }
+    else if ( cbReturnValue == 1 ) {
+      handle->drainCounter = 1;
+      handle->internalDrain = true;
+    }
+  }
+
+  HRESULT result;
+  DWORD currentWritePointer, safeWritePointer;
+  DWORD currentReadPointer, safeReadPointer;
+  UINT nextWritePointer;
+
+  LPVOID buffer1 = NULL;
+  LPVOID buffer2 = NULL;
+  DWORD bufferSize1 = 0;
+  DWORD bufferSize2 = 0;
+
+  char *buffer;
+  long bufferBytes;
+
+  MUTEX_LOCK( &stream_.mutex );
+  if ( stream_.state == STREAM_STOPPED ) {
+    MUTEX_UNLOCK( &stream_.mutex );
+    return;
+  }
+
+  if ( buffersRolling == false ) {
+    if ( stream_.mode == DUPLEX ) {
+      //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 startSafeWritePointer, startSafeReadPointer;
+
+      result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+        errorText_ = errorStream_.str();
+        MUTEX_UNLOCK( &stream_.mutex );
+        error( RtAudioError::SYSTEM_ERROR );
+        return;
+      }
+      result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+        errorText_ = errorStream_.str();
+        MUTEX_UNLOCK( &stream_.mutex );
+        error( RtAudioError::SYSTEM_ERROR );
+        return;
+      }
+      while ( true ) {
+        result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
+        if ( FAILED( result ) ) {
+          errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+          errorText_ = errorStream_.str();
+          MUTEX_UNLOCK( &stream_.mutex );
+          error( RtAudioError::SYSTEM_ERROR );
+          return;
+        }
+        result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
+        if ( FAILED( result ) ) {
+          errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+          errorText_ = errorStream_.str();
+          MUTEX_UNLOCK( &stream_.mutex );
+          error( RtAudioError::SYSTEM_ERROR );
+          return;
+        }
+        if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
+        Sleep( 1 );
+      }
+
+      //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
+
+      handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
+      if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
+      handle->bufferPointer[1] = safeReadPointer;
+    }
+    else if ( stream_.mode == OUTPUT ) {
+
+      // Set the proper nextWritePosition after initial startup.
+      LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+      result = dsWriteBuffer->GetCurrentPosition( &currentWritePointer, &safeWritePointer );
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+        errorText_ = errorStream_.str();
+        MUTEX_UNLOCK( &stream_.mutex );
+        error( RtAudioError::SYSTEM_ERROR );
+        return;
+      }
+      handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
+      if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
+    }
+
+    buffersRolling = true;
+  }
+
+  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];
+    nextWritePointer = handle->bufferPointer[0];
+
+    DWORD endWrite, leadPointer;
+    while ( true ) {
+      // Find out where the read and "safe write" pointers are.
+      result = dsBuffer->GetCurrentPosition( &currentWritePointer, &safeWritePointer );
+      if ( FAILED( result ) ) {
+        errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+        errorText_ = errorStream_.str();
+        MUTEX_UNLOCK( &stream_.mutex );
+        error( RtAudioError::SYSTEM_ERROR );
+        return;
+      }
+
+      // We will copy our output buffer into the region between
+      // safeWritePointer and leadPointer.  If leadPointer is not
+      // beyond the next endWrite position, wait until it is.
+      leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
+      //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
+      if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
+      if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
+      endWrite = nextWritePointer + bufferBytes;
+
+      // Check whether the entire write region is behind the play pointer.
+      if ( leadPointer >= endWrite ) break;
+
+      // If we are here, then we must wait until the leadPointer advances
+      // beyond the end of our next write region. We use the
+      // Sleep() function to suspend operation until that happens.
+      double millis = ( endWrite - leadPointer ) * 1000.0;
+      millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
+      if ( millis < 1.0 ) millis = 1.0;
+      Sleep( (DWORD) millis );
+    }
+
+    if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
+         || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) { 
+      // We've strayed into the forbidden zone ... resync the read pointer.
+      handle->xrun[0] = true;
+      nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
+      if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
+      handle->bufferPointer[0] = nextWritePointer;
+      endWrite = nextWritePointer + bufferBytes;
+    }
+
+    // Lock free space in the buffer
+    result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
+                             &bufferSize1, &buffer2, &bufferSize2, 0 );
+    if ( FAILED( result ) ) {
+      errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
+      errorText_ = errorStream_.str();
+      MUTEX_UNLOCK( &stream_.mutex );
+      error( RtAudioError::SYSTEM_ERROR );
+      return;
+    }
+
+    // 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();
+      MUTEX_UNLOCK( &stream_.mutex );
+      error( RtAudioError::SYSTEM_ERROR );
+      return;
+    }
+    nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
+    handle->bufferPointer[0] = nextWritePointer;
+  }
+
+  // Don't bother draining input
+  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 nextReadPointer = handle->bufferPointer[1];
+    DWORD dsBufferSize = handle->dsBufferSize[1];
+
+    // Find out where the write and "safe read" pointers are.
+    result = dsBuffer->GetCurrentPosition( &currentReadPointer, &safeReadPointer );
+    if ( FAILED( result ) ) {
+      errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+      errorText_ = errorStream_.str();
+      MUTEX_UNLOCK( &stream_.mutex );
+      error( RtAudioError::SYSTEM_ERROR );
+      return;
+    }
+
+    if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
+    DWORD endRead = nextReadPointer + 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 ( safeReadPointer < endRead ) {
+        if ( duplexPrerollBytes <= 0 ) {
+          // Pre-roll time over. Be more agressive.
+          int adjustment = endRead-safeReadPointer;
+
+          handle->xrun[1] = true;
+          // 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 )
+            nextReadPointer = safeReadPointer-2*bufferBytes;
+          else
+            nextReadPointer = safeReadPointer-bufferBytes-adjustment;
+
+          if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
+
+        }
+        else {
+          // In pre=roll time. Just do it.
+          nextReadPointer = safeReadPointer - bufferBytes;
+          while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
+        }
+        endRead = nextReadPointer + bufferBytes;
+      }
+    }
+    else { // mode == INPUT
+      while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
+        // See comments for playback.
+        double millis = (endRead - safeReadPointer) * 1000.0;
+        millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
+        if ( millis < 1.0 ) millis = 1.0;
+        Sleep( (DWORD) millis );
+
+        // Wake up and find out where we are now.
+        result = dsBuffer->GetCurrentPosition( &currentReadPointer, &safeReadPointer );
+        if ( FAILED( result ) ) {
+          errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+          errorText_ = errorStream_.str();
+          MUTEX_UNLOCK( &stream_.mutex );
+          error( RtAudioError::SYSTEM_ERROR );
+          return;
+        }
+      
+        if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
+      }
+    }
+
+    // Lock free space in the buffer
+    result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
+                             &bufferSize1, &buffer2, &bufferSize2, 0 );
+    if ( FAILED( result ) ) {
+      errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
+      errorText_ = errorStream_.str();
+      MUTEX_UNLOCK( &stream_.mutex );
+      error( RtAudioError::SYSTEM_ERROR );
+      return;
+    }
+
+    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
+    nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
+    dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
+    if ( FAILED( result ) ) {
+      errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
+      errorText_ = errorStream_.str();
+      MUTEX_UNLOCK( &stream_.mutex );
+      error( RtAudioError::SYSTEM_ERROR );
+      return;
+    }
+    handle->bufferPointer[1] = nextReadPointer;
+
+    // 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] );
+  }
+
+ unlock:
+  MUTEX_UNLOCK( &stream_.mutex );
+  RtApi::tickStreamTime();
+}
+
+// Definitions for utility functions and callbacks
+// specific to the DirectSound implementation.
+
+static 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;
+}
+
+static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
+                                          LPCTSTR description,
+                                          LPCTSTR /*module*/,
+                                          LPVOID lpContext )
+{
+  struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
+  std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
+
+  HRESULT hr;
+  bool validDevice = false;
+  if ( probeInfo.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 )
+        validDevice = true;
+    }
+    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 )
+        validDevice = true;
+    }
+    object->Release();
+  }
+
+  // If good device, then save its name and guid.
+  std::string name = convertCharPointerToStdString( description );
+  //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
+  if ( lpguid == NULL )
+    name = "Default Device";
+  if ( validDevice ) {
+    for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
+      if ( dsDevices[i].name == name ) {
+        dsDevices[i].found = true;
+        if ( probeInfo.isInput ) {
+          dsDevices[i].id[1] = lpguid;
+          dsDevices[i].validId[1] = true;
+        }
+        else {
+          dsDevices[i].id[0] = lpguid;
+          dsDevices[i].validId[0] = true;
+        }
+        return TRUE;
+      }
+    }
+
+    DsDevice device;
+    device.name = name;
+    device.found = true;
+    if ( probeInfo.isInput ) {
+      device.id[1] = lpguid;
+      device.validId[1] = true;
+    }
+    else {
+      device.id[0] = lpguid;
+      device.validId[0] = true;
+    }
+    dsDevices.push_back( device );
+  }
+
+  return TRUE;
+}
+
+static const 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_cv;
+  bool runnable;
+
+  AlsaHandle()
+#if _cplusplus >= 201103L
+    :handles{nullptr, nullptr}, synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
+#else 
+    : synchronized(false), runnable(false) { handles[0] = NULL; handles[1] = NULL; xrun[0] = false; xrun[1] = false; }
+#endif
+};
+
+static 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 = 0;
+
+  strcpy(name, "default");
+  result = snd_ctl_open( &handle, "default", 0 );
+  if (result == 0) {
+    nDevices++;
+    snd_ctl_close( 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 ) {
+      handle = 0;
+      errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      error( RtAudioError::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( RtAudioError::WARNING );
+        break;
+      }
+      if ( subdevice < 0 )
+        break;
+      nDevices++;
+    }
+  nextcard:
+    if ( handle )
+        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=-1, subdevice=-1, card=-1;
+  char name[64];
+  snd_ctl_t *chandle = 0;
+
+  result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
+  if ( result == 0 ) {
+    if ( nDevices++ == device ) {
+      strcpy( name, "default" );
+      goto foundDevice;
+    }
+  }
+  if ( chandle )
+    snd_ctl_close( chandle );
+
+  // Count cards and devices
+  snd_card_next( &card );
+  while ( card >= 0 ) {
+    sprintf( name, "hw:%d", card );
+    result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
+    if ( result < 0 ) {
+      chandle = 0;
+      errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      error( RtAudioError::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( RtAudioError::WARNING );
+        break;
+      }
+      if ( subdevice < 0 ) break;
+      if ( nDevices == device ) {
+        sprintf( name, "hw:%d,%d", card, subdevice );
+        goto foundDevice;
+      }
+      nDevices++;
+    }
+  nextcard:
+    if ( chandle )
+        snd_ctl_close( chandle );
+    snd_card_next( &card );
+  }
+
+  if ( nDevices == 0 ) {
+    errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
+    error( RtAudioError::INVALID_USE );
+    return info;
+  }
+
+  if ( device >= nDevices ) {
+    errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
+    error( RtAudioError::INVALID_USE );
+    return info;
+  }
+
+ 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 ) ) {
+    snd_ctl_close( chandle );
+    if ( device >= devices_.size() ) {
+      errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
+      error( RtAudioError::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 unless default device (which has subdev -1)
+  stream = SND_PCM_STREAM_PLAYBACK;
+  snd_pcm_info_set_stream( pcminfo, stream );
+  if ( subdevice != -1 ) {
+    snd_pcm_info_set_device( pcminfo, subdevice );
+    snd_pcm_info_set_subdevice( pcminfo, 0 );
+
+    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( RtAudioError::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( RtAudioError::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( RtAudioError::WARNING );
+    goto captureProbe;
+  }
+  info.outputChannels = value;
+  snd_pcm_close( phandle );
+
+ captureProbe:
+  stream = SND_PCM_STREAM_CAPTURE;
+  snd_pcm_info_set_stream( pcminfo, stream );
+
+  // Now try for capture unless default device (with subdev = -1)
+  if ( subdevice != -1 ) {
+    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;
+    }
+  }
+  else
+    snd_ctl_close( chandle );
+
+  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( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::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.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
+        info.preferredSampleRate = 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( RtAudioError::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 ) {
+    snd_pcm_close( phandle );
+    errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::WARNING );
+    return info;
+  }
+
+  // Get the device name
+  if (strncmp(name, "default", 7)!=0) {
+    char *cardname;
+    result = snd_card_get_name( card, &cardname );
+    if ( result >= 0 ) {
+      sprintf( name, "hw:%s,%d", cardname, subdevice );
+      free( cardname );
+    }
+  }
+  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__)
+  struct SndOutputTdealloc {
+    SndOutputTdealloc() : _out(NULL) { snd_output_stdio_attach(&_out, stderr, 0); }
+    ~SndOutputTdealloc() { snd_output_close(_out); }
+    operator snd_output_t*() { return _out; }
+    snd_output_t *_out;
+  } out;
+#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;
+
+  if ( device == 0
+       || (options && options->flags & RTAUDIO_ALSA_USE_DEFAULT) )
+  {
+    strcpy(name, "default");
+    result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
+    if ( result == 0 ) {
+      if ( nDevices == device ) {
+        strcpy( name, "default" );
+        snd_ctl_close( chandle );
+        goto foundDevice;
+      }
+      nDevices++;
+    }
+  }
+
+  else {
+    nDevices++;
+    // 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.
+  snd_pcm_close( phandle );
+  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 (or period) size.
+  int dir = 0;
+  snd_pcm_uframes_t periodSize = *bufferSize;
+  result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
+  if ( result < 0 ) {
+    snd_pcm_close( phandle );
+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
+    errorText_ = errorStream_.str();
+    return FAILURE;
+  }
+  *bufferSize = periodSize;
+
+  // Set the buffer number, which in ALSA is referred to as the "period".
+  unsigned int periods = 0;
+  if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
+  if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
+  if ( periods < 2 ) periods = 4; // a fairly safe default value
+  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 ) {
+    snd_pcm_close( phandle );
+    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_cv, 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;
+  phandle = 0;
+
+  // Allocate necessary internal buffers.
+  unsigned long bufferBytes;
+  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+  if ( stream_.userBuffer[mode] == NULL ) {
+    errorText_ = "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( RtAudioError::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 (e.g. NetBSD 1.6.x with GNU Pthread)
+    if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+      stream_.callbackInfo.doRealtime = true;
+      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;
+
+      // Set the policy BEFORE the priority. Otherwise it fails.
+      pthread_attr_setschedpolicy(&attr, SCHED_RR);
+      pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
+      // This is definitely required. Otherwise it fails.
+      pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
+      pthread_attr_setschedparam(&attr, &param);
+    }
+    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 ) {
+      // Failed. Try instead with default attributes.
+      result = pthread_create( &stream_.callbackInfo.thread, NULL, alsaCallbackHandler, &stream_.callbackInfo );
+      if ( result ) {
+        stream_.callbackInfo.isRunning = false;
+        errorText_ = "RtApiAlsa::error creating callback thread!";
+        goto error;
+      }
+    }
+  }
+
+  return SUCCESS;
+
+ error:
+  if ( apiInfo ) {
+    pthread_cond_destroy( &apiInfo->runnable_cv );
+    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;
+  }
+
+  if ( phandle) snd_pcm_close( phandle );
+
+  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_.state = STREAM_CLOSED;
+  return FAILURE;
+}
+
+void RtApiAlsa :: closeStream()
+{
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+  stream_.callbackInfo.isRunning = false;
+  MUTEX_LOCK( &stream_.mutex );
+  if ( stream_.state == STREAM_STOPPED ) {
+    apiInfo->runnable = true;
+    pthread_cond_signal( &apiInfo->runnable_cv );
+  }
+  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_cv );
+    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( RtAudioError::WARNING );
+    return;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  #if defined( HAVE_GETTIMEOFDAY )
+  gettimeofday( &stream_.lastTickTimestamp, NULL );
+  #endif
+
+  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 ) {
+    result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
+    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:
+  apiInfo->runnable = true;
+  pthread_cond_signal( &apiInfo->runnable_cv );
+  MUTEX_UNLOCK( &stream_.mutex );
+
+  if ( result >= 0 ) return;
+  error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiAlsa :: stopStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  stream_.state = STREAM_STOPPED;
+  MUTEX_LOCK( &stream_.mutex );
+
+  int result = 0;
+  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+  snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+    if ( apiInfo->synchronized ) 
+      result = snd_pcm_drop( handle[0] );
+    else
+      result = snd_pcm_drain( handle[0] );
+    if ( result < 0 ) {
+      errorStream_ << "RtApiAlsa::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:
+  apiInfo->runnable = false; // fixes high CPU usage when stopped
+  MUTEX_UNLOCK( &stream_.mutex );
+
+  if ( result >= 0 ) return;
+  error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiAlsa :: abortStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  stream_.state = STREAM_STOPPED;
+  MUTEX_LOCK( &stream_.mutex );
+
+  int result = 0;
+  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+  snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+    result = snd_pcm_drop( handle[0] );
+    if ( result < 0 ) {
+      errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
+      errorText_ = errorStream_.str();
+      goto unlock;
+    }
+  }
+
+  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:
+  apiInfo->runnable = false; // fixes high CPU usage when stopped
+  MUTEX_UNLOCK( &stream_.mutex );
+
+  if ( result >= 0 ) return;
+  error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiAlsa :: callbackEvent()
+{
+  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+  if ( stream_.state == STREAM_STOPPED ) {
+    MUTEX_LOCK( &stream_.mutex );
+    while ( !apiInfo->runnable )
+      pthread_cond_wait( &apiInfo->runnable_cv, &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( RtAudioError::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( RtAudioError::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
+            errorText_ =  "RtApiAlsa::callbackEvent: audio write error, underrun.";
+        }
+        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( RtAudioError::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();
+}
+
+static void *alsaCallbackHandler( void *ptr )
+{
+  CallbackInfo *info = (CallbackInfo *) ptr;
+  RtApiAlsa *object = (RtApiAlsa *) info->object;
+  bool *isRunning = &info->isRunning;
+
+#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
+  if ( info->doRealtime ) {
+    std::cerr << "RtAudio alsa: " << 
+             (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") << 
+             "running realtime scheduling" << std::endl;
+  }
+#endif
+
+  while ( *isRunning == true ) {
+    pthread_testcancel();
+    object->callbackEvent();
+  }
+
+  pthread_exit( NULL );
+}
+
+//******************** End of __LINUX_ALSA__ *********************//
+#endif
+
+#if defined(__LINUX_PULSE__)
+
+// Code written by Peter Meerwald, pmeerw@pmeerw.net
+// and Tristan Matthews.
+
+#include <pulse/error.h>
+#include <pulse/simple.h>
+#include <pulse/pulseaudio.h>
+#include <cstdio>
+
+static pa_mainloop_api *rt_pa_mainloop_api = NULL;
+struct PaDeviceInfo {
+  PaDeviceInfo() : sink_index(-1), source_index(-1) {}
+  int sink_index;
+  int source_index;
+  std::string sink_name;
+  std::string source_name;
+  RtAudio::DeviceInfo info;
+};
+static struct {
+  std::vector<PaDeviceInfo> dev;
+  std::string default_sink_name;
+  std::string default_source_name;
+  int default_rate;
+} rt_pa_info;
+
+static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
+                                                      44100, 48000, 96000, 0};
+
+struct rtaudio_pa_format_mapping_t {
+  RtAudioFormat rtaudio_format;
+  pa_sample_format_t pa_format;
+};
+
+static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
+  {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
+  {RTAUDIO_SINT24, PA_SAMPLE_S24LE},
+  {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
+  {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
+  {0, PA_SAMPLE_INVALID}};
+
+struct PulseAudioHandle {
+  pa_simple *s_play;
+  pa_simple *s_rec;
+  pthread_t thread;
+  pthread_cond_t runnable_cv;
+  bool runnable;
+  PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
+};
+
+static void rt_pa_mainloop_api_quit(int ret) {
+    rt_pa_mainloop_api->quit(rt_pa_mainloop_api, ret);
+}
+
+static void rt_pa_server_callback(pa_context *context, const pa_server_info *info, void *data){
+  (void)context;
+  (void)data;
+  pa_sample_spec ss;
+
+  if (!info)
+    rt_pa_mainloop_api_quit(1);
+
+  ss = info->sample_spec;
+
+  rt_pa_info.default_rate = ss.rate;
+  rt_pa_info.default_sink_name = info->default_sink_name;
+  rt_pa_info.default_source_name = info->default_source_name;
+  rt_pa_mainloop_api_quit(0);
+}
+
+static void rt_pa_sink_info_cb(pa_context * /*c*/, const pa_sink_info *i,
+                               int eol, void * /*userdata*/)
+{
+  if (eol) return;
+  PaDeviceInfo inf;
+  inf.info.name = pa_proplist_gets(i->proplist, "device.description");
+  inf.info.probed = true;
+  inf.info.outputChannels = i->sample_spec.channels;
+  inf.info.preferredSampleRate = i->sample_spec.rate;
+  inf.info.isDefaultOutput = (rt_pa_info.default_sink_name == i->name);
+  inf.sink_index = i->index;
+  inf.sink_name = i->name;
+  for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
+    inf.info.sampleRates.push_back( *sr );
+  for ( const rtaudio_pa_format_mapping_t *fm = supported_sampleformats;
+        fm->rtaudio_format; ++fm )
+    inf.info.nativeFormats |= fm->rtaudio_format;
+  for (size_t i=0; i < rt_pa_info.dev.size(); i++)
+  {
+    /* Attempt to match up sink and source records by device description. */
+    if (rt_pa_info.dev[i].info.name == inf.info.name) {
+      rt_pa_info.dev[i].sink_index = inf.sink_index;
+      rt_pa_info.dev[i].sink_name = inf.sink_name;
+      rt_pa_info.dev[i].info.outputChannels = inf.info.outputChannels;
+      rt_pa_info.dev[i].info.isDefaultOutput = inf.info.isDefaultOutput;
+      /* Assume duplex channels are minimum of input and output channels. */
+      /* Uncomment if we add support for DUPLEX
+      if (rt_pa_info.dev[i].source_index > -1)
+        (inf.info.outputChannels < rt_pa_info.dev[i].info.inputChannels)
+          ? inf.info.outputChannels : rt_pa_info.dev[i].info.inputChannels;
+      */
+      return;
+    }
+  }
+  /* try to ensure device #0 is the default */
+  if (inf.info.isDefaultOutput)
+    rt_pa_info.dev.insert(rt_pa_info.dev.begin(), inf);
+  else
+    rt_pa_info.dev.push_back(inf);
+}
+
+static void rt_pa_source_info_cb(pa_context * /*c*/, const pa_source_info *i,
+                                 int eol, void * /*userdata*/)
+{
+  if (eol) return;
+  PaDeviceInfo inf;
+  inf.info.name = pa_proplist_gets(i->proplist, "device.description");
+  inf.info.probed = true;
+  inf.info.inputChannels = i->sample_spec.channels;
+  inf.info.preferredSampleRate = i->sample_spec.rate;
+  inf.info.isDefaultInput = (rt_pa_info.default_source_name == i->name);
+  inf.source_index = i->index;
+  inf.source_name = i->name;
+  for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
+    inf.info.sampleRates.push_back( *sr );
+  for ( const rtaudio_pa_format_mapping_t *fm = supported_sampleformats;
+        fm->rtaudio_format; ++fm )
+    inf.info.nativeFormats |= fm->rtaudio_format;
+
+  for (size_t i=0; i < rt_pa_info.dev.size(); i++)
+  {
+    /* Attempt to match up sink and source records by device description. */
+    if (rt_pa_info.dev[i].info.name == inf.info.name) {
+      rt_pa_info.dev[i].source_index = inf.source_index;
+      rt_pa_info.dev[i].source_name = inf.source_name;
+      rt_pa_info.dev[i].info.inputChannels = inf.info.inputChannels;
+      rt_pa_info.dev[i].info.isDefaultInput = inf.info.isDefaultInput;
+      /* Assume duplex channels are minimum of input and output channels. */
+      /* Uncomment if we add support for DUPLEX
+      if (rt_pa_info.dev[i].sink_index > -1) {
+        rt_pa_info.dev[i].info.duplexChannels =
+          (inf.info.inputChannels < rt_pa_info.dev[i].info.outputChannels)
+          ? inf.info.inputChannels : rt_pa_info.dev[i].info.outputChannels;
+      }
+      */
+      return;
+    }
+  }
+  /* try to ensure device #0 is the default */
+  if (inf.info.isDefaultInput)
+    rt_pa_info.dev.insert(rt_pa_info.dev.begin(), inf);
+  else
+    rt_pa_info.dev.push_back(inf);
+}
+
+static void rt_pa_context_state_callback(pa_context *context, void *userdata) {
+  (void)userdata;
+
+  switch (pa_context_get_state(context)) {
+    case PA_CONTEXT_CONNECTING:
+    case PA_CONTEXT_AUTHORIZING:
+    case PA_CONTEXT_SETTING_NAME:
+      break;
+
+    case PA_CONTEXT_READY:
+      rt_pa_info.dev.clear();
+      pa_context_get_server_info(context, rt_pa_server_callback, NULL);
+      pa_context_get_sink_info_list(context, rt_pa_sink_info_cb, NULL);
+      pa_context_get_source_info_list(context, rt_pa_source_info_cb, NULL);
+      break;
+
+    case PA_CONTEXT_TERMINATED:
+      rt_pa_mainloop_api_quit(0);
+      break;
+
+    case PA_CONTEXT_FAILED:
+    default:
+      rt_pa_mainloop_api_quit(1);
+  }
+}
+
+RtApiPulse::~RtApiPulse()
+{
+  if ( stream_.state != STREAM_CLOSED )
+    closeStream();
+}
+
+void RtApiPulse::collectDeviceInfo( void )
+{
+  pa_context *context = NULL;
+  pa_mainloop *m = NULL;
+  char *server = NULL;
+  int ret = 1;
+
+  if (!(m = pa_mainloop_new())) {
+    errorStream_ << "RtApiPulse::DeviceInfo pa_mainloop_new() failed.";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::WARNING );
+    goto quit;
+  }
+
+  rt_pa_mainloop_api = pa_mainloop_get_api(m);
+
+  if (!(context = pa_context_new_with_proplist(rt_pa_mainloop_api, NULL, NULL))) {
+    errorStream_ << "pa_context_new() failed.";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::WARNING );
+    goto quit;
+  }
+
+  pa_context_set_state_callback(context, rt_pa_context_state_callback, NULL);
+
+  if (pa_context_connect(context, server, PA_CONTEXT_NOFLAGS, NULL) < 0) {
+    errorStream_ << "RtApiPulse::DeviceInfo pa_context_connect() failed: "
+      << pa_strerror(pa_context_errno(context));
+    errorText_ = errorStream_.str();
+    error( RtAudioError::WARNING );
+    goto quit;
+  }
+
+  if (pa_mainloop_run(m, &ret) < 0) {
+    errorStream_ << "pa_mainloop_run() failed.";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::WARNING );
+    goto quit;
+  }
+
+quit:
+  if (context)
+    pa_context_unref(context);
+
+  if (m) {
+    pa_mainloop_free(m);
+  }
+
+  pa_xfree(server);
+}
+
+unsigned int RtApiPulse::getDeviceCount( void )
+{
+  collectDeviceInfo();
+  return rt_pa_info.dev.size();
+}
+
+RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int device )
+{
+  if (rt_pa_info.dev.size()==0)
+      collectDeviceInfo();
+  if (device < rt_pa_info.dev.size())
+    return rt_pa_info.dev[device].info;
+  return RtAudio::DeviceInfo();
+}
+
+static void *pulseaudio_callback( void * user )
+{
+  CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
+  RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
+  volatile bool *isRunning = &cbi->isRunning;
+  
+#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
+  if (cbi->doRealtime) {
+    std::cerr << "RtAudio pulse: " << 
+             (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") << 
+             "running realtime scheduling" << std::endl;
+  }
+#endif
+  
+  while ( *isRunning ) {
+    pthread_testcancel();
+    context->callbackEvent();
+  }
+
+  pthread_exit( NULL );
+}
+
+void RtApiPulse::closeStream( void )
+{
+  PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+  stream_.callbackInfo.isRunning = false;
+  if ( pah ) {
+    MUTEX_LOCK( &stream_.mutex );
+    if ( stream_.state == STREAM_STOPPED ) {
+      pah->runnable = true;
+      pthread_cond_signal( &pah->runnable_cv );
+    }
+    MUTEX_UNLOCK( &stream_.mutex );
+
+    pthread_join( pah->thread, 0 );
+    if ( pah->s_play ) {
+      pa_simple_flush( pah->s_play, NULL );
+      pa_simple_free( pah->s_play );
+    }
+    if ( pah->s_rec )
+      pa_simple_free( pah->s_rec );
+
+    pthread_cond_destroy( &pah->runnable_cv );
+    delete pah;
+    stream_.apiHandle = 0;
+  }
+
+  if ( stream_.userBuffer[0] ) {
+    free( stream_.userBuffer[0] );
+    stream_.userBuffer[0] = 0;
+  }
+  if ( stream_.userBuffer[1] ) {
+    free( stream_.userBuffer[1] );
+    stream_.userBuffer[1] = 0;
+  }
+
+  stream_.state = STREAM_CLOSED;
+  stream_.mode = UNINITIALIZED;
+}
+
+void RtApiPulse::callbackEvent( void )
+{
+  PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+  if ( stream_.state == STREAM_STOPPED ) {
+    MUTEX_LOCK( &stream_.mutex );
+    while ( !pah->runnable )
+      pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
+
+    if ( stream_.state != STREAM_RUNNING ) {
+      MUTEX_UNLOCK( &stream_.mutex );
+      return;
+    }
+    MUTEX_UNLOCK( &stream_.mutex );
+  }
+
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
+      "this shouldn't happen!";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+  double streamTime = getStreamTime();
+  RtAudioStreamStatus status = 0;
+  int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
+                               stream_.bufferSize, streamTime, status,
+                               stream_.callbackInfo.userData );
+
+  if ( doStopStream == 2 ) {
+    abortStream();
+    return;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+  void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
+  void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
+
+  if ( stream_.state != STREAM_RUNNING )
+    goto unlock;
+
+  int pa_error;
+  size_t bytes;
+  if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+    if ( stream_.doConvertBuffer[OUTPUT] ) {
+        convertBuffer( stream_.deviceBuffer,
+                       stream_.userBuffer[OUTPUT],
+                       stream_.convertInfo[OUTPUT] );
+        bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
+                formatBytes( stream_.deviceFormat[OUTPUT] );
+    } else
+        bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
+                formatBytes( stream_.userFormat );
+
+    if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
+      errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
+        pa_strerror( pa_error ) << ".";
+      errorText_ = errorStream_.str();
+      error( RtAudioError::WARNING );
+    }
+  }
+
+  if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
+    if ( stream_.doConvertBuffer[INPUT] )
+      bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
+        formatBytes( stream_.deviceFormat[INPUT] );
+    else
+      bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
+        formatBytes( stream_.userFormat );
+            
+    if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
+      errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
+        pa_strerror( pa_error ) << ".";
+      errorText_ = errorStream_.str();
+      error( RtAudioError::WARNING );
+    }
+    if ( stream_.doConvertBuffer[INPUT] ) {
+      convertBuffer( stream_.userBuffer[INPUT],
+                     stream_.deviceBuffer,
+                     stream_.convertInfo[INPUT] );
+    }
+  }
+
+ unlock:
+  MUTEX_UNLOCK( &stream_.mutex );
+  RtApi::tickStreamTime();
+
+  if ( doStopStream == 1 )
+    stopStream();
+}
+
+void RtApiPulse::startStream( void )
+{
+  PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiPulse::startStream(): the stream is not open!";
+    error( RtAudioError::INVALID_USE );
+    return;
+  }
+  if ( stream_.state == STREAM_RUNNING ) {
+    errorText_ = "RtApiPulse::startStream(): the stream is already running!";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  #if defined( HAVE_GETTIMEOFDAY )
+  gettimeofday( &stream_.lastTickTimestamp, NULL );
+  #endif
+
+  stream_.state = STREAM_RUNNING;
+
+  pah->runnable = true;
+  pthread_cond_signal( &pah->runnable_cv );
+  MUTEX_UNLOCK( &stream_.mutex );
+}
+
+void RtApiPulse::stopStream( void )
+{
+  PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
+    error( RtAudioError::INVALID_USE );
+    return;
+  }
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  stream_.state = STREAM_STOPPED;
+  MUTEX_LOCK( &stream_.mutex );
+
+  if ( pah ) {
+    pah->runnable = false;
+    if ( pah->s_play ) {
+      int pa_error;
+      if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
+        errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
+          pa_strerror( pa_error ) << ".";
+        errorText_ = errorStream_.str();
+        MUTEX_UNLOCK( &stream_.mutex );
+        error( RtAudioError::SYSTEM_ERROR );
+        return;
+      }
+    }
+  }
+
+  stream_.state = STREAM_STOPPED;
+  MUTEX_UNLOCK( &stream_.mutex );
+}
+
+void RtApiPulse::abortStream( void )
+{
+  PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
+
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
+    error( RtAudioError::INVALID_USE );
+    return;
+  }
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
+    error( RtAudioError::WARNING );
+    return;
+  }
+
+  stream_.state = STREAM_STOPPED;
+  MUTEX_LOCK( &stream_.mutex );
+
+  if ( pah ) {
+    pah->runnable = false;
+    if ( pah->s_play ) {
+      int pa_error;
+      if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
+        errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
+          pa_strerror( pa_error ) << ".";
+        errorText_ = errorStream_.str();
+        MUTEX_UNLOCK( &stream_.mutex );
+        error( RtAudioError::SYSTEM_ERROR );
+        return;
+      }
+    }
+  }
+
+  stream_.state = STREAM_STOPPED;
+  MUTEX_UNLOCK( &stream_.mutex );
+}
+
+bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
+                                  unsigned int channels, unsigned int firstChannel,
+                                  unsigned int sampleRate, RtAudioFormat format,
+                                  unsigned int *bufferSize, RtAudio::StreamOptions *options )
+{
+  PulseAudioHandle *pah = 0;
+  unsigned long bufferBytes = 0;
+  pa_sample_spec ss;
+
+  if ( device >= rt_pa_info.dev.size() ) return false;
+  if ( firstChannel != 0 ) {
+    errorText_ = "PulseAudio does not support channel offset mapping.";
+    return false;
+  }
+
+  /* these may be NULL for default, but we've already got the names */
+  const char *dev_input = NULL;
+  const char *dev_output = NULL;
+  if (!rt_pa_info.dev[device].source_name.empty())
+    dev_input = rt_pa_info.dev[device].source_name.c_str();
+  if (!rt_pa_info.dev[device].sink_name.empty())
+    dev_output = rt_pa_info.dev[device].sink_name.c_str();
+
+  if (mode==INPUT && rt_pa_info.dev[device].info.inputChannels == 0) {
+    errorText_ = "PulseAudio device does not support input.";
+    return false;
+  }
+  if (mode==OUTPUT && rt_pa_info.dev[device].info.outputChannels == 0) {
+    errorText_ = "PulseAudio device does not support output.";
+    return false;
+  }
+  if (mode==DUPLEX && rt_pa_info.dev[device].info.duplexChannels == 0) {
+    /* Note: will always error, DUPLEX not yet supported */
+    errorText_ = "PulseAudio device does not support duplex.";
+    return false;
+  }
+
+  if (mode==INPUT && rt_pa_info.dev[device].info.inputChannels < channels) {
+    errorText_ = "PulseAudio: unsupported number of input channels.";
+    return false;
+  }
+
+  if (mode==OUTPUT && rt_pa_info.dev[device].info.outputChannels < channels) {
+    errorText_ = "PulseAudio: unsupported number of output channels.";
+    return false;
+  }
+
+  if (mode==DUPLEX && rt_pa_info.dev[device].info.duplexChannels < channels) {
+    /* Note: will always error, DUPLEX not yet supported */
+    errorText_ = "PulseAudio: unsupported number of duplex channels.";
+    return false;
+  }
+
+  ss.channels = channels;
+
+  bool sr_found = false;
+  for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
+    if ( sampleRate == *sr ) {
+      sr_found = true;
+      stream_.sampleRate = sampleRate;
+      ss.rate = sampleRate;
+      break;
+    }
+  }
+  if ( !sr_found ) {
+    errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
+    return false;
+  }
+
+  bool sf_found = 0;
+  for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
+        sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
+    if ( format == sf->rtaudio_format ) {
+      sf_found = true;
+      stream_.userFormat = sf->rtaudio_format;
+      stream_.deviceFormat[mode] = stream_.userFormat;
+      ss.format = sf->pa_format;
+      break;
+    }
+  }
+  if ( !sf_found ) { // Use internal data format conversion.
+    stream_.userFormat = format;
+    stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+    ss.format = PA_SAMPLE_FLOAT32LE;
+  }
+
+  // Set other stream parameters.
+  if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+  else stream_.userInterleaved = true;
+  stream_.deviceInterleaved[mode] = true;
+  stream_.nBuffers = options ? options->numberOfBuffers : 1;
+  stream_.doByteSwap[mode] = false;
+  stream_.nUserChannels[mode] = channels;
+  stream_.nDeviceChannels[mode] = channels + firstChannel;
+  stream_.channelOffset[mode] = 0;
+  std::string streamName = "RtAudio";
+
+  // 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_.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_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
+    goto error;
+  }
+  stream_.bufferSize = *bufferSize;
+
+  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_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
+        goto error;
+      }
+    }
+  }
+
+  stream_.device[mode] = device;
+
+  // Setup the buffer conversion information structure.
+  if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+
+  if ( !stream_.apiHandle ) {
+    PulseAudioHandle *pah = new PulseAudioHandle;
+    if ( !pah ) {
+      errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
+      goto error;
+    }
+
+    stream_.apiHandle = pah;
+    if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
+      errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
+      goto error;
+    }
+  }
+  pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+  int error;
+  if ( options && !options->streamName.empty() ) streamName = options->streamName;
+  switch ( mode ) {
+    pa_buffer_attr buffer_attr;
+  case INPUT:
+    buffer_attr.fragsize = bufferBytes;
+    buffer_attr.maxlength = -1;
+
+    pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD,
+                                dev_input, "Record", &ss, NULL, &buffer_attr, &error );
+    if ( !pah->s_rec ) {
+      errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
+      goto error;
+    }
+    break;
+  case OUTPUT: {
+    pa_buffer_attr * attr_ptr;
+
+    if ( options && options->numberOfBuffers > 0 ) {
+      // pa_buffer_attr::fragsize is recording-only.
+      // Hopefully PortAudio won't access uninitialized fields.
+      buffer_attr.maxlength = bufferBytes * options->numberOfBuffers;
+      buffer_attr.minreq = -1;
+      buffer_attr.prebuf = -1;
+      buffer_attr.tlength = -1;
+      attr_ptr = &buffer_attr;
+    } else {
+      attr_ptr = nullptr;
+    }
+
+    pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK,
+                                 dev_output, "Playback", &ss, NULL, attr_ptr, &error );
+    if ( !pah->s_play ) {
+      errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
+      goto error;
+    }
+    break;
+  }
+  case DUPLEX:
+    /* Note: We could add DUPLEX by synchronizing multiple streams,
+       but it would mean moving from Simple API to Asynchronous API:
+       https://freedesktop.org/software/pulseaudio/doxygen/streams.html#sync_streams */
+    errorText_ = "RtApiPulse::probeDeviceOpen: duplex not supported for PulseAudio.";
+    goto error;
+  default:
+    goto error;
+  }
+
+  if ( stream_.mode == UNINITIALIZED )
+    stream_.mode = mode;
+  else if ( stream_.mode == mode )
+    goto error;
+  else
+    stream_.mode = DUPLEX;
+
+  if ( !stream_.callbackInfo.isRunning ) {
+    stream_.callbackInfo.object = this;
+    
+    stream_.state = STREAM_STOPPED;
+    // 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 (e.g. NetBSD 1.6.x with GNU Pthread)
+    if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+      stream_.callbackInfo.doRealtime = true;
+      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;
+      
+      // Set the policy BEFORE the priority. Otherwise it fails.
+      pthread_attr_setschedpolicy(&attr, SCHED_RR);
+      pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
+      // This is definitely required. Otherwise it fails.
+      pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
+      pthread_attr_setschedparam(&attr, &param);
+    }
+    else
+      pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#else
+    pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#endif
+
+    stream_.callbackInfo.isRunning = true;
+    int result = pthread_create( &pah->thread, &attr, pulseaudio_callback, (void *)&stream_.callbackInfo);
+    pthread_attr_destroy(&attr);
+    if(result != 0) {
+      // Failed. Try instead with default attributes.
+      result = pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo);
+      if(result != 0) {
+        stream_.callbackInfo.isRunning = false;
+        errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
+        goto error;
+      }
+    }
+  }
+
+  return SUCCESS;
+ 
+ error:
+  if ( pah && stream_.callbackInfo.isRunning ) {
+    pthread_cond_destroy( &pah->runnable_cv );
+    delete pah;
+    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_.state = STREAM_CLOSED;
+  return FAILURE;
+}
+
+//******************** End of __LINUX_PULSE__ *********************//
+#endif
+
+#if defined(__LINUX_OSS__)
+
+#include <unistd.h>
+#include <sys/ioctl.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/soundcard.h>
+#include <errno.h>
+#include <math.h>
+
+static 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( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::WARNING );
+    return info;
+  }
+
+  unsigned nDevices = sysinfo.numaudios;
+  if ( nDevices == 0 ) {
+    close( mixerfd );
+    errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
+    error( RtAudioError::INVALID_USE );
+    return info;
+  }
+
+  if ( device >= nDevices ) {
+    close( mixerfd );
+    errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
+    error( RtAudioError::INVALID_USE );
+    return info;
+  }
+
+  oss_audioinfo ainfo;
+  ainfo.dev = device;
+  result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
+  close( mixerfd );
+  if ( result == -1 ) {
+    errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::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;
+#ifdef AFMT_FLOAT
+  if ( mask & AFMT_FLOAT )
+    info.nativeFormats |= RTAUDIO_FLOAT32;
+#endif
+  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( RtAudioError::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] );
+
+          if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
+            info.preferredSampleRate = 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.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
+          info.preferredSampleRate = SAMPLE_RATES[k];
+      }
+    }
+  }
+
+  if ( info.sampleRates.size() == 0 ) {
+    errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
+    errorText_ = errorStream_.str();
+    error( RtAudioError::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 - (int)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 (e.g. NetBSD 1.6.x with GNU Pthread)
+    if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+      stream_.callbackInfo.doRealtime = true;
+      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;
+      
+      // Set the policy BEFORE the priority. Otherwise it fails.
+      pthread_attr_setschedpolicy(&attr, SCHED_RR);
+      pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
+      // This is definitely required. Otherwise it fails.
+      pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
+      pthread_attr_setschedparam(&attr, &param);
+    }
+    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 ) {
+      // Failed. Try instead with default attributes.
+      result = pthread_create( &stream_.callbackInfo.thread, NULL, ossCallbackHandler, &stream_.callbackInfo );
+      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;
+  }
+
+  stream_.state = STREAM_CLOSED;
+  return FAILURE;
+}
+
+void RtApiOss :: closeStream()
+{
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApiOss::closeStream(): no open stream to close!";
+    error( RtAudioError::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( RtAudioError::WARNING );
+    return;
+  }
+
+  MUTEX_LOCK( &stream_.mutex );
+
+  #if defined( HAVE_GETTIMEOFDAY )
+  gettimeofday( &stream_.lastTickTimestamp, NULL );
+  #endif
+
+  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( RtAudioError::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( RtAudioError::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( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiOss :: abortStream()
+{
+  verifyStream();
+  if ( stream_.state == STREAM_STOPPED ) {
+    errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
+    error( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::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();
+}
+
+static void *ossCallbackHandler( void *ptr )
+{
+  CallbackInfo *info = (CallbackInfo *) ptr;
+  RtApiOss *object = (RtApiOss *) info->object;
+  bool *isRunning = &info->isRunning;
+
+#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
+  if (info->doRealtime) {
+    std::cerr << "RtAudio oss: " << 
+             (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") << 
+             "running realtime scheduling" << std::endl;
+  }
+#endif
+
+  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( RtAudioError::Type type )
+{
+  errorStream_.str(""); // clear the ostringstream
+
+  RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
+  if ( errorCallback ) {
+    // abortStream() can generate new error messages. Ignore them. Just keep original one.
+
+    if ( firstErrorOccurred_ )
+      return;
+
+    firstErrorOccurred_ = true;
+    const std::string errorMessage = errorText_;
+
+    if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
+      stream_.callbackInfo.isRunning = false; // exit from the thread
+      abortStream();
+    }
+
+    errorCallback( type, errorMessage );
+    firstErrorOccurred_ = false;
+    return;
+  }
+
+  if ( type == RtAudioError::WARNING && showWarnings_ == true )
+    std::cerr << '\n' << errorText_ << "\n\n";
+  else if ( type != RtAudioError::WARNING )
+    throw( RtAudioError( errorText_, type ) );
+}
+
+void RtApi :: verifyStream()
+{
+  if ( stream_.state == STREAM_CLOSED ) {
+    errorText_ = "RtApi:: a stream is not open!";
+    error( RtAudioError::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;
+  stream_.callbackInfo.errorCallback = 0;
+  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_SINT32 || format == RTAUDIO_FLOAT32 )
+    return 4;
+  else if ( format == RTAUDIO_FLOAT64 )
+    return 8;
+  else if ( format == RTAUDIO_SINT24 )
+    return 3;
+  else if ( format == RTAUDIO_SINT8 )
+    return 1;
+
+  errorText_ = "RtApi::formatBytes: undefined format.";
+  error( RtAudioError::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 lower three bytes of a 32-bit integer.
+
+  // Clear our duplex device output buffer if there are more device outputs than user outputs
+  if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX && info.outJump > info.inJump )
+    memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
+
+  int j;
+  if (info.outFormat == RTAUDIO_FLOAT64) {
+    Float64 *out = (Float64 *)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]] = (Float64) in[info.inOffset[j]] / 128.0;
+        }
+        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]] = (Float64) in[info.inOffset[j]] / 32768.0;
+        }
+        in += info.inJump;
+        out += info.outJump;
+      }
+    }
+    else if (info.inFormat == RTAUDIO_SINT24) {
+      Int24 *in = (Int24 *)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]].asInt() / 8388608.0;
+        }
+        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]] = (Float64) in[info.inOffset[j]] / 2147483648.0;
+        }
+        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 *out = (Float32 *)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]] = (Float32) in[info.inOffset[j]] / 128.f;
+        }
+        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]] = (Float32) in[info.inOffset[j]] / 32768.f;
+        }
+        in += info.inJump;
+        out += info.outJump;
+      }
+    }
+    else if (info.inFormat == RTAUDIO_SINT24) {
+      Int24 *in = (Int24 *)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]].asInt() / 8388608.f;
+        }
+        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]] = (Float32) in[info.inOffset[j]] / 2147483648.f;
+        }
+        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) {
+      Int24 *in = (Int24 *)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]].asInt();
+          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++) {
+          // Use llround() which returns `long long` which is guaranteed to be at least 64 bits.
+          out[info.outOffset[j]] = (Int32) std::min(std::llround(in[info.inOffset[j]] * 2147483648.f), 2147483647LL);
+        }
+        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) std::min(std::llround(in[info.inOffset[j]] * 2147483648.0), 2147483647LL);
+        }
+        in += info.inJump;
+        out += info.outJump;
+      }
+    }
+  }
+  else if (info.outFormat == RTAUDIO_SINT24) {
+    Int24 *out = (Int24 *)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]] << 16);
+          //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]] << 8);
+          //out[info.outOffset[j]] <<= 8;
+        }
+        in += info.inJump;
+        out += info.outJump;
+      }
+    }
+    else if (info.inFormat == RTAUDIO_SINT24) {
+      // Channel compensation and/or (de)interleaving only.
+      Int24 *in = (Int24 *)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]] >> 8);
+          //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) std::min(std::llround(in[info.inOffset[j]] * 8388608.f), 8388607LL);
+        }
+        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) std::min(std::llround(in[info.inOffset[j]] * 8388608.0), 8388607LL);
+        }
+        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) {
+      Int24 *in = (Int24 *)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]].asInt() >> 8);
+        }
+        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) std::min(std::llround(in[info.inOffset[j]] * 32768.f), 32767LL);
+        }
+        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) std::min(std::llround(in[info.inOffset[j]] * 32768.0), 32767LL);
+        }
+        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) {
+      Int24 *in = (Int24 *)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]].asInt() >> 16);
+        }
+        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) std::min(std::llround(in[info.inOffset[j]] * 128.f), 127LL);
+        }
+        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) std::min(std::llround(in[info.inOffset[j]] * 128.0), 127LL);
+        }
+        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 )
+{
+  char val;
+  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_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_SINT24 ) {
+    for ( unsigned int i=0; i<samples; i++ ) {
+      // Swap 1st and 3rd bytes.
+      val = *(ptr);
+      *(ptr) = *(ptr+2);
+      *(ptr+2) = val;
+
+      // Increment 2 more bytes.
+      ptr += 2;
+    }
+  }
+  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
+