Updates for release 4.0.8, including new python binding, new teststops.cpp program, ALSA "default" flag, and various changes to stopping behavior (GS).

1 files changed, 7960 insertions, 7915 deletions

diff --git a/RtAudio.cpp b/RtAudio.cpp
index 028fffa..7dd1460 100644
--- a/RtAudio.cpp
+++ b/RtAudio.cpp
@@ -1,7915 +1,7960 @@
-/************************************************************************/
-/*! \class RtAudio
-    \brief Realtime audio i/o C++ classes.
-
-    RtAudio provides a common API (Application Programming Interface)
-    for realtime audio input/output across Linux (native ALSA, Jack,
-    and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
-    (DirectSound and ASIO) operating systems.
-
-    RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
-
-    RtAudio: realtime audio i/o C++ classes
-    Copyright (c) 2001-2011 Gary P. Scavone
-
-    Permission is hereby granted, free of charge, to any person
-    obtaining a copy of this software and associated documentation files
-    (the "Software"), to deal in the Software without restriction,
-    including without limitation the rights to use, copy, modify, merge,
-    publish, distribute, sublicense, and/or sell copies of the Software,
-    and to permit persons to whom the Software is furnished to do so,
-    subject to the following conditions:
-
-    The above copyright notice and this permission notice shall be
-    included in all copies or substantial portions of the Software.
-
-    Any person wishing to distribute modifications to the Software is
-    asked to send the modifications to the original developer so that
-    they can be incorporated into the canonical version.  This is,
-    however, not a binding provision of this license.
-
-    THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
-    EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
-    MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
-    IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
-    ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
-    CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
-    WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
-*/
-/************************************************************************/
-
-// RtAudio: Version 4.0.8
-
-#include "RtAudio.h"
-#include <iostream>
-#include <cstdlib>
-#include <cstring>
-#include <climits>
-
-// Static variable definitions.
-const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
-const unsigned int RtApi::SAMPLE_RATES[] = {
-  4000, 5512, 8000, 9600, 11025, 16000, 22050,
-  32000, 44100, 48000, 88200, 96000, 176400, 192000
-};
-
-#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__)
-  #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
-  #define MUTEX_DESTROY(A)    DeleteCriticalSection(A)
-  #define MUTEX_LOCK(A)       EnterCriticalSection(A)
-  #define MUTEX_UNLOCK(A)     LeaveCriticalSection(A)
-#elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
-  // pthread API
-  #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
-  #define MUTEX_DESTROY(A)    pthread_mutex_destroy(A)
-  #define MUTEX_LOCK(A)       pthread_mutex_lock(A)
-  #define MUTEX_UNLOCK(A)     pthread_mutex_unlock(A)
-#else
-  #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions
-  #define MUTEX_DESTROY(A)    abs(*A) // dummy definitions
-#endif
-
-// *************************************************** //
-//
-// RtAudio definitions.
-//
-// *************************************************** //
-
-void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()
-{
-  apis.clear();
-
-  // The order here will control the order of RtAudio's API search in
-  // the constructor.
-#if defined(__UNIX_JACK__)
-  apis.push_back( UNIX_JACK );
-#endif
-#if defined(__LINUX_ALSA__)
-  apis.push_back( LINUX_ALSA );
-#endif
-#if defined(__LINUX_OSS__)
-  apis.push_back( LINUX_OSS );
-#endif
-#if defined(__WINDOWS_ASIO__)
-  apis.push_back( WINDOWS_ASIO );
-#endif
-#if defined(__WINDOWS_DS__)
-  apis.push_back( WINDOWS_DS );
-#endif
-#if defined(__MACOSX_CORE__)
-  apis.push_back( MACOSX_CORE );
-#endif
-#if defined(__RTAUDIO_DUMMY__)
-  apis.push_back( RTAUDIO_DUMMY );
-#endif
-}
-
-void RtAudio :: openRtApi( RtAudio::Api api )
-{
-#if defined(__UNIX_JACK__)
-  if ( api == UNIX_JACK )
-    rtapi_ = new RtApiJack();
-#endif
-#if defined(__LINUX_ALSA__)
-  if ( api == LINUX_ALSA )
-    rtapi_ = new RtApiAlsa();
-#endif
-#if defined(__LINUX_OSS__)
-  if ( api == LINUX_OSS )
-    rtapi_ = new RtApiOss();
-#endif
-#if defined(__WINDOWS_ASIO__)
-  if ( api == WINDOWS_ASIO )
-    rtapi_ = new RtApiAsio();
-#endif
-#if defined(__WINDOWS_DS__)
-  if ( api == WINDOWS_DS )
-    rtapi_ = new RtApiDs();
-#endif
-#if defined(__MACOSX_CORE__)
-  if ( api == MACOSX_CORE )
-    rtapi_ = new RtApiCore();
-#endif
-#if defined(__RTAUDIO_DUMMY__)
-  if ( api == RTAUDIO_DUMMY )
-    rtapi_ = new RtApiDummy();
-#endif
-}
-
-RtAudio :: RtAudio( RtAudio::Api api ) throw()
-{
-  rtapi_ = 0;
-
-  if ( api != UNSPECIFIED ) {
-    // Attempt to open the specified API.
-    openRtApi( api );
-    if ( rtapi_ ) return;
-
-    // No compiled support for specified API value.  Issue a debug
-    // warning and continue as if no API was specified.
-    std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
-  }
-
-  // Iterate through the compiled APIs and return as soon as we find
-  // one with at least one device or we reach the end of the list.
-  std::vector< RtAudio::Api > apis;
-  getCompiledApi( apis );
-  for ( unsigned int i=0; i<apis.size(); i++ ) {
-    openRtApi( apis[i] );
-    if ( rtapi_->getDeviceCount() ) break;
-  }
-
-  if ( rtapi_ ) return;
-
-  // It should not be possible to get here because the preprocessor
-  // definition __RTAUDIO_DUMMY__ is automatically defined if no
-  // API-specific definitions are passed to the compiler. But just in
-  // case something weird happens, we'll print out an error message.
-  std::cerr << "\nRtAudio: no compiled API support found ... critical error!!\n\n";
-}
-
-RtAudio :: ~RtAudio() throw()
-{
-  delete rtapi_;
-}
-
-void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
-                            RtAudio::StreamParameters *inputParameters,
-                            RtAudioFormat format, unsigned int sampleRate,
-                            unsigned int *bufferFrames,
-                            RtAudioCallback callback, void *userData,
-                            RtAudio::StreamOptions *options )
-{
-  return rtapi_->openStream( outputParameters, inputParameters, format,
-                             sampleRate, bufferFrames, callback,
-                             userData, options );
-}
-
-// *************************************************** //
-//
-// Public RtApi definitions (see end of file for
-// private or protected utility functions).
-//
-// *************************************************** //
-
-RtApi :: RtApi()
-{
-  stream_.state = STREAM_CLOSED;
-  stream_.mode = UNINITIALIZED;
-  stream_.apiHandle = 0;
-  stream_.userBuffer[0] = 0;
-  stream_.userBuffer[1] = 0;
-  MUTEX_INITIALIZE( &stream_.mutex );
-  showWarnings_ = true;
-}
-
-RtApi :: ~RtApi()
-{
-  MUTEX_DESTROY( &stream_.mutex );
-}
-
-void RtApi :: openStream( RtAudio::StreamParameters *oParams,
-                          RtAudio::StreamParameters *iParams,
-                          RtAudioFormat format, unsigned int sampleRate,
-                          unsigned int *bufferFrames,
-                          RtAudioCallback callback, void *userData,
-                          RtAudio::StreamOptions *options )
-{
-  if ( stream_.state != STREAM_CLOSED ) {
-    errorText_ = "RtApi::openStream: a stream is already open!";
-    error( RtError::INVALID_USE );
-  }
-
-  if ( oParams && oParams->nChannels < 1 ) {
-    errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
-    error( RtError::INVALID_USE );
-  }
-
-  if ( iParams && iParams->nChannels < 1 ) {
-    errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
-    error( RtError::INVALID_USE );
-  }
-
-  if ( oParams == NULL && iParams == NULL ) {
-    errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
-    error( RtError::INVALID_USE );
-  }
-
-  if ( formatBytes(format) == 0 ) {
-    errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
-    error( RtError::INVALID_USE );
-  }
-
-  unsigned int nDevices = getDeviceCount();
-  unsigned int oChannels = 0;
-  if ( oParams ) {
-    oChannels = oParams->nChannels;
-    if ( oParams->deviceId >= nDevices ) {
-      errorText_ = "RtApi::openStream: output device parameter value is invalid.";
-      error( RtError::INVALID_USE );
-    }
-  }
-
-  unsigned int iChannels = 0;
-  if ( iParams ) {
-    iChannels = iParams->nChannels;
-    if ( iParams->deviceId >= nDevices ) {
-      errorText_ = "RtApi::openStream: input device parameter value is invalid.";
-      error( RtError::INVALID_USE );
-    }
-  }
-
-  clearStreamInfo();
-  bool result;
-
-  if ( oChannels > 0 ) {
-
-    result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
-                              sampleRate, format, bufferFrames, options );
-    if ( result == false ) error( RtError::SYSTEM_ERROR );
-  }
-
-  if ( iChannels > 0 ) {
-
-    result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
-                              sampleRate, format, bufferFrames, options );
-    if ( result == false ) {
-      if ( oChannels > 0 ) closeStream();
-      error( RtError::SYSTEM_ERROR );
-    }
-  }
-
-  stream_.callbackInfo.callback = (void *) callback;
-  stream_.callbackInfo.userData = userData;
-
-  if ( options ) options->numberOfBuffers = stream_.nBuffers;
-  stream_.state = STREAM_STOPPED;
-}
-
-unsigned int RtApi :: getDefaultInputDevice( void )
-{
-  // Should be implemented in subclasses if possible.
-  return 0;
-}
-
-unsigned int RtApi :: getDefaultOutputDevice( void )
-{
-  // Should be implemented in subclasses if possible.
-  return 0;
-}
-
-void RtApi :: closeStream( void )
-{
-  // MUST be implemented in subclasses!
-  return;
-}
-
-bool RtApi :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-                               unsigned int firstChannel, unsigned int sampleRate,
-                               RtAudioFormat format, unsigned int *bufferSize,
-                               RtAudio::StreamOptions *options )
-{
-  // MUST be implemented in subclasses!
-  return FAILURE;
-}
-
-void RtApi :: tickStreamTime( void )
-{
-  // Subclasses that do not provide their own implementation of
-  // getStreamTime should call this function once per buffer I/O to
-  // provide basic stream time support.
-
-  stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
-
-#if defined( HAVE_GETTIMEOFDAY )
-  gettimeofday( &stream_.lastTickTimestamp, NULL );
-#endif
-}
-
-long RtApi :: getStreamLatency( void )
-{
-  verifyStream();
-
-  long totalLatency = 0;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
-    totalLatency = stream_.latency[0];
-  if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
-    totalLatency += stream_.latency[1];
-
-  return totalLatency;
-}
-
-double RtApi :: getStreamTime( void )
-{
-  verifyStream();
-
-#if defined( HAVE_GETTIMEOFDAY )
-  // Return a very accurate estimate of the stream time by
-  // adding in the elapsed time since the last tick.
-  struct timeval then;
-  struct timeval now;
-
-  if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
-    return stream_.streamTime;
-
-  gettimeofday( &now, NULL );
-  then = stream_.lastTickTimestamp;
-  return stream_.streamTime +
-    ((now.tv_sec + 0.000001 * now.tv_usec) -
-     (then.tv_sec + 0.000001 * then.tv_usec));     
-#else
-  return stream_.streamTime;
-#endif
-}
-
-unsigned int RtApi :: getStreamSampleRate( void )
-{
- verifyStream();
-
- return stream_.sampleRate;
-}
-
-
-// *************************************************** //
-//
-// OS/API-specific methods.
-//
-// *************************************************** //
-
-#if defined(__MACOSX_CORE__)
-
-// The OS X CoreAudio API is designed to use a separate callback
-// procedure for each of its audio devices.  A single RtAudio duplex
-// stream using two different devices is supported here, though it
-// cannot be guaranteed to always behave correctly because we cannot
-// synchronize these two callbacks.
-//
-// A property listener is installed for over/underrun information.
-// However, no functionality is currently provided to allow property
-// listeners to trigger user handlers because it is unclear what could
-// be done if a critical stream parameter (buffer size, sample rate,
-// device disconnect) notification arrived.  The listeners entail
-// quite a bit of extra code and most likely, a user program wouldn't
-// be prepared for the result anyway.  However, we do provide a flag
-// to the client callback function to inform of an over/underrun.
-//
-// The mechanism for querying and setting system parameters was
-// updated (and perhaps simplified) in OS-X version 10.4.  However,
-// since 10.4 support is not necessarily available to all users, I've
-// decided not to update the respective code at this time.  Perhaps
-// this will happen when Apple makes 10.4 free for everyone. :-)
-
-// A structure to hold various information related to the CoreAudio API
-// implementation.
-struct CoreHandle {
-  AudioDeviceID id[2];    // device ids
-#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
-  AudioDeviceIOProcID procId[2];
-#endif
-  UInt32 iStream[2];      // device stream index (or first if using multiple)
-  UInt32 nStreams[2];     // number of streams to use
-  bool xrun[2];
-  char *deviceBuffer;
-  pthread_cond_t condition;
-  int drainCounter;       // Tracks callback counts when draining
-  bool internalDrain;     // Indicates if stop is initiated from callback or not.
-
-  CoreHandle()
-    :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
-};
-
-RtApiCore :: RtApiCore()
-{
-  // Nothing to do here.
-}
-
-RtApiCore :: ~RtApiCore()
-{
-  // The subclass destructor gets called before the base class
-  // destructor, so close an existing stream before deallocating
-  // apiDeviceId memory.
-  if ( stream_.state != STREAM_CLOSED ) closeStream();
-}
-
-unsigned int RtApiCore :: getDeviceCount( void )
-{
-  // Find out how many audio devices there are, if any.
-  UInt32 dataSize;
-  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( RtError::WARNING );
-    return 0;
-  }
-
-  return dataSize / sizeof( AudioDeviceID );
-}
-
-unsigned int RtApiCore :: getDefaultInputDevice( void )
-{
-  unsigned int nDevices = getDeviceCount();
-  if ( nDevices <= 1 ) return 0;
-
-  AudioDeviceID id;
-  UInt32 dataSize = sizeof( AudioDeviceID );
-  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( RtError::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( RtError::WARNING );
-    return 0;
-  }
-
-  for ( unsigned int i=0; i<nDevices; i++ )
-    if ( id == deviceList[i] ) return i;
-
-  errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
-  error( RtError::WARNING );
-  return 0;
-}
-
-unsigned int RtApiCore :: getDefaultOutputDevice( void )
-{
-  unsigned int nDevices = getDeviceCount();
-  if ( nDevices <= 1 ) return 0;
-
-  AudioDeviceID id;
-  UInt32 dataSize = sizeof( AudioDeviceID );
-  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( RtError::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( RtError::WARNING );
-    return 0;
-  }
-
-  for ( unsigned int i=0; i<nDevices; i++ )
-    if ( id == deviceList[i] ) return i;
-
-  errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
-  error( RtError::WARNING );
-  return 0;
-}
-
-RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
-{
-  RtAudio::DeviceInfo info;
-  info.probed = false;
-
-  // Get device ID
-  unsigned int nDevices = getDeviceCount();
-  if ( nDevices == 0 ) {
-    errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
-    error( RtError::INVALID_USE );
-  }
-
-  if ( device >= nDevices ) {
-    errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
-    error( RtError::INVALID_USE );
-  }
-
-  AudioDeviceID deviceList[ nDevices ];
-  UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
-  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( RtError::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( RtError::WARNING );
-    return info;
-  }
-
-  //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
-  int length = CFStringGetLength(cfname);
-  char *mname = (char *)malloc(length * 3 + 1);
-  CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());
-  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( RtError::WARNING );
-    return info;
-  }
-
-  //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
-  length = CFStringGetLength(cfname);
-  char *name = (char *)malloc(length * 3 + 1);
-  CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());
-  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( RtError::WARNING );
-    return info;
-  }
-
-  // Allocate the AudioBufferList.
-  bufferList = (AudioBufferList *) malloc( dataSize );
-  if ( bufferList == NULL ) {
-    errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
-    error( RtError::WARNING );
-    return info;
-  }
-
-  result = 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( RtError::WARNING );
-    return info;
-  }
-
-  // Get output channel information.
-  unsigned int i, nStreams = bufferList->mNumberBuffers;
-  for ( i=0; i<nStreams; i++ )
-    info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
-  free( bufferList );
-
-  // Get the input stream "configuration".
-  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( RtError::WARNING );
-    return info;
-  }
-
-  // Allocate the AudioBufferList.
-  bufferList = (AudioBufferList *) malloc( dataSize );
-  if ( bufferList == NULL ) {
-    errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
-    error( RtError::WARNING );
-    return info;
-  }
-
-  result = 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( RtError::WARNING );
-    return info;
-  }
-
-  // Get input channel information.
-  nStreams = bufferList->mNumberBuffers;
-  for ( i=0; i<nStreams; i++ )
-    info.inputChannels += bufferList->mBuffers[i].mNumberChannels;
-  free( bufferList );
-
-  // If device opens for both playback and capture, we determine the channels.
-  if ( info.outputChannels > 0 && info.inputChannels > 0 )
-    info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
-
-  // Probe the device sample rates.
-  bool isInput = false;
-  if ( info.outputChannels == 0 ) isInput = true;
-
-  // Determine the supported sample rates.
-  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( RtError::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( RtError::WARNING );
-    return info;
-  }
-
-  Float64 minimumRate = 100000000.0, maximumRate = 0.0;
-  for ( UInt32 i=0; i<nRanges; i++ ) {
-    if ( rangeList[i].mMinimum < minimumRate ) minimumRate = rangeList[i].mMinimum;
-    if ( rangeList[i].mMaximum > maximumRate ) maximumRate = rangeList[i].mMaximum;
-  }
-
-  info.sampleRates.clear();
-  for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
-    if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate )
-      info.sampleRates.push_back( SAMPLE_RATES[k] );
-  }
-
-  if ( info.sampleRates.size() == 0 ) {
-    errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
-
-  // CoreAudio always uses 32-bit floating point data for PCM streams.
-  // Thus, any other "physical" formats supported by the device are of
-  // no interest to the client.
-  info.nativeFormats = RTAUDIO_FLOAT32;
-
-  if ( info.outputChannels > 0 )
-    if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
-  if ( info.inputChannels > 0 )
-    if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
-
-  info.probed = true;
-  return info;
-}
-
-OSStatus callbackHandler( AudioDeviceID inDevice,
-                          const AudioTimeStamp* inNow,
-                          const AudioBufferList* inInputData,
-                          const AudioTimeStamp* inInputTime,
-                          AudioBufferList* outOutputData,
-                          const AudioTimeStamp* inOutputTime, 
-                          void* infoPointer )
-{
-  CallbackInfo *info = (CallbackInfo *) infoPointer;
-
-  RtApiCore *object = (RtApiCore *) info->object;
-  if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
-    return kAudioHardwareUnspecifiedError;
-  else
-    return kAudioHardwareNoError;
-}
-
-OSStatus deviceListener( 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;
-}
-
-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) {
-    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;
-
-  // 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 ) {
-    nominalRate = (Float64) sampleRate;
-    result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );
-
-    if ( result != noErr ) {
-      errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";
-      errorText_ = errorStream_.str();
-      return FAILURE;
-    }
-  }
-
-  // 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;
-      }
-    }
-  }
-
-  // Get the stream ID(s) so we can set the stream format.
-  AudioStreamID streamIDs[ nStreams ];
-  dataSize = nStreams * sizeof( AudioStreamID );
-  property.mSelector = kAudioDevicePropertyStreams;
-  result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &streamIDs );
-
-  if ( result != noErr ) {
-    errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream ID(s) for device (" << device << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Now set the stream format for each stream.  Also, check the
-  // physical format of the device and change that if necessary.
-  AudioStreamBasicDescription	description;
-  dataSize = sizeof( AudioStreamBasicDescription );
-
-  bool updateFormat;
-  for ( UInt32 i=0; i<streamCount; i++ ) {
-
-    property.mSelector = kAudioStreamPropertyVirtualFormat;
-    result = AudioObjectGetPropertyData( streamIDs[firstStream+i], &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.
-    updateFormat = false;
-    if ( fabs( description.mSampleRate - (double)sampleRate ) > 1.0 ) {
-      description.mSampleRate = (double) sampleRate;
-      updateFormat = true;
-    }
-
-    if ( description.mFormatID != kAudioFormatLinearPCM ) {
-      description.mFormatID = kAudioFormatLinearPCM;
-      updateFormat = true;
-    }
-
-    if ( updateFormat ) {
-      result = AudioObjectSetPropertyData( streamIDs[firstStream+i], &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( streamIDs[firstStream+i], &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;
-    }
-
-    if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 24 ) {
-      description.mFormatID = kAudioFormatLinearPCM;
-      AudioStreamBasicDescription	testDescription = description;
-      unsigned long formatFlags;
-
-      // We'll try higher bit rates first and then work our way down.
-      testDescription.mBitsPerChannel = 32;
-      testDescription.mBytesPerFrame =  testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
-      testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
-      formatFlags = description.mFormatFlags | kLinearPCMFormatFlagIsFloat & ~kLinearPCMFormatFlagIsSignedInteger;
-      testDescription.mFormatFlags = formatFlags;
-      result = AudioObjectSetPropertyData( streamIDs[firstStream+i], &property, 0, NULL, dataSize, &testDescription );
-      if ( result == noErr ) continue;
-
-      testDescription = description;
-      testDescription.mBitsPerChannel = 32;
-      testDescription.mBytesPerFrame =  testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
-      testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
-      formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger) & ~kLinearPCMFormatFlagIsFloat;
-      testDescription.mFormatFlags = formatFlags;
-      result = AudioObjectSetPropertyData( streamIDs[firstStream+i], &property, 0, NULL, dataSize, &testDescription );
-      if ( result == noErr ) continue;
-
-      testDescription = description;
-      testDescription.mBitsPerChannel = 24;
-      testDescription.mBytesPerFrame =  testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
-      testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
-      testDescription.mFormatFlags = formatFlags;
-      result = AudioObjectSetPropertyData( streamIDs[firstStream+i], &property, 0, NULL, dataSize, &testDescription );
-      if ( result == noErr ) continue;
-
-      testDescription = description;
-      testDescription.mBitsPerChannel = 16;
-      testDescription.mBytesPerFrame =  testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
-      testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
-      testDescription.mFormatFlags = formatFlags;
-      result = AudioObjectSetPropertyData( streamIDs[firstStream+i], &property, 0, NULL, dataSize, &testDescription );
-      if ( result == noErr ) continue;
-
-      testDescription = description;
-      testDescription.mBitsPerChannel = 8;
-      testDescription.mBytesPerFrame =  testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
-      testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
-      testDescription.mFormatFlags = formatFlags;
-      result = AudioObjectSetPropertyData( streamIDs[firstStream+i], &property, 0, NULL, dataSize, &testDescription );
-      if ( result != noErr ) {
-        errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
-        errorText_ = errorStream_.str();
-        return FAILURE;
-      }
-    }
-  }
-
-
-  // Get the stream latency.  There can be latency in both the device
-  // and the stream.  First, attempt to get the device latency on the
-  // master channel or the first open channel.  Errors that might
-  // occur here are not deemed critical.
-
-  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( RtError::WARNING );
-    }
-  }
-
-  // Now try to get the stream latency.  For multiple streams, I assume the
-  // latency is equal for each.
-  result = AudioObjectGetPropertyData( streamIDs[firstStream], &property, 0, NULL, &dataSize, &latency );
-  if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] += latency;
-  else {
-    errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream latency for device (" << device << ").";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-  }
-
-  // Byte-swapping: According to AudioHardware.h, the stream data will
-  // always be presented in native-endian format, so we should never
-  // need to byte swap.
-  stream_.doByteSwap[mode] = false;
-
-  // From the CoreAudio documentation, PCM data must be supplied as
-  // 32-bit floats.
-  stream_.userFormat = format;
-  stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
-
-  if ( streamCount == 1 )
-    stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
-  else // multiple streams
-    stream_.nDeviceChannels[mode] = channels;
-  stream_.nUserChannels[mode] = channels;
-  stream_.channelOffset[mode] = channelOffset;  // offset within a CoreAudio stream
-  if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
-  else stream_.userInterleaved = true;
-  stream_.deviceInterleaved[mode] = true;
-  if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;
-
-  // Set flags for buffer conversion.
-  stream_.doConvertBuffer[mode] = false;
-  if ( stream_.userFormat != stream_.deviceFormat[mode] )
-    stream_.doConvertBuffer[mode] = true;
-  if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
-    stream_.doConvertBuffer[mode] = true;
-  if ( streamCount == 1 ) {
-    if ( stream_.nUserChannels[mode] > 1 &&
-         stream_.userInterleaved != stream_.deviceInterleaved[mode] )
-      stream_.doConvertBuffer[mode] = true;
-  }
-  else if ( monoMode && stream_.userInterleaved )
-    stream_.doConvertBuffer[mode] = true;
-
-  // Allocate our CoreHandle structure for the stream.
-  CoreHandle *handle = 0;
-  if ( stream_.apiHandle == 0 ) {
-    try {
-      handle = new CoreHandle;
-    }
-    catch ( std::bad_alloc& ) {
-      errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";
-      goto error;
-    }
-
-    if ( pthread_cond_init( &handle->condition, NULL ) ) {
-      errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";
-      goto error;
-    }
-    stream_.apiHandle = (void *) handle;
-  }
-  else
-    handle = (CoreHandle *) stream_.apiHandle;
-  handle->iStream[mode] = firstStream;
-  handle->nStreams[mode] = streamCount;
-  handle->id[mode] = id;
-
-  // Allocate necessary internal buffers.
-  unsigned long bufferBytes;
-  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
-  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
-  if ( stream_.userBuffer[mode] == NULL ) {
-    errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";
-    goto error;
-  }
-
-  // If possible, we will make use of the CoreAudio stream buffers as
-  // "device buffers".  However, we can't do this if using multiple
-  // streams.
-  if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {
-
-    bool makeBuffer = true;
-    bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
-    if ( mode == INPUT ) {
-      if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
-        unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
-        if ( bufferBytes <= bytesOut ) makeBuffer = false;
-      }
-    }
-
-    if ( makeBuffer ) {
-      bufferBytes *= *bufferSize;
-      if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
-      stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
-      if ( stream_.deviceBuffer == NULL ) {
-        errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
-        goto error;
-      }
-    }
-  }
-
-  stream_.sampleRate = sampleRate;
-  stream_.device[mode] = device;
-  stream_.state = STREAM_STOPPED;
-  stream_.callbackInfo.object = (void *) this;
-
-  // Setup the buffer conversion information structure.
-  if ( stream_.doConvertBuffer[mode] ) {
-    if ( streamCount > 1 ) setConvertInfo( mode, 0 );
-    else setConvertInfo( mode, channelOffset );
-  }
-
-  if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
-    // Only one callback procedure per device.
-    stream_.mode = DUPLEX;
-  else {
-#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
-    result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );
-#else
-    // deprecated in favor of AudioDeviceCreateIOProcID()
-    result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
-#endif
-    if ( result != noErr ) {
-      errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";
-      errorText_ = errorStream_.str();
-      goto error;
-    }
-    if ( stream_.mode == OUTPUT && mode == INPUT )
-      stream_.mode = DUPLEX;
-    else
-      stream_.mode = mode;
-  }
-
-  // Setup the device property listener for over/underload.
-  property.mSelector = kAudioDeviceProcessorOverload;
-  result = AudioObjectAddPropertyListener( id, &property, deviceListener, (void *) handle );
-
-  return SUCCESS;
-
- error:
-  if ( handle ) {
-    pthread_cond_destroy( &handle->condition );
-    delete handle;
-    stream_.apiHandle = 0;
-  }
-
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
-    }
-  }
-
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
-
-  return FAILURE;
-}
-
-void RtApiCore :: closeStream( void )
-{
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiCore::closeStream(): no open stream to close!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-    if ( stream_.state == STREAM_RUNNING )
-      AudioDeviceStop( handle->id[0], callbackHandler );
-#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
-    AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
-#else
-    // deprecated in favor of AudioDeviceDestroyIOProcID()
-    AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
-#endif
-  }
-
-  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
-    if ( stream_.state == STREAM_RUNNING )
-      AudioDeviceStop( handle->id[1], callbackHandler );
-#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
-    AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
-#else
-    // deprecated in favor of AudioDeviceDestroyIOProcID()
-    AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
-#endif
-  }
-
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
-    }
-  }
-
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
-
-  // Destroy pthread condition variable.
-  pthread_cond_destroy( &handle->condition );
-  delete handle;
-  stream_.apiHandle = 0;
-
-  stream_.mode = UNINITIALIZED;
-  stream_.state = STREAM_CLOSED;
-}
-
-void RtApiCore :: startStream( void )
-{
-  verifyStream();
-  if ( stream_.state == STREAM_RUNNING ) {
-    errorText_ = "RtApiCore::startStream(): the stream is already running!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-
-  OSStatus result = noErr;
-  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
-    result = AudioDeviceStart( handle->id[0], callbackHandler );
-    if ( result != noErr ) {
-      errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-  }
-
-  if ( stream_.mode == INPUT ||
-       ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
-
-    result = AudioDeviceStart( handle->id[1], callbackHandler );
-    if ( result != noErr ) {
-      errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-  }
-
-  handle->drainCounter = 0;
-  handle->internalDrain = false;
-  stream_.state = STREAM_RUNNING;
-
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  if ( result == noErr ) return;
-  error( RtError::SYSTEM_ERROR );
-}
-
-void RtApiCore :: stopStream( void )
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-
-  if ( stream_.state == STREAM_STOPPED ) {
-    MUTEX_UNLOCK( &stream_.mutex );
-    return;
-  }
-
-  OSStatus result = noErr;
-  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
-    if ( handle->drainCounter == 0 ) {
-      handle->drainCounter = 1;
-      pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
-    }
-
-    MUTEX_UNLOCK( &stream_.mutex );
-    result = AudioDeviceStop( handle->id[0], callbackHandler );
-    MUTEX_LOCK( &stream_.mutex );
-    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] ) ) {
-
-    MUTEX_UNLOCK( &stream_.mutex );
-    result = AudioDeviceStop( handle->id[1], callbackHandler );
-    MUTEX_LOCK( &stream_.mutex );
-    if ( result != noErr ) {
-      errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-  }
-
-  stream_.state = STREAM_STOPPED;
-
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  if ( result == noErr ) return;
-  error( RtError::SYSTEM_ERROR );
-}
-
-void RtApiCore :: abortStream( void )
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
-  handle->drainCounter = 1;
-
-  stopStream();
-}
-
-bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
-                                 const AudioBufferList *inBufferList,
-                                 const AudioBufferList *outBufferList )
-{
-  if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
-    error( RtError::WARNING );
-    return FAILURE;
-  }
-
-  CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
-  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
-
-  // Check if we were draining the stream and signal is finished.
-  if ( handle->drainCounter > 3 ) {
-    if ( handle->internalDrain == false )
-      pthread_cond_signal( &handle->condition );
-    else
-      stopStream();
-    return SUCCESS;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-
-  // The state might change while waiting on a mutex.
-  if ( stream_.state == STREAM_STOPPED ) {
-    MUTEX_UNLOCK( &stream_.mutex );
-    return SUCCESS;
-  }
-
-  AudioDeviceID outputDevice = handle->id[0];
-
-  // Invoke user callback to get fresh output data UNLESS we are
-  // draining stream or duplex mode AND the input/output devices are
-  // different AND this function is called for the input device.
-  if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
-    RtAudioCallback callback = (RtAudioCallback) info->callback;
-    double streamTime = getStreamTime();
-    RtAudioStreamStatus status = 0;
-    if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
-      status |= RTAUDIO_OUTPUT_UNDERFLOW;
-      handle->xrun[0] = false;
-    }
-    if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
-      status |= RTAUDIO_INPUT_OVERFLOW;
-      handle->xrun[1] = false;
-    }
-
-    handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
-                                     stream_.bufferSize, streamTime, status, info->userData );
-    if ( handle->drainCounter == 2 ) {
-      MUTEX_UNLOCK( &stream_.mutex );
-      abortStream();
-      return SUCCESS;
-    }
-    else if ( handle->drainCounter == 1 )
-      handle->internalDrain = true;
-  }
-
-  if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
-
-    if ( handle->drainCounter > 1 ) { // write zeros to the output stream
-
-      if ( handle->nStreams[0] == 1 ) {
-        memset( outBufferList->mBuffers[handle->iStream[0]].mData,
-                0,
-                outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
-      }
-      else { // fill multiple streams with zeros
-        for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
-          memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
-                  0,
-                  outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
-        }
-      }
-    }
-    else if ( handle->nStreams[0] == 1 ) {
-      if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
-        convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
-                       stream_.userBuffer[0], stream_.convertInfo[0] );
-      }
-      else { // copy from user buffer
-        memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
-                stream_.userBuffer[0],
-                outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
-      }
-    }
-    else { // fill multiple streams
-      Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
-      if ( stream_.doConvertBuffer[0] ) {
-        convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
-        inBuffer = (Float32 *) stream_.deviceBuffer;
-      }
-
-      if ( stream_.deviceInterleaved[0] == false ) { // mono mode
-        UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
-        for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
-          memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
-                  (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
-        }
-      }
-      else { // fill multiple multi-channel streams with interleaved data
-        UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
-        Float32 *out, *in;
-
-        bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
-        UInt32 inChannels = stream_.nUserChannels[0];
-        if ( stream_.doConvertBuffer[0] ) {
-          inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
-          inChannels = stream_.nDeviceChannels[0];
-        }
-
-        if ( inInterleaved ) inOffset = 1;
-        else inOffset = stream_.bufferSize;
-
-        channelsLeft = inChannels;
-        for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
-          in = inBuffer;
-          out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
-          streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;
-
-          outJump = 0;
-          // Account for possible channel offset in first stream
-          if ( i == 0 && stream_.channelOffset[0] > 0 ) {
-            streamChannels -= stream_.channelOffset[0];
-            outJump = stream_.channelOffset[0];
-            out += outJump;
-          }
-
-          // Account for possible unfilled channels at end of the last stream
-          if ( streamChannels > channelsLeft ) {
-            outJump = streamChannels - channelsLeft;
-            streamChannels = channelsLeft;
-          }
-
-          // Determine input buffer offsets and skips
-          if ( inInterleaved ) {
-            inJump = inChannels;
-            in += inChannels - channelsLeft;
-          }
-          else {
-            inJump = 1;
-            in += (inChannels - channelsLeft) * inOffset;
-          }
-
-          for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
-            for ( unsigned int j=0; j<streamChannels; j++ ) {
-              *out++ = in[j*inOffset];
-            }
-            out += outJump;
-            in += inJump;
-          }
-          channelsLeft -= streamChannels;
-        }
-      }
-    }
-
-    if ( handle->drainCounter ) {
-      handle->drainCounter++;
-      goto unlock;
-    }
-  }
-
-  AudioDeviceID inputDevice;
-  inputDevice = handle->id[1];
-  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
-
-    if ( handle->nStreams[1] == 1 ) {
-      if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer
-        convertBuffer( stream_.userBuffer[1],
-                       (char *) inBufferList->mBuffers[handle->iStream[1]].mData,
-                       stream_.convertInfo[1] );
-      }
-      else { // copy to user buffer
-        memcpy( stream_.userBuffer[1],
-                inBufferList->mBuffers[handle->iStream[1]].mData,
-                inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
-      }
-    }
-    else { // read from multiple streams
-      Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];
-      if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;
-
-      if ( stream_.deviceInterleaved[1] == false ) { // mono mode
-        UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
-        for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
-          memcpy( (void *)&outBuffer[i*stream_.bufferSize],
-                  inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
-        }
-      }
-      else { // read from multiple multi-channel streams
-        UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;
-        Float32 *out, *in;
-
-        bool outInterleaved = ( stream_.userInterleaved ) ? true : false;
-        UInt32 outChannels = stream_.nUserChannels[1];
-        if ( stream_.doConvertBuffer[1] ) {
-          outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
-          outChannels = stream_.nDeviceChannels[1];
-        }
-
-        if ( outInterleaved ) outOffset = 1;
-        else outOffset = stream_.bufferSize;
-
-        channelsLeft = outChannels;
-        for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {
-          out = outBuffer;
-          in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;
-          streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;
-
-          inJump = 0;
-          // Account for possible channel offset in first stream
-          if ( i == 0 && stream_.channelOffset[1] > 0 ) {
-            streamChannels -= stream_.channelOffset[1];
-            inJump = stream_.channelOffset[1];
-            in += inJump;
-          }
-
-          // Account for possible unread channels at end of the last stream
-          if ( streamChannels > channelsLeft ) {
-            inJump = streamChannels - channelsLeft;
-            streamChannels = channelsLeft;
-          }
-
-          // Determine output buffer offsets and skips
-          if ( outInterleaved ) {
-            outJump = outChannels;
-            out += outChannels - channelsLeft;
-          }
-          else {
-            outJump = 1;
-            out += (outChannels - channelsLeft) * outOffset;
-          }
-
-          for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
-            for ( unsigned int j=0; j<streamChannels; j++ ) {
-              out[j*outOffset] = *in++;
-            }
-            out += outJump;
-            in += inJump;
-          }
-          channelsLeft -= streamChannels;
-        }
-      }
-      
-      if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
-        convertBuffer( stream_.userBuffer[1],
-                       stream_.deviceBuffer,
-                       stream_.convertInfo[1] );
-      }
-    }
-  }
-
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  RtApi::tickStreamTime();
-  return SUCCESS;
-}
-
-const char* RtApiCore :: getErrorCode( OSStatus code )
-{
-  switch( code ) {
-
-  case kAudioHardwareNotRunningError:
-    return "kAudioHardwareNotRunningError";
-
-  case kAudioHardwareUnspecifiedError:
-    return "kAudioHardwareUnspecifiedError";
-
-  case kAudioHardwareUnknownPropertyError:
-    return "kAudioHardwareUnknownPropertyError";
-
-  case kAudioHardwareBadPropertySizeError:
-    return "kAudioHardwareBadPropertySizeError";
-
-  case kAudioHardwareIllegalOperationError:
-    return "kAudioHardwareIllegalOperationError";
-
-  case kAudioHardwareBadObjectError:
-    return "kAudioHardwareBadObjectError";
-
-  case kAudioHardwareBadDeviceError:
-    return "kAudioHardwareBadDeviceError";
-
-  case kAudioHardwareBadStreamError:
-    return "kAudioHardwareBadStreamError";
-
-  case kAudioHardwareUnsupportedOperationError:
-    return "kAudioHardwareUnsupportedOperationError";
-
-  case kAudioDeviceUnsupportedFormatError:
-    return "kAudioDeviceUnsupportedFormatError";
-
-  case kAudioDevicePermissionsError:
-    return "kAudioDevicePermissionsError";
-
-  default:
-    return "CoreAudio unknown error";
-  }
-}
-
-  //******************** End of __MACOSX_CORE__ *********************//
-#endif
-
-#if defined(__UNIX_JACK__)
-
-// JACK is a low-latency audio server, originally written for the
-// GNU/Linux operating system and now also ported to OS-X. It can
-// connect a number of different applications to an audio device, as
-// well as allowing them to share audio between themselves.
-//
-// When using JACK with RtAudio, "devices" refer to JACK clients that
-// have ports connected to the server.  The JACK server is typically
-// started in a terminal as follows:
-//
-// .jackd -d alsa -d hw:0
-//
-// or through an interface program such as qjackctl.  Many of the
-// parameters normally set for a stream are fixed by the JACK server
-// and can be specified when the JACK server is started.  In
-// particular,
-//
-// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
-//
-// specifies a sample rate of 44100 Hz, a buffer size of 512 sample
-// frames, and number of buffers = 4.  Once the server is running, it
-// is not possible to override these values.  If the values are not
-// specified in the command-line, the JACK server uses default values.
-//
-// The JACK server does not have to be running when an instance of
-// RtApiJack is created, though the function getDeviceCount() will
-// report 0 devices found until JACK has been started.  When no
-// devices are available (i.e., the JACK server is not running), a
-// stream cannot be opened.
-
-#include <jack/jack.h>
-#include <unistd.h>
-#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; }
-};
-
-ThreadHandle threadId;
-void jackSilentError( const char * ) {};
-
-RtApiJack :: RtApiJack()
-{
-  // Nothing to do here.
-#if !defined(__RTAUDIO_DEBUG__)
-  // Turn off Jack's internal error reporting.
-  jack_set_error_function( &jackSilentError );
-#endif
-}
-
-RtApiJack :: ~RtApiJack()
-{
-  if ( stream_.state != STREAM_CLOSED ) closeStream();
-}
-
-unsigned int RtApiJack :: getDeviceCount( void )
-{
-  // See if we can become a jack client.
-  jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
-  jack_status_t *status = NULL;
-  jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
-  if ( client == 0 ) return 0;
-
-  const char **ports;
-  std::string port, previousPort;
-  unsigned int nChannels = 0, nDevices = 0;
-  ports = jack_get_ports( client, NULL, NULL, 0 );
-  if ( ports ) {
-    // Parse the port names up to the first colon (:).
-    size_t iColon = 0;
-    do {
-      port = (char *) ports[ nChannels ];
-      iColon = port.find(":");
-      if ( iColon != std::string::npos ) {
-        port = port.substr( 0, iColon + 1 );
-        if ( port != previousPort ) {
-          nDevices++;
-          previousPort = port;
-        }
-      }
-    } while ( ports[++nChannels] );
-    free( ports );
-  }
-
-  jack_client_close( client );
-  return nDevices;
-}
-
-RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
-{
-  RtAudio::DeviceInfo info;
-  info.probed = false;
-
-  jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption
-  jack_status_t *status = NULL;
-  jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
-  if ( client == 0 ) {
-    errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
-    error( RtError::WARNING );
-    return info;
-  }
-
-  const char **ports;
-  std::string port, previousPort;
-  unsigned int nPorts = 0, nDevices = 0;
-  ports = jack_get_ports( client, NULL, NULL, 0 );
-  if ( ports ) {
-    // Parse the port names up to the first colon (:).
-    size_t iColon = 0;
-    do {
-      port = (char *) ports[ nPorts ];
-      iColon = port.find(":");
-      if ( iColon != std::string::npos ) {
-        port = port.substr( 0, iColon );
-        if ( port != previousPort ) {
-          if ( nDevices == device ) info.name = port;
-          nDevices++;
-          previousPort = port;
-        }
-      }
-    } while ( ports[++nPorts] );
-    free( ports );
-  }
-
-  if ( device >= nDevices ) {
-    errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
-    error( RtError::INVALID_USE );
-  }
-
-  // Get the current jack server sample rate.
-  info.sampleRates.clear();
-  info.sampleRates.push_back( jack_get_sample_rate( client ) );
-
-  // Count the available ports containing the client name as device
-  // channels.  Jack "input ports" equal RtAudio output channels.
-  unsigned int nChannels = 0;
-  ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
-  if ( ports ) {
-    while ( ports[ nChannels ] ) nChannels++;
-    free( ports );
-    info.outputChannels = nChannels;
-  }
-
-  // Jack "output ports" equal RtAudio input channels.
-  nChannels = 0;
-  ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
-  if ( ports ) {
-    while ( ports[ nChannels ] ) nChannels++;
-    free( ports );
-    info.inputChannels = nChannels;
-  }
-
-  if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
-    jack_client_close(client);
-    errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
-    error( RtError::WARNING );
-    return info;
-  }
-
-  // If device opens for both playback and capture, we determine the channels.
-  if ( info.outputChannels > 0 && info.inputChannels > 0 )
-    info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
-
-  // Jack always uses 32-bit floats.
-  info.nativeFormats = RTAUDIO_FLOAT32;
-
-  // Jack doesn't provide default devices so we'll use the first available one.
-  if ( device == 0 && info.outputChannels > 0 )
-    info.isDefaultOutput = true;
-  if ( device == 0 && info.inputChannels > 0 )
-    info.isDefaultInput = true;
-
-  jack_client_close(client);
-  info.probed = true;
-  return info;
-}
-
-int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
-{
-  CallbackInfo *info = (CallbackInfo *) infoPointer;
-
-  RtApiJack *object = (RtApiJack *) info->object;
-  if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
-
-  return 0;
-}
-
-// 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.
-extern "C" void *jackCloseStream( void *ptr )
-{
-  CallbackInfo *info = (CallbackInfo *) ptr;
-  RtApiJack *object = (RtApiJack *) info->object;
-
-  object->closeStream();
-
-  pthread_exit( NULL );
-}
-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;
-
-  pthread_create( &threadId, NULL, jackCloseStream, info );
-  std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
-}
-
-int jackXrun( void *infoPointer )
-{
-  JackHandle *handle = (JackHandle *) infoPointer;
-
-  if ( handle->ports[0] ) handle->xrun[0] = true;
-  if ( handle->ports[1] ) handle->xrun[1] = true;
-
-  return 0;
-}
-
-bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-                                   unsigned int firstChannel, unsigned int sampleRate,
-                                   RtAudioFormat format, unsigned int *bufferSize,
-                                   RtAudio::StreamOptions *options )
-{
-  JackHandle *handle = (JackHandle *) stream_.apiHandle;
-
-  // Look for jack server and try to become a client (only do once per stream).
-  jack_client_t *client = 0;
-  if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
-    jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
-    jack_status_t *status = NULL;
-    if ( options && !options->streamName.empty() )
-      client = jack_client_open( options->streamName.c_str(), jackoptions, status );
-    else
-      client = jack_client_open( "RtApiJack", jackoptions, status );
-    if ( client == 0 ) {
-      errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
-      error( RtError::WARNING );
-      return FAILURE;
-    }
-  }
-  else {
-    // The handle must have been created on an earlier pass.
-    client = handle->client;
-  }
-
-  const char **ports;
-  std::string port, previousPort, deviceName;
-  unsigned int nPorts = 0, nDevices = 0;
-  ports = jack_get_ports( client, NULL, NULL, 0 );
-  if ( ports ) {
-    // Parse the port names up to the first colon (:).
-    size_t iColon = 0;
-    do {
-      port = (char *) ports[ nPorts ];
-      iColon = port.find(":");
-      if ( iColon != std::string::npos ) {
-        port = port.substr( 0, iColon );
-        if ( port != previousPort ) {
-          if ( nDevices == device ) deviceName = port;
-          nDevices++;
-          previousPort = port;
-        }
-      }
-    } while ( ports[++nPorts] );
-    free( ports );
-  }
-
-  if ( device >= nDevices ) {
-    errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
-    return FAILURE;
-  }
-
-  // Count the available ports containing the client name as device
-  // channels.  Jack "input ports" equal RtAudio output channels.
-  unsigned int nChannels = 0;
-  unsigned long flag = JackPortIsInput;
-  if ( mode == INPUT ) flag = JackPortIsOutput;
-  ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
-  if ( ports ) {
-    while ( ports[ nChannels ] ) nChannels++;
-    free( ports );
-  }
-
-  // Compare the jack ports for specified client to the requested number of channels.
-  if ( nChannels < (channels + firstChannel) ) {
-    errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Check the jack server sample rate.
-  unsigned int jackRate = jack_get_sample_rate( client );
-  if ( sampleRate != jackRate ) {
-    jack_client_close( client );
-    errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-  stream_.sampleRate = jackRate;
-
-  // Get the latency of the JACK port.
-  ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
-  if ( ports[ firstChannel ] )
-    stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
-  free( ports );
-
-  // The jack server always uses 32-bit floating-point data.
-  stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
-  stream_.userFormat = format;
-
-  if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
-  else stream_.userInterleaved = true;
-
-  // Jack always uses non-interleaved buffers.
-  stream_.deviceInterleaved[mode] = false;
-
-  // Jack always provides host byte-ordered data.
-  stream_.doByteSwap[mode] = false;
-
-  // Get the buffer size.  The buffer size and number of buffers
-  // (periods) is set when the jack server is started.
-  stream_.bufferSize = (int) jack_get_buffer_size( client );
-  *bufferSize = stream_.bufferSize;
-
-  stream_.nDeviceChannels[mode] = channels;
-  stream_.nUserChannels[mode] = channels;
-
-  // Set flags for buffer conversion.
-  stream_.doConvertBuffer[mode] = false;
-  if ( stream_.userFormat != stream_.deviceFormat[mode] )
-    stream_.doConvertBuffer[mode] = true;
-  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
-       stream_.nUserChannels[mode] > 1 )
-    stream_.doConvertBuffer[mode] = true;
-
-  // Allocate our JackHandle structure for the stream.
-  if ( handle == 0 ) {
-    try {
-      handle = new JackHandle;
-    }
-    catch ( std::bad_alloc& ) {
-      errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
-      goto error;
-    }
-
-    if ( pthread_cond_init(&handle->condition, NULL) ) {
-      errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
-      goto error;
-    }
-    stream_.apiHandle = (void *) handle;
-    handle->client = client;
-  }
-  handle->deviceName[mode] = deviceName;
-
-  // Allocate necessary internal buffers.
-  unsigned long bufferBytes;
-  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
-  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
-  if ( stream_.userBuffer[mode] == NULL ) {
-    errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
-    goto error;
-  }
-
-  if ( stream_.doConvertBuffer[mode] ) {
-
-    bool makeBuffer = true;
-    if ( mode == OUTPUT )
-      bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
-    else { // mode == INPUT
-      bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
-      if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
-        unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
-        if ( bufferBytes < bytesOut ) makeBuffer = false;
-      }
-    }
-
-    if ( makeBuffer ) {
-      bufferBytes *= *bufferSize;
-      if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
-      stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
-      if ( stream_.deviceBuffer == NULL ) {
-        errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
-        goto error;
-      }
-    }
-  }
-
-  // Allocate memory for the Jack ports (channels) identifiers.
-  handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
-  if ( handle->ports[mode] == NULL )  {
-    errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
-    goto error;
-  }
-
-  stream_.device[mode] = device;
-  stream_.channelOffset[mode] = firstChannel;
-  stream_.state = STREAM_STOPPED;
-  stream_.callbackInfo.object = (void *) this;
-
-  if ( stream_.mode == OUTPUT && mode == INPUT )
-    // We had already set up the stream for output.
-    stream_.mode = DUPLEX;
-  else {
-    stream_.mode = mode;
-    jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
-    jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );
-    jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
-  }
-
-  // Register our ports.
-  char label[64];
-  if ( mode == OUTPUT ) {
-    for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
-      snprintf( label, 64, "outport %d", i );
-      handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
-                                                JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
-    }
-  }
-  else {
-    for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
-      snprintf( label, 64, "inport %d", i );
-      handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
-                                                JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
-    }
-  }
-
-  // Setup the buffer conversion information structure.  We don't use
-  // buffers to do channel offsets, so we override that parameter
-  // here.
-  if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
-
-  return SUCCESS;
-
- error:
-  if ( handle ) {
-    pthread_cond_destroy( &handle->condition );
-    jack_client_close( handle->client );
-
-    if ( handle->ports[0] ) free( handle->ports[0] );
-    if ( handle->ports[1] ) free( handle->ports[1] );
-
-    delete handle;
-    stream_.apiHandle = 0;
-  }
-
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
-    }
-  }
-
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
-
-  return FAILURE;
-}
-
-void RtApiJack :: closeStream( void )
-{
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiJack::closeStream(): no open stream to close!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  JackHandle *handle = (JackHandle *) stream_.apiHandle;
-  if ( handle ) {
-
-    if ( stream_.state == STREAM_RUNNING )
-      jack_deactivate( handle->client );
-
-    jack_client_close( handle->client );
-  }
-
-  if ( handle ) {
-    if ( handle->ports[0] ) free( handle->ports[0] );
-    if ( handle->ports[1] ) free( handle->ports[1] );
-    pthread_cond_destroy( &handle->condition );
-    delete handle;
-    stream_.apiHandle = 0;
-  }
-
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
-    }
-  }
-
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
-
-  stream_.mode = UNINITIALIZED;
-  stream_.state = STREAM_CLOSED;
-}
-
-void RtApiJack :: startStream( void )
-{
-  verifyStream();
-  if ( stream_.state == STREAM_RUNNING ) {
-    errorText_ = "RtApiJack::startStream(): the stream is already running!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  MUTEX_LOCK(&stream_.mutex);
-
-  JackHandle *handle = (JackHandle *) stream_.apiHandle;
-  int result = jack_activate( handle->client );
-  if ( result ) {
-    errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
-    goto unlock;
-  }
-
-  const char **ports;
-
-  // Get the list of available ports.
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-    result = 1;
-    ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);
-    if ( ports == NULL) {
-      errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
-      goto unlock;
-    }
-
-    // Now make the port connections.  Since RtAudio wasn't designed to
-    // allow the user to select particular channels of a device, we'll
-    // just open the first "nChannels" ports with offset.
-    for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
-      result = 1;
-      if ( ports[ stream_.channelOffset[0] + i ] )
-        result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
-      if ( result ) {
-        free( ports );
-        errorText_ = "RtApiJack::startStream(): error connecting output ports!";
-        goto unlock;
-      }
-    }
-    free(ports);
-  }
-
-  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-    result = 1;
-    ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );
-    if ( ports == NULL) {
-      errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
-      goto unlock;
-    }
-
-    // Now make the port connections.  See note above.
-    for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
-      result = 1;
-      if ( ports[ stream_.channelOffset[1] + i ] )
-        result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
-      if ( result ) {
-        free( ports );
-        errorText_ = "RtApiJack::startStream(): error connecting input ports!";
-        goto unlock;
-      }
-    }
-    free(ports);
-  }
-
-  handle->drainCounter = 0;
-  handle->internalDrain = false;
-  stream_.state = STREAM_RUNNING;
-
- unlock:
-  MUTEX_UNLOCK(&stream_.mutex);
-
-  if ( result == 0 ) return;
-  error( RtError::SYSTEM_ERROR );
-}
-
-void RtApiJack :: stopStream( void )
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-
-  if ( stream_.state == STREAM_STOPPED ) {
-    MUTEX_UNLOCK( &stream_.mutex );
-    return;
-  }
-
-  JackHandle *handle = (JackHandle *) stream_.apiHandle;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
-    if ( handle->drainCounter == 0 ) {
-      handle->drainCounter = 1;
-      pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
-    }
-  }
-
-  jack_deactivate( handle->client );
-  stream_.state = STREAM_STOPPED;
-
-  MUTEX_UNLOCK( &stream_.mutex );
-}
-
-void RtApiJack :: abortStream( void )
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  JackHandle *handle = (JackHandle *) stream_.apiHandle;
-  handle->drainCounter = 1;
-
-  stopStream();
-}
-
-// This function will be called by a spawned thread when the user
-// callback function signals that the stream should be stopped or
-// aborted.  It is necessary to handle it this way because the
-// callbackEvent() function must return before the jack_deactivate()
-// function will return.
-extern "C" void *jackStopStream( void *ptr )
-{
-  CallbackInfo *info = (CallbackInfo *) ptr;
-  RtApiJack *object = (RtApiJack *) info->object;
-
-  object->stopStream();
-
-  pthread_exit( NULL );
-}
-
-bool RtApiJack :: callbackEvent( unsigned long nframes )
-{
-  if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
-    error( RtError::WARNING );
-    return FAILURE;
-  }
-  if ( stream_.bufferSize != nframes ) {
-    errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
-    error( RtError::WARNING );
-    return FAILURE;
-  }
-
-  CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
-  JackHandle *handle = (JackHandle *) stream_.apiHandle;
-
-  // Check if we were draining the stream and signal is finished.
-  if ( handle->drainCounter > 3 ) {
-    if ( handle->internalDrain == true ) {
-      pthread_create( &threadId, NULL, jackStopStream, info );
-    }
-    else
-      pthread_cond_signal( &handle->condition );
-    return SUCCESS;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-
-  // The state might change while waiting on a mutex.
-  if ( stream_.state == STREAM_STOPPED ) {
-    MUTEX_UNLOCK( &stream_.mutex );
-    return SUCCESS;
-  }
-
-  // Invoke user callback first, to get fresh output data.
-  if ( handle->drainCounter == 0 ) {
-    RtAudioCallback callback = (RtAudioCallback) info->callback;
-    double streamTime = getStreamTime();
-    RtAudioStreamStatus status = 0;
-    if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
-      status |= RTAUDIO_OUTPUT_UNDERFLOW;
-      handle->xrun[0] = false;
-    }
-    if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
-      status |= RTAUDIO_INPUT_OVERFLOW;
-      handle->xrun[1] = false;
-    }
-    handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
-                                     stream_.bufferSize, streamTime, status, info->userData );
-    if ( handle->drainCounter == 2 ) {
-      MUTEX_UNLOCK( &stream_.mutex );
-      ThreadHandle id;
-      pthread_create( &id, NULL, jackStopStream, info );
-      return SUCCESS;
-    }
-    else if ( handle->drainCounter == 1 )
-      handle->internalDrain = true;
-  }
-
-  jack_default_audio_sample_t *jackbuffer;
-  unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
-    if ( handle->drainCounter > 0 ) { // write zeros to the output stream
-
-      for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
-        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
-        memset( jackbuffer, 0, bufferBytes );
-      }
-
-    }
-    else if ( stream_.doConvertBuffer[0] ) {
-
-      convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
-
-      for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
-        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
-        memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
-      }
-    }
-    else { // no buffer conversion
-      for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
-        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
-        memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
-      }
-    }
-
-    if ( handle->drainCounter ) {
-      handle->drainCounter++;
-      goto unlock;
-    }
-  }
-
-  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-
-    if ( stream_.doConvertBuffer[1] ) {
-      for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
-        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
-        memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
-      }
-      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
-    }
-    else { // no buffer conversion
-      for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
-        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
-        memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
-      }
-    }
-  }
-
- unlock:
-  MUTEX_UNLOCK(&stream_.mutex);
-
-  RtApi::tickStreamTime();
-  return SUCCESS;
-}
-  //******************** End of __UNIX_JACK__ *********************//
-#endif
-
-#if defined(__WINDOWS_ASIO__) // ASIO API on Windows
-
-// The ASIO API is designed around a callback scheme, so this
-// implementation is similar to that used for OS-X CoreAudio and Linux
-// Jack.  The primary constraint with ASIO is that it only allows
-// access to a single driver at a time.  Thus, it is not possible to
-// have more than one simultaneous RtAudio stream.
-//
-// This implementation also requires a number of external ASIO files
-// and a few global variables.  The ASIO callback scheme does not
-// allow for the passing of user data, so we must create a global
-// pointer to our callbackInfo structure.
-//
-// On unix systems, we make use of a pthread condition variable.
-// Since there is no equivalent in Windows, I hacked something based
-// on information found in
-// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
-
-#include "asiosys.h"
-#include "asio.h"
-#include "iasiothiscallresolver.h"
-#include "asiodrivers.h"
-#include <cmath>
-
-AsioDrivers drivers;
-ASIOCallbacks asioCallbacks;
-ASIODriverInfo driverInfo;
-CallbackInfo *asioCallbackInfo;
-bool asioXRun;
-
-struct AsioHandle {
-  int drainCounter;       // Tracks callback counts when draining
-  bool internalDrain;     // Indicates if stop is initiated from callback or not.
-  ASIOBufferInfo *bufferInfos;
-  HANDLE condition;
-
-  AsioHandle()
-    :drainCounter(0), internalDrain(false), bufferInfos(0) {}
-};
-
-// Function declarations (definitions at end of section)
-static const char* getAsioErrorString( ASIOError result );
-void sampleRateChanged( ASIOSampleRate sRate );
-long asioMessages( long selector, long value, void* message, double* opt );
-
-RtApiAsio :: RtApiAsio()
-{
-  // ASIO cannot run on a multi-threaded appartment. You can call
-  // CoInitialize beforehand, but it must be for appartment threading
-  // (in which case, CoInitilialize will return S_FALSE here).
-  coInitialized_ = false;
-  HRESULT hr = CoInitialize( NULL ); 
-  if ( FAILED(hr) ) {
-    errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
-    error( RtError::WARNING );
-  }
-  coInitialized_ = true;
-
-  drivers.removeCurrentDriver();
-  driverInfo.asioVersion = 2;
-
-  // See note in DirectSound implementation about GetDesktopWindow().
-  driverInfo.sysRef = GetForegroundWindow();
-}
-
-RtApiAsio :: ~RtApiAsio()
-{
-  if ( stream_.state != STREAM_CLOSED ) closeStream();
-  if ( coInitialized_ ) CoUninitialize();
-}
-
-unsigned int RtApiAsio :: getDeviceCount( void )
-{
-  return (unsigned int) drivers.asioGetNumDev();
-}
-
-RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
-{
-  RtAudio::DeviceInfo info;
-  info.probed = false;
-
-  // Get device ID
-  unsigned int nDevices = getDeviceCount();
-  if ( nDevices == 0 ) {
-    errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
-    error( RtError::INVALID_USE );
-  }
-
-  if ( device >= nDevices ) {
-    errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
-    error( RtError::INVALID_USE );
-  }
-
-  // If a stream is already open, we cannot probe other devices.  Thus, use the saved results.
-  if ( stream_.state != STREAM_CLOSED ) {
-    if ( device >= devices_.size() ) {
-      errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
-      error( RtError::WARNING );
-      return info;
-    }
-    return devices_[ device ];
-  }
-
-  char driverName[32];
-  ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
-  if ( result != ASE_OK ) {
-    errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
-
-  info.name = driverName;
-
-  if ( !drivers.loadDriver( driverName ) ) {
-    errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
-
-  result = ASIOInit( &driverInfo );
-  if ( result != ASE_OK ) {
-    errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
-
-  // Determine the device channel information.
-  long inputChannels, outputChannels;
-  result = ASIOGetChannels( &inputChannels, &outputChannels );
-  if ( result != ASE_OK ) {
-    drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
-
-  info.outputChannels = outputChannels;
-  info.inputChannels = inputChannels;
-  if ( info.outputChannels > 0 && info.inputChannels > 0 )
-    info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
-
-  // Determine the supported sample rates.
-  info.sampleRates.clear();
-  for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
-    result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
-    if ( result == ASE_OK )
-      info.sampleRates.push_back( SAMPLE_RATES[i] );
-  }
-
-  // Determine supported data types ... just check first channel and assume rest are the same.
-  ASIOChannelInfo channelInfo;
-  channelInfo.channel = 0;
-  channelInfo.isInput = true;
-  if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
-  result = ASIOGetChannelInfo( &channelInfo );
-  if ( result != ASE_OK ) {
-    drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
-
-  info.nativeFormats = 0;
-  if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
-    info.nativeFormats |= RTAUDIO_SINT16;
-  else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
-    info.nativeFormats |= RTAUDIO_SINT32;
-  else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
-    info.nativeFormats |= RTAUDIO_FLOAT32;
-  else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
-    info.nativeFormats |= RTAUDIO_FLOAT64;
-
-  if ( 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;
-}
-
-void bufferSwitch( long index, ASIOBool processNow )
-{
-  RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
-  object->callbackEvent( index );
-}
-
-void RtApiAsio :: saveDeviceInfo( void )
-{
-  devices_.clear();
-
-  unsigned int nDevices = getDeviceCount();
-  devices_.resize( nDevices );
-  for ( unsigned int i=0; i<nDevices; i++ )
-    devices_[i] = getDeviceInfo( i );
-}
-
-bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-                                   unsigned int firstChannel, unsigned int sampleRate,
-                                   RtAudioFormat format, unsigned int *bufferSize,
-                                   RtAudio::StreamOptions *options )
-{
-  // For ASIO, a duplex stream MUST use the same driver.
-  if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) {
-    errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
-    return FAILURE;
-  }
-
-  char driverName[32];
-  ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
-  if ( result != ASE_OK ) {
-    errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // The getDeviceInfo() function will not work when a stream is open
-  // because ASIO does not allow multiple devices to run at the same
-  // time.  Thus, we'll probe the system before opening a stream and
-  // save the results for use by getDeviceInfo().
-  this->saveDeviceInfo();
-
-  // Only load the driver once for duplex stream.
-  if ( mode != INPUT || stream_.mode != OUTPUT ) {
-    if ( !drivers.loadDriver( driverName ) ) {
-      errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
-      errorText_ = errorStream_.str();
-      return FAILURE;
-    }
-
-    result = ASIOInit( &driverInfo );
-    if ( result != ASE_OK ) {
-      errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
-      errorText_ = errorStream_.str();
-      return FAILURE;
-    }
-  }
-
-  // Check the device channel count.
-  long inputChannels, outputChannels;
-  result = ASIOGetChannels( &inputChannels, &outputChannels );
-  if ( result != ASE_OK ) {
-    drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
-       ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
-    drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-  stream_.nDeviceChannels[mode] = channels;
-  stream_.nUserChannels[mode] = channels;
-  stream_.channelOffset[mode] = firstChannel;
-
-  // Verify the sample rate is supported.
-  result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
-  if ( result != ASE_OK ) {
-    drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Get the current sample rate
-  ASIOSampleRate currentRate;
-  result = ASIOGetSampleRate( &currentRate );
-  if ( result != ASE_OK ) {
-    drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Set the sample rate only if necessary
-  if ( currentRate != sampleRate ) {
-    result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
-    if ( result != ASE_OK ) {
-      drivers.removeCurrentDriver();
-      errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
-      errorText_ = errorStream_.str();
-      return FAILURE;
-    }
-  }
-
-  // Determine the driver data type.
-  ASIOChannelInfo channelInfo;
-  channelInfo.channel = 0;
-  if ( mode == OUTPUT ) channelInfo.isInput = false;
-  else channelInfo.isInput = true;
-  result = ASIOGetChannelInfo( &channelInfo );
-  if ( result != ASE_OK ) {
-    drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Assuming WINDOWS host is always little-endian.
-  stream_.doByteSwap[mode] = false;
-  stream_.userFormat = format;
-  stream_.deviceFormat[mode] = 0;
-  if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
-    stream_.deviceFormat[mode] = RTAUDIO_SINT16;
-    if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
-  }
-  else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
-    stream_.deviceFormat[mode] = RTAUDIO_SINT32;
-    if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
-  }
-  else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
-    stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
-    if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
-  }
-  else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
-    stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
-    if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
-  }
-
-  if ( stream_.deviceFormat[mode] == 0 ) {
-    drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Set the buffer size.  For a duplex stream, this will end up
-  // setting the buffer size based on the input constraints, which
-  // should be ok.
-  long minSize, maxSize, preferSize, granularity;
-  result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
-  if ( result != ASE_OK ) {
-    drivers.removeCurrentDriver();
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
-  else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
-  else if ( granularity == -1 ) {
-    // Make sure bufferSize is a power of two.
-    int log2_of_min_size = 0;
-    int log2_of_max_size = 0;
-
-    for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
-      if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
-      if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
-    }
-
-    long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
-    int min_delta_num = log2_of_min_size;
-
-    for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
-      long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
-      if (current_delta < min_delta) {
-        min_delta = current_delta;
-        min_delta_num = i;
-      }
-    }
-
-    *bufferSize = ( (unsigned int)1 << min_delta_num );
-    if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
-    else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
-  }
-  else if ( granularity != 0 ) {
-    // Set to an even multiple of granularity, rounding up.
-    *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
-  }
-
-  if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) {
-    drivers.removeCurrentDriver();
-    errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
-    return FAILURE;
-  }
-
-  stream_.bufferSize = *bufferSize;
-  stream_.nBuffers = 2;
-
-  if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
-  else stream_.userInterleaved = true;
-
-  // ASIO always uses non-interleaved buffers.
-  stream_.deviceInterleaved[mode] = false;
-
-  // Allocate, if necessary, our AsioHandle structure for the stream.
-  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-  if ( handle == 0 ) {
-    try {
-      handle = new AsioHandle;
-    }
-    catch ( std::bad_alloc& ) {
-      //if ( handle == NULL ) {    
-      drivers.removeCurrentDriver();
-      errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
-      return FAILURE;
-    }
-    handle->bufferInfos = 0;
-
-    // Create a manual-reset event.
-    handle->condition = CreateEvent( NULL,   // no security
-                                     TRUE,   // manual-reset
-                                     FALSE,  // non-signaled initially
-                                     NULL ); // unnamed
-    stream_.apiHandle = (void *) handle;
-  }
-
-  // Create the ASIO internal buffers.  Since RtAudio sets up input
-  // and output separately, we'll have to dispose of previously
-  // created output buffers for a duplex stream.
-  long inputLatency, outputLatency;
-  if ( mode == INPUT && stream_.mode == OUTPUT ) {
-    ASIODisposeBuffers();
-    if ( handle->bufferInfos ) free( handle->bufferInfos );
-  }
-
-  // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
-  bool buffersAllocated = false;
-  unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
-  handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
-  if ( handle->bufferInfos == NULL ) {
-    errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
-    errorText_ = errorStream_.str();
-    goto error;
-  }
-
-  ASIOBufferInfo *infos;
-  infos = handle->bufferInfos;
-  for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
-    infos->isInput = ASIOFalse;
-    infos->channelNum = i + stream_.channelOffset[0];
-    infos->buffers[0] = infos->buffers[1] = 0;
-  }
-  for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
-    infos->isInput = ASIOTrue;
-    infos->channelNum = i + stream_.channelOffset[1];
-    infos->buffers[0] = infos->buffers[1] = 0;
-  }
-
-  // Set up the ASIO callback structure and create the ASIO data buffers.
-  asioCallbacks.bufferSwitch = &bufferSwitch;
-  asioCallbacks.sampleRateDidChange = &sampleRateChanged;
-  asioCallbacks.asioMessage = &asioMessages;
-  asioCallbacks.bufferSwitchTimeInfo = NULL;
-  result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
-  if ( result != ASE_OK ) {
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
-    errorText_ = errorStream_.str();
-    goto error;
-  }
-  buffersAllocated = true;
-
-  // Set flags for buffer conversion.
-  stream_.doConvertBuffer[mode] = false;
-  if ( stream_.userFormat != stream_.deviceFormat[mode] )
-    stream_.doConvertBuffer[mode] = true;
-  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
-       stream_.nUserChannels[mode] > 1 )
-    stream_.doConvertBuffer[mode] = true;
-
-  // Allocate necessary internal buffers
-  unsigned long bufferBytes;
-  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
-  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
-  if ( stream_.userBuffer[mode] == NULL ) {
-    errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
-    goto error;
-  }
-
-  if ( stream_.doConvertBuffer[mode] ) {
-
-    bool makeBuffer = true;
-    bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
-    if ( mode == INPUT ) {
-      if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
-        unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
-        if ( bufferBytes <= bytesOut ) makeBuffer = false;
-      }
-    }
-
-    if ( makeBuffer ) {
-      bufferBytes *= *bufferSize;
-      if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
-      stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
-      if ( stream_.deviceBuffer == NULL ) {
-        errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
-        goto error;
-      }
-    }
-  }
-
-  stream_.sampleRate = sampleRate;
-  stream_.device[mode] = device;
-  stream_.state = STREAM_STOPPED;
-  asioCallbackInfo = &stream_.callbackInfo;
-  stream_.callbackInfo.object = (void *) this;
-  if ( stream_.mode == OUTPUT && mode == INPUT )
-    // We had already set up an output stream.
-    stream_.mode = DUPLEX;
-  else
-    stream_.mode = mode;
-
-  // Determine device latencies
-  result = ASIOGetLatencies( &inputLatency, &outputLatency );
-  if ( result != ASE_OK ) {
-    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING); // warn but don't fail
-  }
-  else {
-    stream_.latency[0] = outputLatency;
-    stream_.latency[1] = inputLatency;
-  }
-
-  // Setup the buffer conversion information structure.  We don't use
-  // buffers to do channel offsets, so we override that parameter
-  // here.
-  if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
-
-  return SUCCESS;
-
- error:
-  if ( buffersAllocated )
-    ASIODisposeBuffers();
-  drivers.removeCurrentDriver();
-
-  if ( handle ) {
-    CloseHandle( handle->condition );
-    if ( handle->bufferInfos )
-      free( handle->bufferInfos );
-    delete handle;
-    stream_.apiHandle = 0;
-  }
-
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
-    }
-  }
-
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
-
-  return FAILURE;
-}
-
-void RtApiAsio :: closeStream()
-{
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  if ( stream_.state == STREAM_RUNNING ) {
-    stream_.state = STREAM_STOPPED;
-    ASIOStop();
-  }
-  ASIODisposeBuffers();
-  drivers.removeCurrentDriver();
-
-  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-  if ( handle ) {
-    CloseHandle( handle->condition );
-    if ( handle->bufferInfos )
-      free( handle->bufferInfos );
-    delete handle;
-    stream_.apiHandle = 0;
-  }
-
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
-    }
-  }
-
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
-
-  stream_.mode = UNINITIALIZED;
-  stream_.state = STREAM_CLOSED;
-}
-
-void RtApiAsio :: startStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_RUNNING ) {
-    errorText_ = "RtApiAsio::startStream(): the stream is already running!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-
-  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-  ASIOError result = ASIOStart();
-  if ( result != ASE_OK ) {
-    errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
-    errorText_ = errorStream_.str();
-    goto unlock;
-  }
-
-  handle->drainCounter = 0;
-  handle->internalDrain = false;
-  stream_.state = STREAM_RUNNING;
-  asioXRun = false;
-
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  if ( result == ASE_OK ) return;
-  error( RtError::SYSTEM_ERROR );
-}
-
-void RtApiAsio :: stopStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-
-  if ( stream_.state == STREAM_STOPPED ) {
-    MUTEX_UNLOCK( &stream_.mutex );
-    return;
-  }
-
-  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-    if ( handle->drainCounter == 0 ) {
-      handle->drainCounter = 1;
-      MUTEX_UNLOCK( &stream_.mutex );
-      WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE );  // block until signaled
-      ResetEvent( handle->condition );
-      MUTEX_LOCK( &stream_.mutex );
-    }
-  }
-
-  ASIOError result = ASIOStop();
-  if ( result != ASE_OK ) {
-    errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
-    errorText_ = errorStream_.str();
-  }
-
-  stream_.state = STREAM_STOPPED;
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  if ( result == ASE_OK ) return;
-  error( RtError::SYSTEM_ERROR );
-}
-
-void RtApiAsio :: abortStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  // The following lines were commented-out because some behavior was
-  // noted where the device buffers need to be zeroed to avoid
-  // continuing sound, even when the device buffers are completely
-  // disposed.  So now, calling abort is the same as calling stop.
-  // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-  // handle->drainCounter = 1;
-  stopStream();
-}
-
-bool RtApiAsio :: callbackEvent( long bufferIndex )
-{
-  if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
-    error( RtError::WARNING );
-    return FAILURE;
-  }
-
-  CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
-  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
-
-  // Check if we were draining the stream and signal is finished.
-  if ( handle->drainCounter > 3 ) {
-    if ( handle->internalDrain == false )
-      SetEvent( handle->condition );
-    else
-      stopStream();
-    return SUCCESS;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-
-  // The state might change while waiting on a mutex.
-  if ( stream_.state == STREAM_STOPPED ) goto unlock;
-
-  // Invoke user callback to get fresh output data UNLESS we are
-  // draining stream.
-  if ( handle->drainCounter == 0 ) {
-    RtAudioCallback callback = (RtAudioCallback) info->callback;
-    double streamTime = getStreamTime();
-    RtAudioStreamStatus status = 0;
-    if ( stream_.mode != INPUT && asioXRun == true ) {
-      status |= RTAUDIO_OUTPUT_UNDERFLOW;
-      asioXRun = false;
-    }
-    if ( stream_.mode != OUTPUT && asioXRun == true ) {
-      status |= RTAUDIO_INPUT_OVERFLOW;
-      asioXRun = false;
-    }
-    handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
-                                     stream_.bufferSize, streamTime, status, info->userData );
-    if ( handle->drainCounter == 2 ) {
-      MUTEX_UNLOCK( &stream_.mutex );
-      abortStream();
-      return SUCCESS;
-    }
-    else if ( handle->drainCounter == 1 )
-      handle->internalDrain = true;
-  }
-
-  unsigned int nChannels, bufferBytes, i, j;
-  nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
-    bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
-
-    if ( handle->drainCounter > 1 ) { // write zeros to the output stream
-
-      for ( i=0, j=0; i<nChannels; i++ ) {
-        if ( handle->bufferInfos[i].isInput != ASIOTrue )
-          memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
-      }
-
-    }
-    else if ( stream_.doConvertBuffer[0] ) {
-
-      convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
-      if ( stream_.doByteSwap[0] )
-        byteSwapBuffer( stream_.deviceBuffer,
-                        stream_.bufferSize * stream_.nDeviceChannels[0],
-                        stream_.deviceFormat[0] );
-
-      for ( i=0, j=0; i<nChannels; i++ ) {
-        if ( handle->bufferInfos[i].isInput != ASIOTrue )
-          memcpy( handle->bufferInfos[i].buffers[bufferIndex],
-                  &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
-      }
-
-    }
-    else {
-
-      if ( stream_.doByteSwap[0] )
-        byteSwapBuffer( stream_.userBuffer[0],
-                        stream_.bufferSize * stream_.nUserChannels[0],
-                        stream_.userFormat );
-
-      for ( i=0, j=0; i<nChannels; i++ ) {
-        if ( handle->bufferInfos[i].isInput != ASIOTrue )
-          memcpy( handle->bufferInfos[i].buffers[bufferIndex],
-                  &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
-      }
-
-    }
-
-    if ( handle->drainCounter ) {
-      handle->drainCounter++;
-      goto unlock;
-    }
-  }
-
-  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-
-    bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
-
-    if (stream_.doConvertBuffer[1]) {
-
-      // Always interleave ASIO input data.
-      for ( i=0, j=0; i<nChannels; i++ ) {
-        if ( handle->bufferInfos[i].isInput == ASIOTrue )
-          memcpy( &stream_.deviceBuffer[j++*bufferBytes],
-                  handle->bufferInfos[i].buffers[bufferIndex],
-                  bufferBytes );
-      }
-
-      if ( stream_.doByteSwap[1] )
-        byteSwapBuffer( stream_.deviceBuffer,
-                        stream_.bufferSize * stream_.nDeviceChannels[1],
-                        stream_.deviceFormat[1] );
-      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
-
-    }
-    else {
-      for ( i=0, j=0; i<nChannels; i++ ) {
-        if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
-          memcpy( &stream_.userBuffer[1][bufferBytes*j++],
-                  handle->bufferInfos[i].buffers[bufferIndex],
-                  bufferBytes );
-        }
-      }
-
-      if ( stream_.doByteSwap[1] )
-        byteSwapBuffer( stream_.userBuffer[1],
-                        stream_.bufferSize * stream_.nUserChannels[1],
-                        stream_.userFormat );
-    }
-  }
-
- unlock:
-  // The following call was suggested by Malte Clasen.  While the API
-  // documentation indicates it should not be required, some device
-  // drivers apparently do not function correctly without it.
-  ASIOOutputReady();
-
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  RtApi::tickStreamTime();
-  return SUCCESS;
-}
-
-void sampleRateChanged( ASIOSampleRate sRate )
-{
-  // The ASIO documentation says that this usually only happens during
-  // external sync.  Audio processing is not stopped by the driver,
-  // actual sample rate might not have even changed, maybe only the
-  // sample rate status of an AES/EBU or S/PDIF digital input at the
-  // audio device.
-
-  RtApi *object = (RtApi *) asioCallbackInfo->object;
-  try {
-    object->stopStream();
-  }
-  catch ( RtError &exception ) {
-    std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
-    return;
-  }
-
-  std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
-}
-
-long asioMessages( long selector, long value, void* message, double* opt )
-{
-  long ret = 0;
-
-  switch( selector ) {
-  case kAsioSelectorSupported:
-    if ( value == kAsioResetRequest
-         || value == kAsioEngineVersion
-         || value == kAsioResyncRequest
-         || value == kAsioLatenciesChanged
-         // The following three were added for ASIO 2.0, you don't
-         // necessarily have to support them.
-         || value == kAsioSupportsTimeInfo
-         || value == kAsioSupportsTimeCode
-         || value == kAsioSupportsInputMonitor)
-      ret = 1L;
-    break;
-  case kAsioResetRequest:
-    // Defer the task and perform the reset of the driver during the
-    // next "safe" situation.  You cannot reset the driver right now,
-    // as this code is called from the driver.  Reset the driver is
-    // done by completely destruct is. I.e. ASIOStop(),
-    // ASIODisposeBuffers(), Destruction Afterwards you initialize the
-    // driver again.
-    std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
-    ret = 1L;
-    break;
-  case kAsioResyncRequest:
-    // This informs the application that the driver encountered some
-    // non-fatal data loss.  It is used for synchronization purposes
-    // of different media.  Added mainly to work around the Win16Mutex
-    // problems in Windows 95/98 with the Windows Multimedia system,
-    // which could lose data because the Mutex was held too long by
-    // another thread.  However a driver can issue it in other
-    // situations, too.
-    // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
-    asioXRun = true;
-    ret = 1L;
-    break;
-  case kAsioLatenciesChanged:
-    // This will inform the host application that the drivers were
-    // latencies changed.  Beware, it this does not mean that the
-    // buffer sizes have changed!  You might need to update internal
-    // delay data.
-    std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
-    ret = 1L;
-    break;
-  case kAsioEngineVersion:
-    // Return the supported ASIO version of the host application.  If
-    // a host application does not implement this selector, ASIO 1.0
-    // is assumed by the driver.
-    ret = 2L;
-    break;
-  case kAsioSupportsTimeInfo:
-    // Informs the driver whether the
-    // asioCallbacks.bufferSwitchTimeInfo() callback is supported.
-    // For compatibility with ASIO 1.0 drivers the host application
-    // should always support the "old" bufferSwitch method, too.
-    ret = 0;
-    break;
-  case kAsioSupportsTimeCode:
-    // Informs the driver whether application is interested in time
-    // code info.  If an application does not need to know about time
-    // code, the driver has less work to do.
-    ret = 0;
-    break;
-  }
-  return ret;
-}
-
-static const char* getAsioErrorString( ASIOError result )
-{
-  struct Messages 
-  {
-    ASIOError value;
-    const char*message;
-  };
-
-  static Messages m[] = 
-    {
-      {   ASE_NotPresent,    "Hardware input or output is not present or available." },
-      {   ASE_HWMalfunction,  "Hardware is malfunctioning." },
-      {   ASE_InvalidParameter, "Invalid input parameter." },
-      {   ASE_InvalidMode,      "Invalid mode." },
-      {   ASE_SPNotAdvancing,     "Sample position not advancing." },
-      {   ASE_NoClock,            "Sample clock or rate cannot be determined or is not present." },
-      {   ASE_NoMemory,           "Not enough memory to complete the request." }
-    };
-
-  for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
-    if ( m[i].value == result ) return m[i].message;
-
-  return "Unknown error.";
-}
-//******************** End of __WINDOWS_ASIO__ *********************//
-#endif
-
-
-#if defined(__WINDOWS_DS__) // Windows DirectSound API
-
-// Modified by Robin Davies, October 2005
-// - Improvements to DirectX pointer chasing. 
-// - 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 <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 );
-
-extern "C" 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; }
-};
-
-std::vector< 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 ( coInitialized_ ) CoUninitialize(); // balanced call.
-  if ( stream_.state != STREAM_CLOSED ) closeStream();
-}
-
-// 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.
-  bool isInput = false;
-  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &isInput );
-  if ( FAILED( result ) ) {
-    errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-  }
-
-  // Query DirectSoundCapture devices.
-  isInput = true;
-  result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &isInput );
-  if ( FAILED( result ) ) {
-    errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-  }
-
-  // Clean out any devices that may have disappeared.
-  std::vector< DsDevice > :: iterator it;
-  for ( it=dsDevices.begin(); it < dsDevices.end(); it++ )
-    if ( it->found == false ) dsDevices.erase( it );
-
-  return 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( RtError::INVALID_USE );
-    }
-  }
-
-  if ( device >= dsDevices.size() ) {
-    errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
-    error( RtError::INVALID_USE );
-  }
-
-  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( RtError::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( RtError::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] );
-  }
-
-  // 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( RtError::WARNING );
-    return info;
-  }
-
-  DSCCAPS inCaps;
-  inCaps.dwSize = sizeof( inCaps );
-  result = input->GetCaps( &inCaps );
-  if ( FAILED( result ) ) {
-    input->Release();
-    errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
-    errorText_ = errorStream_.str();
-    error( RtError::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] );
-  }
-  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;
-  }
-
-  unsigned int 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.
-  unsigned threadId;
-  stream_.callbackInfo.object = (void *) this;
-  stream_.callbackInfo.isRunning = true;
-  stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
-                                                &stream_.callbackInfo, 0, &threadId );
-  if ( stream_.callbackInfo.thread == 0 ) {
-    errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
-    goto error;
-  }
-
-  // Boost DS thread priority
-  SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
-  return SUCCESS;
-
- error:
-  if ( handle ) {
-    if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
-      LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
-      LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
-      if ( buffer ) buffer->Release();
-      object->Release();
-    }
-    if ( handle->buffer[1] ) {
-      LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
-      LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
-      if ( buffer ) buffer->Release();
-      object->Release();
-    }
-    CloseHandle( handle->condition );
-    delete handle;
-    stream_.apiHandle = 0;
-  }
-
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
-    }
-  }
-
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
-
-  return FAILURE;
-}
-
-void RtApiDs :: closeStream()
-{
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiDs::closeStream(): no open stream to close!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  // Stop the callback thread.
-  stream_.callbackInfo.isRunning = false;
-  WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
-  CloseHandle( (HANDLE) stream_.callbackInfo.thread );
-
-  DsHandle *handle = (DsHandle *) stream_.apiHandle;
-  if ( handle ) {
-    if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
-      LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
-      LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
-      if ( buffer ) {
-        buffer->Stop();
-        buffer->Release();
-      }
-      object->Release();
-    }
-    if ( handle->buffer[1] ) {
-      LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
-      LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
-      if ( buffer ) {
-        buffer->Stop();
-        buffer->Release();
-      }
-      object->Release();
-    }
-    CloseHandle( handle->condition );
-    delete handle;
-    stream_.apiHandle = 0;
-  }
-
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
-    }
-  }
-
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
-
-  stream_.mode = UNINITIALIZED;
-  stream_.state = STREAM_CLOSED;
-}
-
-void RtApiDs :: startStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_RUNNING ) {
-    errorText_ = "RtApiDs::startStream(): the stream is already running!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-  
-  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;
-  stream_.state = STREAM_RUNNING;
-
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
-}
-
-void RtApiDs :: stopStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-
-  if ( stream_.state == STREAM_STOPPED ) {
-    MUTEX_UNLOCK( &stream_.mutex );
-    return;
-  }
-
-  HRESULT result = 0;
-  LPVOID audioPtr;
-  DWORD dataLen;
-  DsHandle *handle = (DsHandle *) stream_.apiHandle;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-    if ( handle->drainCounter == 0 ) {
-      handle->drainCounter = 1;
-      MUTEX_UNLOCK( &stream_.mutex );
-      WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE );  // block until signaled
-      ResetEvent( handle->condition );
-      MUTEX_LOCK( &stream_.mutex );
-    }
-
-    // Stop the buffer and clear memory
-    LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
-    result = buffer->Stop();
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-
-    // Lock the buffer and clear it so that if we start to play again,
-    // we won't have old data playing.
-    result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-
-    // Zero the DS buffer
-    ZeroMemory( audioPtr, dataLen );
-
-    // Unlock the DS buffer
-    result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-
-    // If we start playing again, we must begin at beginning of buffer.
-    handle->bufferPointer[0] = 0;
-  }
-
-  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-    LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
-    audioPtr = NULL;
-    dataLen = 0;
-
-    result = buffer->Stop();
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-
-    // Lock the buffer and clear it so that if we start to play again,
-    // we won't have old data playing.
-    result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-
-    // Zero the DS buffer
-    ZeroMemory( audioPtr, dataLen );
-
-    // Unlock the DS buffer
-    result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-
-    // If we start recording again, we must begin at beginning of buffer.
-    handle->bufferPointer[1] = 0;
-  }
-
- unlock:
-  timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
-  stream_.state = STREAM_STOPPED;
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
-}
-
-void RtApiDs :: abortStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  DsHandle *handle = (DsHandle *) stream_.apiHandle;
-  handle->drainCounter = 1;
-
-  stopStream();
-}
-
-void RtApiDs :: callbackEvent()
-{
-  if ( stream_.state == STREAM_STOPPED ) {
-    Sleep( 50 ); // sleep 50 milliseconds
-    return;
-  }
-
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
-  DsHandle *handle = (DsHandle *) stream_.apiHandle;
-
-  // Check if we were draining the stream and signal is finished.
-  if ( handle->drainCounter > stream_.nBuffers + 2 ) {
-    if ( handle->internalDrain == false )
-      SetEvent( handle->condition );
-    else
-      stopStream();
-    return;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-
-  // The state might change while waiting on a mutex.
-  if ( stream_.state == STREAM_STOPPED ) {
-    MUTEX_UNLOCK( &stream_.mutex );
-    return;
-  }
-
-  // Invoke user callback to get fresh output data UNLESS we are
-  // draining stream.
-  if ( handle->drainCounter == 0 ) {
-    RtAudioCallback callback = (RtAudioCallback) info->callback;
-    double streamTime = getStreamTime();
-    RtAudioStreamStatus status = 0;
-    if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
-      status |= RTAUDIO_OUTPUT_UNDERFLOW;
-      handle->xrun[0] = false;
-    }
-    if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
-      status |= RTAUDIO_INPUT_OVERFLOW;
-      handle->xrun[1] = false;
-    }
-    handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
-                                     stream_.bufferSize, streamTime, status, info->userData );
-    if ( handle->drainCounter == 2 ) {
-      MUTEX_UNLOCK( &stream_.mutex );
-      abortStream();
-      return;
-    }
-    else if ( handle->drainCounter == 1 )
-      handle->internalDrain = true;
-  }
-
-  HRESULT result;
-  DWORD currentWritePointer, safeWritePointer;
-  DWORD currentReadPointer, safeReadPointer;
-  UINT nextWritePointer;
-
-  LPVOID buffer1 = NULL;
-  LPVOID buffer2 = NULL;
-  DWORD bufferSize1 = 0;
-  DWORD bufferSize2 = 0;
-
-  char *buffer;
-  long bufferBytes;
-
-  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();
-        error( RtError::SYSTEM_ERROR );
-      }
-      result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
-      if ( FAILED( result ) ) {
-        errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
-        errorText_ = errorStream_.str();
-        error( RtError::SYSTEM_ERROR );
-      }
-      while ( true ) {
-        result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
-        if ( FAILED( result ) ) {
-          errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
-          errorText_ = errorStream_.str();
-          error( RtError::SYSTEM_ERROR );
-        }
-        result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
-        if ( FAILED( result ) ) {
-          errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
-          errorText_ = errorStream_.str();
-          error( RtError::SYSTEM_ERROR );
-        }
-        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();
-        error( RtError::SYSTEM_ERROR );
-      }
-      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();
-        error( RtError::SYSTEM_ERROR );
-      }
-
-      // 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();
-      error( RtError::SYSTEM_ERROR );
-    }
-
-    // Copy our buffer into the DS buffer
-    CopyMemory( buffer1, buffer, bufferSize1 );
-    if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
-
-    // Update our buffer offset and unlock sound buffer
-    dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
-    if ( FAILED( result ) ) {
-      errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
-      errorText_ = errorStream_.str();
-      error( RtError::SYSTEM_ERROR );
-    }
-    nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
-    handle->bufferPointer[0] = nextWritePointer;
-
-    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();
-      error( RtError::SYSTEM_ERROR );
-    }
-
-    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 ) {
-        // 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();
-          error( RtError::SYSTEM_ERROR );
-        }
-      
-        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();
-      error( RtError::SYSTEM_ERROR );
-    }
-
-    if ( duplexPrerollBytes <= 0 ) {
-      // Copy our buffer into the DS buffer
-      CopyMemory( buffer, buffer1, bufferSize1 );
-      if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
-    }
-    else {
-      memset( buffer, 0, bufferSize1 );
-      if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
-      duplexPrerollBytes -= bufferSize1 + bufferSize2;
-    }
-
-    // Update our buffer offset and unlock sound buffer
-    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();
-      error( RtError::SYSTEM_ERROR );
-    }
-    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.
-
-extern "C" unsigned __stdcall callbackHandler( void *ptr )
-{
-  CallbackInfo *info = (CallbackInfo *) ptr;
-  RtApiDs *object = (RtApiDs *) info->object;
-  bool* isRunning = &info->isRunning;
-
-  while ( *isRunning == true ) {
-    object->callbackEvent();
-  }
-
-  _endthreadex( 0 );
-  return 0;
-}
-
-#include "tchar.h"
-
-std::string convertTChar( LPCTSTR name )
-{
-  std::string s;
-
-#if defined( UNICODE ) || defined( _UNICODE )
-  // Yes, this conversion doesn't make sense for two-byte characters
-  // but RtAudio is currently written to return an std::string of
-  // one-byte chars for the device name.
-  for ( unsigned int i=0; i<wcslen( name ); i++ )
-    s.push_back( name[i] );
-#else
-  s.append( std::string( name ) );
-#endif
-
-  return s;
-}
-
-static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
-                                          LPCTSTR description,
-                                          LPCTSTR module,
-                                          LPVOID lpContext )
-{
-  bool *isInput = (bool *) lpContext;
-
-  HRESULT hr;
-  bool validDevice = false;
-  if ( *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 = convertTChar( description );
-  if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
-    name = "Default Device";
-  if ( validDevice ) {
-    for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
-      if ( dsDevices[i].name == name ) {
-        dsDevices[i].found = true;
-        if ( *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 ( *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;
-
-  AlsaHandle()
-    :synchronized(false) { xrun[0] = false; xrun[1] = false; }
-};
-
-extern "C" void *alsaCallbackHandler( void * ptr );
-
-RtApiAlsa :: RtApiAlsa()
-{
-  // Nothing to do here.
-}
-
-RtApiAlsa :: ~RtApiAlsa()
-{
-  if ( stream_.state != STREAM_CLOSED ) closeStream();
-}
-
-unsigned int RtApiAlsa :: getDeviceCount( void )
-{
-  unsigned nDevices = 0;
-  int result, subdevice, card;
-  char name[64];
-  snd_ctl_t *handle;
-
-  // Count cards and devices
-  card = -1;
-  snd_card_next( &card );
-  while ( card >= 0 ) {
-    sprintf( name, "hw:%d", card );
-    result = snd_ctl_open( &handle, name, 0 );
-    if ( result < 0 ) {
-      errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
-      errorText_ = errorStream_.str();
-      error( RtError::WARNING );
-      goto nextcard;
-    }
-    subdevice = -1;
-    while( 1 ) {
-      result = snd_ctl_pcm_next_device( handle, &subdevice );
-      if ( result < 0 ) {
-        errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
-        errorText_ = errorStream_.str();
-        error( RtError::WARNING );
-        break;
-      }
-      if ( subdevice < 0 )
-        break;
-      nDevices++;
-    }
-  nextcard:
-    snd_ctl_close( handle );
-    snd_card_next( &card );
-  }
-
-  return nDevices;
-}
-
-RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
-{
-  RtAudio::DeviceInfo info;
-  info.probed = false;
-
-  unsigned nDevices = 0;
-  int result, subdevice, card;
-  char name[64];
-  snd_ctl_t *chandle;
-
-  // Count cards and devices
-  card = -1;
-  snd_card_next( &card );
-  while ( card >= 0 ) {
-    sprintf( name, "hw:%d", card );
-    result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
-    if ( result < 0 ) {
-      errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
-      errorText_ = errorStream_.str();
-      error( RtError::WARNING );
-      goto nextcard;
-    }
-    subdevice = -1;
-    while( 1 ) {
-      result = snd_ctl_pcm_next_device( chandle, &subdevice );
-      if ( result < 0 ) {
-        errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
-        errorText_ = errorStream_.str();
-        error( RtError::WARNING );
-        break;
-      }
-      if ( subdevice < 0 ) break;
-      if ( nDevices == device ) {
-        sprintf( name, "hw:%d,%d", card, subdevice );
-        goto foundDevice;
-      }
-      nDevices++;
-    }
-  nextcard:
-    snd_ctl_close( chandle );
-    snd_card_next( &card );
-  }
-
-  if ( nDevices == 0 ) {
-    errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
-    error( RtError::INVALID_USE );
-  }
-
-  if ( device >= nDevices ) {
-    errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
-    error( RtError::INVALID_USE );
-  }
-
- foundDevice:
-
-  // If a stream is already open, we cannot probe the stream devices.
-  // Thus, use the saved results.
-  if ( stream_.state != STREAM_CLOSED &&
-       ( stream_.device[0] == device || stream_.device[1] == device ) ) {
-    if ( device >= devices_.size() ) {
-      errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
-      error( RtError::WARNING );
-      return info;
-    }
-    return devices_[ device ];
-  }
-
-  int openMode = SND_PCM_ASYNC;
-  snd_pcm_stream_t stream;
-  snd_pcm_info_t *pcminfo;
-  snd_pcm_info_alloca( &pcminfo );
-  snd_pcm_t *phandle;
-  snd_pcm_hw_params_t *params;
-  snd_pcm_hw_params_alloca( &params );
-
-  // First try for playback
-  stream = SND_PCM_STREAM_PLAYBACK;
-  snd_pcm_info_set_device( pcminfo, subdevice );
-  snd_pcm_info_set_subdevice( pcminfo, 0 );
-  snd_pcm_info_set_stream( pcminfo, stream );
-
-  result = snd_ctl_pcm_info( chandle, pcminfo );
-  if ( result < 0 ) {
-    // Device probably doesn't support playback.
-    goto captureProbe;
-  }
-
-  result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
-  if ( result < 0 ) {
-    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    goto captureProbe;
-  }
-
-  // The device is open ... fill the parameter structure.
-  result = snd_pcm_hw_params_any( phandle, params );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    goto captureProbe;
-  }
-
-  // Get output channel information.
-  unsigned int value;
-  result = snd_pcm_hw_params_get_channels_max( params, &value );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    goto captureProbe;
-  }
-  info.outputChannels = value;
-  snd_pcm_close( phandle );
-
- captureProbe:
-  // Now try for capture
-  stream = SND_PCM_STREAM_CAPTURE;
-  snd_pcm_info_set_stream( pcminfo, stream );
-
-  result = snd_ctl_pcm_info( chandle, pcminfo );
-  snd_ctl_close( chandle );
-  if ( result < 0 ) {
-    // Device probably doesn't support capture.
-    if ( info.outputChannels == 0 ) return info;
-    goto probeParameters;
-  }
-
-  result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
-  if ( result < 0 ) {
-    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    if ( info.outputChannels == 0 ) return info;
-    goto probeParameters;
-  }
-
-  // The device is open ... fill the parameter structure.
-  result = snd_pcm_hw_params_any( phandle, params );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    if ( info.outputChannels == 0 ) return info;
-    goto probeParameters;
-  }
-
-  result = snd_pcm_hw_params_get_channels_max( params, &value );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    if ( info.outputChannels == 0 ) return info;
-    goto probeParameters;
-  }
-  info.inputChannels = value;
-  snd_pcm_close( phandle );
-
-  // If device opens for both playback and capture, we determine the channels.
-  if ( info.outputChannels > 0 && info.inputChannels > 0 )
-    info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
-
-  // ALSA doesn't provide default devices so we'll use the first available one.
-  if ( device == 0 && info.outputChannels > 0 )
-    info.isDefaultOutput = true;
-  if ( device == 0 && info.inputChannels > 0 )
-    info.isDefaultInput = true;
-
- probeParameters:
-  // At this point, we just need to figure out the supported data
-  // formats and sample rates.  We'll proceed by opening the device in
-  // the direction with the maximum number of channels, or playback if
-  // they are equal.  This might limit our sample rate options, but so
-  // be it.
-
-  if ( info.outputChannels >= info.inputChannels )
-    stream = SND_PCM_STREAM_PLAYBACK;
-  else
-    stream = SND_PCM_STREAM_CAPTURE;
-  snd_pcm_info_set_stream( pcminfo, stream );
-
-  result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
-  if ( result < 0 ) {
-    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
-
-  // The device is open ... fill the parameter structure.
-  result = snd_pcm_hw_params_any( phandle, params );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
-
-  // Test our discrete set of sample rate values.
-  info.sampleRates.clear();
-  for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
-    if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 )
-      info.sampleRates.push_back( SAMPLE_RATES[i] );
-  }
-  if ( info.sampleRates.size() == 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
-
-  // Probe the supported data formats ... we don't care about endian-ness just yet
-  snd_pcm_format_t format;
-  info.nativeFormats = 0;
-  format = SND_PCM_FORMAT_S8;
-  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
-    info.nativeFormats |= RTAUDIO_SINT8;
-  format = SND_PCM_FORMAT_S16;
-  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
-    info.nativeFormats |= RTAUDIO_SINT16;
-  format = SND_PCM_FORMAT_S24;
-  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
-    info.nativeFormats |= RTAUDIO_SINT24;
-  format = SND_PCM_FORMAT_S32;
-  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
-    info.nativeFormats |= RTAUDIO_SINT32;
-  format = SND_PCM_FORMAT_FLOAT;
-  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
-    info.nativeFormats |= RTAUDIO_FLOAT32;
-  format = SND_PCM_FORMAT_FLOAT64;
-  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
-    info.nativeFormats |= RTAUDIO_FLOAT64;
-
-  // Check that we have at least one supported format
-  if ( info.nativeFormats == 0 ) {
-    errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
-
-  // Get the device name
-  char *cardname;
-  result = snd_card_get_name( card, &cardname );
-  if ( result >= 0 )
-    sprintf( name, "hw:%s,%d", cardname, subdevice );
-  info.name = name;
-
-  // That's all ... close the device and return
-  snd_pcm_close( phandle );
-  info.probed = true;
-  return info;
-}
-
-void RtApiAlsa :: saveDeviceInfo( void )
-{
-  devices_.clear();
-
-  unsigned int nDevices = getDeviceCount();
-  devices_.resize( nDevices );
-  for ( unsigned int i=0; i<nDevices; i++ )
-    devices_[i] = getDeviceInfo( i );
-}
-
-bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-                                   unsigned int firstChannel, unsigned int sampleRate,
-                                   RtAudioFormat format, unsigned int *bufferSize,
-                                   RtAudio::StreamOptions *options )
-
-{
-#if defined(__RTAUDIO_DEBUG__)
-  snd_output_t *out;
-  snd_output_stdio_attach(&out, stderr, 0);
-#endif
-
-  // I'm not using the "plug" interface ... too much inconsistent behavior.
-
-  unsigned nDevices = 0;
-  int result, subdevice, card;
-  char name[64];
-  snd_ctl_t *chandle;
-
-  // Count cards and devices
-  card = -1;
-  snd_card_next( &card );
-  while ( card >= 0 ) {
-    sprintf( name, "hw:%d", card );
-    result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
-    if ( result < 0 ) {
-      errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
-      errorText_ = errorStream_.str();
-      return FAILURE;
-    }
-    subdevice = -1;
-    while( 1 ) {
-      result = snd_ctl_pcm_next_device( chandle, &subdevice );
-      if ( result < 0 ) break;
-      if ( subdevice < 0 ) break;
-      if ( nDevices == device ) {
-        sprintf( name, "hw:%d,%d", card, subdevice );
-        snd_ctl_close( chandle );
-        goto foundDevice;
-      }
-      nDevices++;
-    }
-    snd_ctl_close( chandle );
-    snd_card_next( &card );
-  }
-
-  if ( nDevices == 0 ) {
-    // This should not happen because a check is made before this function is called.
-    errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
-    return FAILURE;
-  }
-
-  if ( device >= nDevices ) {
-    // This should not happen because a check is made before this function is called.
-    errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
-    return FAILURE;
-  }
-
- foundDevice:
-
-  // The getDeviceInfo() function will not work for a device that is
-  // already open.  Thus, we'll probe the system before opening a
-  // stream and save the results for use by getDeviceInfo().
-  if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
-    this->saveDeviceInfo();
-
-  snd_pcm_stream_t stream;
-  if ( mode == OUTPUT )
-    stream = SND_PCM_STREAM_PLAYBACK;
-  else
-    stream = SND_PCM_STREAM_CAPTURE;
-
-  snd_pcm_t *phandle;
-  int openMode = SND_PCM_ASYNC;
-  result = snd_pcm_open( &phandle, name, stream, openMode );
-  if ( result < 0 ) {
-    if ( mode == OUTPUT )
-      errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
-    else
-      errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Fill the parameter structure.
-  snd_pcm_hw_params_t *hw_params;
-  snd_pcm_hw_params_alloca( &hw_params );
-  result = snd_pcm_hw_params_any( phandle, hw_params );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-#if defined(__RTAUDIO_DEBUG__)
-  fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
-  snd_pcm_hw_params_dump( hw_params, out );
-#endif
-
-  // Set access ... check user preference.
-  if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
-    stream_.userInterleaved = false;
-    result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
-    if ( result < 0 ) {
-      result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
-      stream_.deviceInterleaved[mode] =  true;
-    }
-    else
-      stream_.deviceInterleaved[mode] = false;
-  }
-  else {
-    stream_.userInterleaved = true;
-    result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
-    if ( result < 0 ) {
-      result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
-      stream_.deviceInterleaved[mode] =  false;
-    }
-    else
-      stream_.deviceInterleaved[mode] =  true;
-  }
-
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Determine how to set the device format.
-  stream_.userFormat = format;
-  snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
-
-  if ( format == RTAUDIO_SINT8 )
-    deviceFormat = SND_PCM_FORMAT_S8;
-  else if ( format == RTAUDIO_SINT16 )
-    deviceFormat = SND_PCM_FORMAT_S16;
-  else if ( format == RTAUDIO_SINT24 )
-    deviceFormat = SND_PCM_FORMAT_S24;
-  else if ( format == RTAUDIO_SINT32 )
-    deviceFormat = SND_PCM_FORMAT_S32;
-  else if ( format == RTAUDIO_FLOAT32 )
-    deviceFormat = SND_PCM_FORMAT_FLOAT;
-  else if ( format == RTAUDIO_FLOAT64 )
-    deviceFormat = SND_PCM_FORMAT_FLOAT64;
-
-  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
-    stream_.deviceFormat[mode] = format;
-    goto setFormat;
-  }
-
-  // The user requested format is not natively supported by the device.
-  deviceFormat = SND_PCM_FORMAT_FLOAT64;
-  if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
-    stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
-    goto setFormat;
-  }
-
-  deviceFormat = SND_PCM_FORMAT_FLOAT;
-  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
-    stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
-    goto setFormat;
-  }
-
-  deviceFormat = SND_PCM_FORMAT_S32;
-  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
-    stream_.deviceFormat[mode] = RTAUDIO_SINT32;
-    goto setFormat;
-  }
-
-  deviceFormat = SND_PCM_FORMAT_S24;
-  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
-    stream_.deviceFormat[mode] = RTAUDIO_SINT24;
-    goto setFormat;
-  }
-
-  deviceFormat = SND_PCM_FORMAT_S16;
-  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
-    stream_.deviceFormat[mode] = RTAUDIO_SINT16;
-    goto setFormat;
-  }
-
-  deviceFormat = SND_PCM_FORMAT_S8;
-  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
-    stream_.deviceFormat[mode] = RTAUDIO_SINT8;
-    goto setFormat;
-  }
-
-  // If we get here, no supported format was found.
-  errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
-  errorText_ = errorStream_.str();
-  return FAILURE;
-
- setFormat:
-  result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Determine whether byte-swaping is necessary.
-  stream_.doByteSwap[mode] = false;
-  if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
-    result = snd_pcm_format_cpu_endian( deviceFormat );
-    if ( result == 0 )
-      stream_.doByteSwap[mode] = true;
-    else if (result < 0) {
-      snd_pcm_close( phandle );
-      errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
-      errorText_ = errorStream_.str();
-      return FAILURE;
-    }
-  }
-
-  // Set the sample rate.
-  result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Determine the number of channels for this device.  We support a possible
-  // minimum device channel number > than the value requested by the user.
-  stream_.nUserChannels[mode] = channels;
-  unsigned int value;
-  result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
-  unsigned int deviceChannels = value;
-  if ( result < 0 || deviceChannels < channels + firstChannel ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-  deviceChannels = value;
-  if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
-  stream_.nDeviceChannels[mode] = deviceChannels;
-
-  // Set the device channels.
-  result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Set the buffer number, which in ALSA is referred to as the "period".
-  int totalSize, dir = 0;
-  unsigned int periods = 0;
-  if ( options ) periods = options->numberOfBuffers;
-  totalSize = *bufferSize * periods;
-
-  // Set the buffer (or period) size.
-  snd_pcm_uframes_t periodSize = *bufferSize;
-  result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-  *bufferSize = periodSize;
-
-  if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
-  else periods = totalSize / *bufferSize;
-  // Even though the hardware might allow 1 buffer, it won't work reliably.
-  if ( periods < 2 ) periods = 2;
-  result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // If attempting to setup a duplex stream, the bufferSize parameter
-  // MUST be the same in both directions!
-  if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  stream_.bufferSize = *bufferSize;
-
-  // Install the hardware configuration
-  result = snd_pcm_hw_params( phandle, hw_params );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-#if defined(__RTAUDIO_DEBUG__)
-  fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
-  snd_pcm_hw_params_dump( hw_params, out );
-#endif
-
-  // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
-  snd_pcm_sw_params_t *sw_params = NULL;
-  snd_pcm_sw_params_alloca( &sw_params );
-  snd_pcm_sw_params_current( phandle, sw_params );
-  snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
-  snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
-  snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
-
-  // The following two settings were suggested by Theo Veenker
-  //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
-  //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
-
-  // here are two options for a fix
-  //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
-  snd_pcm_uframes_t val;
-  snd_pcm_sw_params_get_boundary( sw_params, &val );
-  snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
-
-  result = snd_pcm_sw_params( phandle, sw_params );
-  if ( result < 0 ) {
-    snd_pcm_close( phandle );
-    errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-#if defined(__RTAUDIO_DEBUG__)
-  fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
-  snd_pcm_sw_params_dump( sw_params, out );
-#endif
-
-  // Set flags for buffer conversion
-  stream_.doConvertBuffer[mode] = false;
-  if ( stream_.userFormat != stream_.deviceFormat[mode] )
-    stream_.doConvertBuffer[mode] = true;
-  if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
-    stream_.doConvertBuffer[mode] = true;
-  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
-       stream_.nUserChannels[mode] > 1 )
-    stream_.doConvertBuffer[mode] = true;
-
-  // Allocate the ApiHandle if necessary and then save.
-  AlsaHandle *apiInfo = 0;
-  if ( stream_.apiHandle == 0 ) {
-    try {
-      apiInfo = (AlsaHandle *) new AlsaHandle;
-    }
-    catch ( std::bad_alloc& ) {
-      errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
-      goto error;
-    }
-
-    if ( pthread_cond_init( &apiInfo->runnable, NULL ) ) {
-      errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
-      goto error;
-    }
-
-    stream_.apiHandle = (void *) apiInfo;
-    apiInfo->handles[0] = 0;
-    apiInfo->handles[1] = 0;
-  }
-  else {
-    apiInfo = (AlsaHandle *) stream_.apiHandle;
-  }
-  apiInfo->handles[mode] = phandle;
-
-  // Allocate necessary internal buffers.
-  unsigned long bufferBytes;
-  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
-  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
-  if ( stream_.userBuffer[mode] == NULL ) {
-    errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
-    goto error;
-  }
-
-  if ( stream_.doConvertBuffer[mode] ) {
-
-    bool makeBuffer = true;
-    bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
-    if ( mode == INPUT ) {
-      if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
-        unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
-        if ( bufferBytes <= bytesOut ) makeBuffer = false;
-      }
-    }
-
-    if ( makeBuffer ) {
-      bufferBytes *= *bufferSize;
-      if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
-      stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
-      if ( stream_.deviceBuffer == NULL ) {
-        errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
-        goto error;
-      }
-    }
-  }
-
-  stream_.sampleRate = sampleRate;
-  stream_.nBuffers = periods;
-  stream_.device[mode] = device;
-  stream_.state = STREAM_STOPPED;
-
-  // Setup the buffer conversion information structure.
-  if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
-
-  // Setup thread if necessary.
-  if ( stream_.mode == OUTPUT && mode == INPUT ) {
-    // We had already set up an output stream.
-    stream_.mode = DUPLEX;
-    // Link the streams if possible.
-    apiInfo->synchronized = false;
-    if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
-      apiInfo->synchronized = true;
-    else {
-      errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
-      error( RtError::WARNING );
-    }
-  }
-  else {
-    stream_.mode = mode;
-
-    // Setup callback thread.
-    stream_.callbackInfo.object = (void *) this;
-
-    // Set the thread attributes for joinable and realtime scheduling
-    // priority (optional).  The higher priority will only take affect
-    // if the program is run as root or suid. Note, under Linux
-    // processes with CAP_SYS_NICE privilege, a user can change
-    // scheduling policy and priority (thus need not be root). See
-    // POSIX "capabilities".
-    pthread_attr_t attr;
-    pthread_attr_init( &attr );
-    pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
-#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
-    if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
-      struct sched_param param;
-      int priority = options->priority;
-      int min = sched_get_priority_min( SCHED_RR );
-      int max = sched_get_priority_max( SCHED_RR );
-      if ( priority < min ) priority = min;
-      else if ( priority > max ) priority = max;
-      param.sched_priority = priority;
-      pthread_attr_setschedparam( &attr, &param );
-      pthread_attr_setschedpolicy( &attr, SCHED_RR );
-    }
-    else
-      pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
-#else
-    pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
-#endif
-
-    stream_.callbackInfo.isRunning = true;
-    result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
-    pthread_attr_destroy( &attr );
-    if ( result ) {
-      stream_.callbackInfo.isRunning = false;
-      errorText_ = "RtApiAlsa::error creating callback thread!";
-      goto error;
-    }
-  }
-
-  return SUCCESS;
-
- error:
-  if ( apiInfo ) {
-    pthread_cond_destroy( &apiInfo->runnable );
-    if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
-    if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
-    delete apiInfo;
-    stream_.apiHandle = 0;
-  }
-
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
-    }
-  }
-
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
-
-  return FAILURE;
-}
-
-void RtApiAlsa :: closeStream()
-{
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
-  stream_.callbackInfo.isRunning = false;
-  MUTEX_LOCK( &stream_.mutex );
-  if ( stream_.state == STREAM_STOPPED )
-    pthread_cond_signal( &apiInfo->runnable );
-  MUTEX_UNLOCK( &stream_.mutex );
-  pthread_join( stream_.callbackInfo.thread, NULL );
-
-  if ( stream_.state == STREAM_RUNNING ) {
-    stream_.state = STREAM_STOPPED;
-    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
-      snd_pcm_drop( apiInfo->handles[0] );
-    if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
-      snd_pcm_drop( apiInfo->handles[1] );
-  }
-
-  if ( apiInfo ) {
-    pthread_cond_destroy( &apiInfo->runnable );
-    if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
-    if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
-    delete apiInfo;
-    stream_.apiHandle = 0;
-  }
-
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
-    }
-  }
-
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
-
-  stream_.mode = UNINITIALIZED;
-  stream_.state = STREAM_CLOSED;
-}
-
-void RtApiAlsa :: startStream()
-{
-  // This method calls snd_pcm_prepare if the device isn't already in that state.
-
-  verifyStream();
-  if ( stream_.state == STREAM_RUNNING ) {
-    errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-
-  int result = 0;
-  snd_pcm_state_t state;
-  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
-  snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-    state = snd_pcm_state( handle[0] );
-    if ( state != SND_PCM_STATE_PREPARED ) {
-      result = snd_pcm_prepare( handle[0] );
-      if ( result < 0 ) {
-        errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
-        errorText_ = errorStream_.str();
-        goto unlock;
-      }
-    }
-  }
-
-  if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
-    state = snd_pcm_state( handle[1] );
-    if ( state != SND_PCM_STATE_PREPARED ) {
-      result = snd_pcm_prepare( handle[1] );
-      if ( result < 0 ) {
-        errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
-        errorText_ = errorStream_.str();
-        goto unlock;
-      }
-    }
-  }
-
-  stream_.state = STREAM_RUNNING;
-
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  pthread_cond_signal( &apiInfo->runnable );
-
-  if ( result >= 0 ) return;
-  error( RtError::SYSTEM_ERROR );
-}
-
-void RtApiAlsa :: stopStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-
-  if ( stream_.state == STREAM_STOPPED ) {
-    MUTEX_UNLOCK( &stream_.mutex );
-    return;
-  }
-
-  int result = 0;
-  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
-  snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-    if ( apiInfo->synchronized ) 
-      result = snd_pcm_drop( handle[0] );
-    else
-      result = snd_pcm_drain( handle[0] );
-    if ( result < 0 ) {
-      errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-  }
-
-  if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
-    result = snd_pcm_drop( handle[1] );
-    if ( result < 0 ) {
-      errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-  }
-
- unlock:
-  stream_.state = STREAM_STOPPED;
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  if ( result >= 0 ) return;
-  error( RtError::SYSTEM_ERROR );
-}
-
-void RtApiAlsa :: abortStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-
-  if ( stream_.state == STREAM_STOPPED ) {
-    MUTEX_UNLOCK( &stream_.mutex );
-    return;
-  }
-
-  int result = 0;
-  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
-  snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-    result = snd_pcm_drop( handle[0] );
-    if ( result < 0 ) {
-      errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-  }
-
-  if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
-    result = snd_pcm_drop( handle[1] );
-    if ( result < 0 ) {
-      errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-  }
-
- unlock:
-  stream_.state = STREAM_STOPPED;
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  if ( result >= 0 ) return;
-  error( RtError::SYSTEM_ERROR );
-}
-
-void RtApiAlsa :: callbackEvent()
-{
-  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
-  if ( stream_.state == STREAM_STOPPED ) {
-    MUTEX_LOCK( &stream_.mutex );
-    pthread_cond_wait( &apiInfo->runnable, &stream_.mutex );
-    if ( stream_.state != STREAM_RUNNING ) {
-      MUTEX_UNLOCK( &stream_.mutex );
-      return;
-    }
-    MUTEX_UNLOCK( &stream_.mutex );
-  }
-
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  int doStopStream = 0;
-  RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
-  double streamTime = getStreamTime();
-  RtAudioStreamStatus status = 0;
-  if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
-    status |= RTAUDIO_OUTPUT_UNDERFLOW;
-    apiInfo->xrun[0] = false;
-  }
-  if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
-    status |= RTAUDIO_INPUT_OVERFLOW;
-    apiInfo->xrun[1] = false;
-  }
-  doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
-                           stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
-
-  if ( doStopStream == 2 ) {
-    abortStream();
-    return;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-
-  // The state might change while waiting on a mutex.
-  if ( stream_.state == STREAM_STOPPED ) goto unlock;
-
-  int result;
-  char *buffer;
-  int channels;
-  snd_pcm_t **handle;
-  snd_pcm_sframes_t frames;
-  RtAudioFormat format;
-  handle = (snd_pcm_t **) apiInfo->handles;
-
-  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-
-    // Setup parameters.
-    if ( stream_.doConvertBuffer[1] ) {
-      buffer = stream_.deviceBuffer;
-      channels = stream_.nDeviceChannels[1];
-      format = stream_.deviceFormat[1];
-    }
-    else {
-      buffer = stream_.userBuffer[1];
-      channels = stream_.nUserChannels[1];
-      format = stream_.userFormat;
-    }
-
-    // Read samples from device in interleaved/non-interleaved format.
-    if ( stream_.deviceInterleaved[1] )
-      result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
-    else {
-      void *bufs[channels];
-      size_t offset = stream_.bufferSize * formatBytes( format );
-      for ( int i=0; i<channels; i++ )
-        bufs[i] = (void *) (buffer + (i * offset));
-      result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
-    }
-
-    if ( result < (int) stream_.bufferSize ) {
-      // Either an error or overrun occured.
-      if ( result == -EPIPE ) {
-        snd_pcm_state_t state = snd_pcm_state( handle[1] );
-        if ( state == SND_PCM_STATE_XRUN ) {
-          apiInfo->xrun[1] = true;
-          result = snd_pcm_prepare( handle[1] );
-          if ( result < 0 ) {
-            errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
-            errorText_ = errorStream_.str();
-          }
-        }
-        else {
-          errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
-          errorText_ = errorStream_.str();
-        }
-      }
-      else {
-        errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
-        errorText_ = errorStream_.str();
-      }
-      error( RtError::WARNING );
-      goto tryOutput;
-    }
-
-    // Do byte swapping if necessary.
-    if ( stream_.doByteSwap[1] )
-      byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
-
-    // Do buffer conversion if necessary.
-    if ( stream_.doConvertBuffer[1] )
-      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
-
-    // Check stream latency
-    result = snd_pcm_delay( handle[1], &frames );
-    if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
-  }
-
- tryOutput:
-
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
-    // Setup parameters and do buffer conversion if necessary.
-    if ( stream_.doConvertBuffer[0] ) {
-      buffer = stream_.deviceBuffer;
-      convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
-      channels = stream_.nDeviceChannels[0];
-      format = stream_.deviceFormat[0];
-    }
-    else {
-      buffer = stream_.userBuffer[0];
-      channels = stream_.nUserChannels[0];
-      format = stream_.userFormat;
-    }
-
-    // Do byte swapping if necessary.
-    if ( stream_.doByteSwap[0] )
-      byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
-
-    // Write samples to device in interleaved/non-interleaved format.
-    if ( stream_.deviceInterleaved[0] )
-      result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
-    else {
-      void *bufs[channels];
-      size_t offset = stream_.bufferSize * formatBytes( format );
-      for ( int i=0; i<channels; i++ )
-        bufs[i] = (void *) (buffer + (i * offset));
-      result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
-    }
-
-    if ( result < (int) stream_.bufferSize ) {
-      // Either an error or underrun occured.
-      if ( result == -EPIPE ) {
-        snd_pcm_state_t state = snd_pcm_state( handle[0] );
-        if ( state == SND_PCM_STATE_XRUN ) {
-          apiInfo->xrun[0] = true;
-          result = snd_pcm_prepare( handle[0] );
-          if ( result < 0 ) {
-            errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
-            errorText_ = errorStream_.str();
-          }
-        }
-        else {
-          errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
-          errorText_ = errorStream_.str();
-        }
-      }
-      else {
-        errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
-        errorText_ = errorStream_.str();
-      }
-      error( RtError::WARNING );
-      goto unlock;
-    }
-
-    // Check stream latency
-    result = snd_pcm_delay( handle[0], &frames );
-    if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
-  }
-
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  RtApi::tickStreamTime();
-  if ( doStopStream == 1 ) this->stopStream();
-}
-
-extern "C" void *alsaCallbackHandler( void *ptr )
-{
-  CallbackInfo *info = (CallbackInfo *) ptr;
-  RtApiAlsa *object = (RtApiAlsa *) info->object;
-  bool *isRunning = &info->isRunning;
-
-  while ( *isRunning == true ) {
-    pthread_testcancel();
-    object->callbackEvent();
-  }
-
-  pthread_exit( NULL );
-}
-
-//******************** End of __LINUX_ALSA__ *********************//
-#endif
-
-
-#if defined(__LINUX_OSS__)
-
-#include <unistd.h>
-#include <sys/ioctl.h>
-#include <unistd.h>
-#include <fcntl.h>
-#include "soundcard.h"
-#include <errno.h>
-#include <math.h>
-
-extern "C" void *ossCallbackHandler(void * ptr);
-
-// A structure to hold various information related to the OSS API
-// implementation.
-struct OssHandle {
-  int id[2];    // device ids
-  bool xrun[2];
-  bool triggered;
-  pthread_cond_t runnable;
-
-  OssHandle()
-    :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
-};
-
-RtApiOss :: RtApiOss()
-{
-  // Nothing to do here.
-}
-
-RtApiOss :: ~RtApiOss()
-{
-  if ( stream_.state != STREAM_CLOSED ) closeStream();
-}
-
-unsigned int RtApiOss :: getDeviceCount( void )
-{
-  int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
-  if ( mixerfd == -1 ) {
-    errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
-    error( RtError::WARNING );
-    return 0;
-  }
-
-  oss_sysinfo sysinfo;
-  if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
-    close( mixerfd );
-    errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
-    error( RtError::WARNING );
-    return 0;
-  }
-
-  close( mixerfd );
-  return sysinfo.numaudios;
-}
-
-RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
-{
-  RtAudio::DeviceInfo info;
-  info.probed = false;
-
-  int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
-  if ( mixerfd == -1 ) {
-    errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
-    error( RtError::WARNING );
-    return info;
-  }
-
-  oss_sysinfo sysinfo;
-  int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
-  if ( result == -1 ) {
-    close( mixerfd );
-    errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
-    error( RtError::WARNING );
-    return info;
-  }
-
-  unsigned nDevices = sysinfo.numaudios;
-  if ( nDevices == 0 ) {
-    close( mixerfd );
-    errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
-    error( RtError::INVALID_USE );
-  }
-
-  if ( device >= nDevices ) {
-    close( mixerfd );
-    errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
-    error( RtError::INVALID_USE );
-  }
-
-  oss_audioinfo ainfo;
-  ainfo.dev = device;
-  result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
-  close( mixerfd );
-  if ( result == -1 ) {
-    errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
-
-  // Probe channels
-  if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
-  if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
-  if ( ainfo.caps & PCM_CAP_DUPLEX ) {
-    if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
-      info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
-  }
-
-  // Probe data formats ... do for input
-  unsigned long mask = ainfo.iformats;
-  if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
-    info.nativeFormats |= RTAUDIO_SINT16;
-  if ( mask & AFMT_S8 )
-    info.nativeFormats |= RTAUDIO_SINT8;
-  if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
-    info.nativeFormats |= RTAUDIO_SINT32;
-  if ( mask & AFMT_FLOAT )
-    info.nativeFormats |= RTAUDIO_FLOAT32;
-  if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
-    info.nativeFormats |= RTAUDIO_SINT24;
-
-  // Check that we have at least one supported format
-  if ( info.nativeFormats == 0 ) {
-    errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-    return info;
-  }
-
-  // Probe the supported sample rates.
-  info.sampleRates.clear();
-  if ( ainfo.nrates ) {
-    for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
-      for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
-        if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
-          info.sampleRates.push_back( SAMPLE_RATES[k] );
-          break;
-        }
-      }
-    }
-  }
-  else {
-    // Check min and max rate values;
-    for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
-      if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] )
-        info.sampleRates.push_back( SAMPLE_RATES[k] );
-    }
-  }
-
-  if ( info.sampleRates.size() == 0 ) {
-    errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
-    errorText_ = errorStream_.str();
-    error( RtError::WARNING );
-  }
-  else {
-    info.probed = true;
-    info.name = ainfo.name;
-  }
-
-  return info;
-}
-
-
-bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
-                                  unsigned int firstChannel, unsigned int sampleRate,
-                                  RtAudioFormat format, unsigned int *bufferSize,
-                                  RtAudio::StreamOptions *options )
-{
-  int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
-  if ( mixerfd == -1 ) {
-    errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
-    return FAILURE;
-  }
-
-  oss_sysinfo sysinfo;
-  int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
-  if ( result == -1 ) {
-    close( mixerfd );
-    errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
-    return FAILURE;
-  }
-
-  unsigned nDevices = sysinfo.numaudios;
-  if ( nDevices == 0 ) {
-    // This should not happen because a check is made before this function is called.
-    close( mixerfd );
-    errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
-    return FAILURE;
-  }
-
-  if ( device >= nDevices ) {
-    // This should not happen because a check is made before this function is called.
-    close( mixerfd );
-    errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
-    return FAILURE;
-  }
-
-  oss_audioinfo ainfo;
-  ainfo.dev = device;
-  result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
-  close( mixerfd );
-  if ( result == -1 ) {
-    errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Check if device supports input or output
-  if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
-       ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
-    if ( mode == OUTPUT )
-      errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
-    else
-      errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  int flags = 0;
-  OssHandle *handle = (OssHandle *) stream_.apiHandle;
-  if ( mode == OUTPUT )
-    flags |= O_WRONLY;
-  else { // mode == INPUT
-    if (stream_.mode == OUTPUT && stream_.device[0] == device) {
-      // We just set the same device for playback ... close and reopen for duplex (OSS only).
-      close( handle->id[0] );
-      handle->id[0] = 0;
-      if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
-        errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
-        errorText_ = errorStream_.str();
-        return FAILURE;
-      }
-      // Check that the number previously set channels is the same.
-      if ( stream_.nUserChannels[0] != channels ) {
-        errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
-        errorText_ = errorStream_.str();
-        return FAILURE;
-      }
-      flags |= O_RDWR;
-    }
-    else
-      flags |= O_RDONLY;
-  }
-
-  // Set exclusive access if specified.
-  if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
-
-  // Try to open the device.
-  int fd;
-  fd = open( ainfo.devnode, flags, 0 );
-  if ( fd == -1 ) {
-    if ( errno == EBUSY )
-      errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
-    else
-      errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // For duplex operation, specifically set this mode (this doesn't seem to work).
-  /*
-    if ( flags | O_RDWR ) {
-    result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
-    if ( result == -1) {
-    errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-    }
-    }
-  */
-
-  // Check the device channel support.
-  stream_.nUserChannels[mode] = channels;
-  if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
-    close( fd );
-    errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Set the number of channels.
-  int deviceChannels = channels + firstChannel;
-  result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
-  if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
-    close( fd );
-    errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-  stream_.nDeviceChannels[mode] = deviceChannels;
-
-  // Get the data format mask
-  int mask;
-  result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
-  if ( result == -1 ) {
-    close( fd );
-    errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Determine how to set the device format.
-  stream_.userFormat = format;
-  int deviceFormat = -1;
-  stream_.doByteSwap[mode] = false;
-  if ( format == RTAUDIO_SINT8 ) {
-    if ( mask & AFMT_S8 ) {
-      deviceFormat = AFMT_S8;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT8;
-    }
-  }
-  else if ( format == RTAUDIO_SINT16 ) {
-    if ( mask & AFMT_S16_NE ) {
-      deviceFormat = AFMT_S16_NE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT16;
-    }
-    else if ( mask & AFMT_S16_OE ) {
-      deviceFormat = AFMT_S16_OE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT16;
-      stream_.doByteSwap[mode] = true;
-    }
-  }
-  else if ( format == RTAUDIO_SINT24 ) {
-    if ( mask & AFMT_S24_NE ) {
-      deviceFormat = AFMT_S24_NE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT24;
-    }
-    else if ( mask & AFMT_S24_OE ) {
-      deviceFormat = AFMT_S24_OE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT24;
-      stream_.doByteSwap[mode] = true;
-    }
-  }
-  else if ( format == RTAUDIO_SINT32 ) {
-    if ( mask & AFMT_S32_NE ) {
-      deviceFormat = AFMT_S32_NE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT32;
-    }
-    else if ( mask & AFMT_S32_OE ) {
-      deviceFormat = AFMT_S32_OE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT32;
-      stream_.doByteSwap[mode] = true;
-    }
-  }
-
-  if ( deviceFormat == -1 ) {
-    // The user requested format is not natively supported by the device.
-    if ( mask & AFMT_S16_NE ) {
-      deviceFormat = AFMT_S16_NE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT16;
-    }
-    else if ( mask & AFMT_S32_NE ) {
-      deviceFormat = AFMT_S32_NE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT32;
-    }
-    else if ( mask & AFMT_S24_NE ) {
-      deviceFormat = AFMT_S24_NE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT24;
-    }
-    else if ( mask & AFMT_S16_OE ) {
-      deviceFormat = AFMT_S16_OE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT16;
-      stream_.doByteSwap[mode] = true;
-    }
-    else if ( mask & AFMT_S32_OE ) {
-      deviceFormat = AFMT_S32_OE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT32;
-      stream_.doByteSwap[mode] = true;
-    }
-    else if ( mask & AFMT_S24_OE ) {
-      deviceFormat = AFMT_S24_OE;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT24;
-      stream_.doByteSwap[mode] = true;
-    }
-    else if ( mask & AFMT_S8) {
-      deviceFormat = AFMT_S8;
-      stream_.deviceFormat[mode] = RTAUDIO_SINT8;
-    }
-  }
-
-  if ( stream_.deviceFormat[mode] == 0 ) {
-    // This really shouldn't happen ...
-    close( fd );
-    errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Set the data format.
-  int temp = deviceFormat;
-  result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
-  if ( result == -1 || deviceFormat != temp ) {
-    close( fd );
-    errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Attempt to set the buffer size.  According to OSS, the minimum
-  // number of buffers is two.  The supposed minimum buffer size is 16
-  // bytes, so that will be our lower bound.  The argument to this
-  // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
-  // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
-  // We'll check the actual value used near the end of the setup
-  // procedure.
-  int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
-  if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
-  int buffers = 0;
-  if ( options ) buffers = options->numberOfBuffers;
-  if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
-  if ( buffers < 2 ) buffers = 3;
-  temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
-  result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
-  if ( result == -1 ) {
-    close( fd );
-    errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-  stream_.nBuffers = buffers;
-
-  // Save buffer size (in sample frames).
-  *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
-  stream_.bufferSize = *bufferSize;
-
-  // Set the sample rate.
-  int srate = sampleRate;
-  result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
-  if ( result == -1 ) {
-    close( fd );
-    errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-
-  // Verify the sample rate setup worked.
-  if ( abs( srate - sampleRate ) > 100 ) {
-    close( fd );
-    errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
-    errorText_ = errorStream_.str();
-    return FAILURE;
-  }
-  stream_.sampleRate = sampleRate;
-
-  if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
-    // We're doing duplex setup here.
-    stream_.deviceFormat[0] = stream_.deviceFormat[1];
-    stream_.nDeviceChannels[0] = deviceChannels;
-  }
-
-  // Set interleaving parameters.
-  stream_.userInterleaved = true;
-  stream_.deviceInterleaved[mode] =  true;
-  if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
-    stream_.userInterleaved = false;
-
-  // Set flags for buffer conversion
-  stream_.doConvertBuffer[mode] = false;
-  if ( stream_.userFormat != stream_.deviceFormat[mode] )
-    stream_.doConvertBuffer[mode] = true;
-  if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
-    stream_.doConvertBuffer[mode] = true;
-  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
-       stream_.nUserChannels[mode] > 1 )
-    stream_.doConvertBuffer[mode] = true;
-
-  // Allocate the stream handles if necessary and then save.
-  if ( stream_.apiHandle == 0 ) {
-    try {
-      handle = new OssHandle;
-    }
-    catch ( std::bad_alloc& ) {
-      errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
-      goto error;
-    }
-
-    if ( pthread_cond_init( &handle->runnable, NULL ) ) {
-      errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
-      goto error;
-    }
-
-    stream_.apiHandle = (void *) handle;
-  }
-  else {
-    handle = (OssHandle *) stream_.apiHandle;
-  }
-  handle->id[mode] = fd;
-
-  // Allocate necessary internal buffers.
-  unsigned long bufferBytes;
-  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
-  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
-  if ( stream_.userBuffer[mode] == NULL ) {
-    errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
-    goto error;
-  }
-
-  if ( stream_.doConvertBuffer[mode] ) {
-
-    bool makeBuffer = true;
-    bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
-    if ( mode == INPUT ) {
-      if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
-        unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
-        if ( bufferBytes <= bytesOut ) makeBuffer = false;
-      }
-    }
-
-    if ( makeBuffer ) {
-      bufferBytes *= *bufferSize;
-      if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
-      stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
-      if ( stream_.deviceBuffer == NULL ) {
-        errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
-        goto error;
-      }
-    }
-  }
-
-  stream_.device[mode] = device;
-  stream_.state = STREAM_STOPPED;
-
-  // Setup the buffer conversion information structure.
-  if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
-
-  // Setup thread if necessary.
-  if ( stream_.mode == OUTPUT && mode == INPUT ) {
-    // We had already set up an output stream.
-    stream_.mode = DUPLEX;
-    if ( stream_.device[0] == device ) handle->id[0] = fd;
-  }
-  else {
-    stream_.mode = mode;
-
-    // Setup callback thread.
-    stream_.callbackInfo.object = (void *) this;
-
-    // Set the thread attributes for joinable and realtime scheduling
-    // priority.  The higher priority will only take affect if the
-    // program is run as root or suid.
-    pthread_attr_t attr;
-    pthread_attr_init( &attr );
-    pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
-#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
-    if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
-      struct sched_param param;
-      int priority = options->priority;
-      int min = sched_get_priority_min( SCHED_RR );
-      int max = sched_get_priority_max( SCHED_RR );
-      if ( priority < min ) priority = min;
-      else if ( priority > max ) priority = max;
-      param.sched_priority = priority;
-      pthread_attr_setschedparam( &attr, &param );
-      pthread_attr_setschedpolicy( &attr, SCHED_RR );
-    }
-    else
-      pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
-#else
-    pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
-#endif
-
-    stream_.callbackInfo.isRunning = true;
-    result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
-    pthread_attr_destroy( &attr );
-    if ( result ) {
-      stream_.callbackInfo.isRunning = false;
-      errorText_ = "RtApiOss::error creating callback thread!";
-      goto error;
-    }
-  }
-
-  return SUCCESS;
-
- error:
-  if ( handle ) {
-    pthread_cond_destroy( &handle->runnable );
-    if ( handle->id[0] ) close( handle->id[0] );
-    if ( handle->id[1] ) close( handle->id[1] );
-    delete handle;
-    stream_.apiHandle = 0;
-  }
-
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
-    }
-  }
-
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
-
-  return FAILURE;
-}
-
-void RtApiOss :: closeStream()
-{
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiOss::closeStream(): no open stream to close!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  OssHandle *handle = (OssHandle *) stream_.apiHandle;
-  stream_.callbackInfo.isRunning = false;
-  MUTEX_LOCK( &stream_.mutex );
-  if ( stream_.state == STREAM_STOPPED )
-    pthread_cond_signal( &handle->runnable );
-  MUTEX_UNLOCK( &stream_.mutex );
-  pthread_join( stream_.callbackInfo.thread, NULL );
-
-  if ( stream_.state == STREAM_RUNNING ) {
-    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
-      ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
-    else
-      ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
-    stream_.state = STREAM_STOPPED;
-  }
-
-  if ( handle ) {
-    pthread_cond_destroy( &handle->runnable );
-    if ( handle->id[0] ) close( handle->id[0] );
-    if ( handle->id[1] ) close( handle->id[1] );
-    delete handle;
-    stream_.apiHandle = 0;
-  }
-
-  for ( int i=0; i<2; i++ ) {
-    if ( stream_.userBuffer[i] ) {
-      free( stream_.userBuffer[i] );
-      stream_.userBuffer[i] = 0;
-    }
-  }
-
-  if ( stream_.deviceBuffer ) {
-    free( stream_.deviceBuffer );
-    stream_.deviceBuffer = 0;
-  }
-
-  stream_.mode = UNINITIALIZED;
-  stream_.state = STREAM_CLOSED;
-}
-
-void RtApiOss :: startStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_RUNNING ) {
-    errorText_ = "RtApiOss::startStream(): the stream is already running!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-
-  stream_.state = STREAM_RUNNING;
-
-  // No need to do anything else here ... OSS automatically starts
-  // when fed samples.
-
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  OssHandle *handle = (OssHandle *) stream_.apiHandle;
-  pthread_cond_signal( &handle->runnable );
-}
-
-void RtApiOss :: stopStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-
-  // The state might change while waiting on a mutex.
-  if ( stream_.state == STREAM_STOPPED ) {
-    MUTEX_UNLOCK( &stream_.mutex );
-    return;
-  }
-
-  int result = 0;
-  OssHandle *handle = (OssHandle *) stream_.apiHandle;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
-    // Flush the output with zeros a few times.
-    char *buffer;
-    int samples;
-    RtAudioFormat format;
-
-    if ( stream_.doConvertBuffer[0] ) {
-      buffer = stream_.deviceBuffer;
-      samples = stream_.bufferSize * stream_.nDeviceChannels[0];
-      format = stream_.deviceFormat[0];
-    }
-    else {
-      buffer = stream_.userBuffer[0];
-      samples = stream_.bufferSize * stream_.nUserChannels[0];
-      format = stream_.userFormat;
-    }
-
-    memset( buffer, 0, samples * formatBytes(format) );
-    for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
-      result = write( handle->id[0], buffer, samples * formatBytes(format) );
-      if ( result == -1 ) {
-        errorText_ = "RtApiOss::stopStream: audio write error.";
-        error( RtError::WARNING );
-      }
-    }
-
-    result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
-    if ( result == -1 ) {
-      errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-    handle->triggered = false;
-  }
-
-  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
-    result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
-    if ( result == -1 ) {
-      errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-  }
-
- unlock:
-  stream_.state = STREAM_STOPPED;
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  if ( result != -1 ) return;
-  error( RtError::SYSTEM_ERROR );
-}
-
-void RtApiOss :: abortStream()
-{
-  verifyStream();
-  if ( stream_.state == STREAM_STOPPED ) {
-    errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-
-  // The state might change while waiting on a mutex.
-  if ( stream_.state == STREAM_STOPPED ) {
-    MUTEX_UNLOCK( &stream_.mutex );
-    return;
-  }
-
-  int result = 0;
-  OssHandle *handle = (OssHandle *) stream_.apiHandle;
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-    result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
-    if ( result == -1 ) {
-      errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-    handle->triggered = false;
-  }
-
-  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
-    result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
-    if ( result == -1 ) {
-      errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
-      errorText_ = errorStream_.str();
-      goto unlock;
-    }
-  }
-
- unlock:
-  stream_.state = STREAM_STOPPED;
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  if ( result != -1 ) return;
-  error( RtError::SYSTEM_ERROR );
-}
-
-void RtApiOss :: callbackEvent()
-{
-  OssHandle *handle = (OssHandle *) stream_.apiHandle;
-  if ( stream_.state == STREAM_STOPPED ) {
-    MUTEX_LOCK( &stream_.mutex );
-    pthread_cond_wait( &handle->runnable, &stream_.mutex );
-    if ( stream_.state != STREAM_RUNNING ) {
-      MUTEX_UNLOCK( &stream_.mutex );
-      return;
-    }
-    MUTEX_UNLOCK( &stream_.mutex );
-  }
-
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
-    error( RtError::WARNING );
-    return;
-  }
-
-  // Invoke user callback to get fresh output data.
-  int doStopStream = 0;
-  RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
-  double streamTime = getStreamTime();
-  RtAudioStreamStatus status = 0;
-  if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
-    status |= RTAUDIO_OUTPUT_UNDERFLOW;
-    handle->xrun[0] = false;
-  }
-  if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
-    status |= RTAUDIO_INPUT_OVERFLOW;
-    handle->xrun[1] = false;
-  }
-  doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
-                           stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
-  if ( doStopStream == 2 ) {
-    this->abortStream();
-    return;
-  }
-
-  MUTEX_LOCK( &stream_.mutex );
-
-  // The state might change while waiting on a mutex.
-  if ( stream_.state == STREAM_STOPPED ) goto unlock;
-
-  int result;
-  char *buffer;
-  int samples;
-  RtAudioFormat format;
-
-  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
-    // Setup parameters and do buffer conversion if necessary.
-    if ( stream_.doConvertBuffer[0] ) {
-      buffer = stream_.deviceBuffer;
-      convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
-      samples = stream_.bufferSize * stream_.nDeviceChannels[0];
-      format = stream_.deviceFormat[0];
-    }
-    else {
-      buffer = stream_.userBuffer[0];
-      samples = stream_.bufferSize * stream_.nUserChannels[0];
-      format = stream_.userFormat;
-    }
-
-    // Do byte swapping if necessary.
-    if ( stream_.doByteSwap[0] )
-      byteSwapBuffer( buffer, samples, format );
-
-    if ( stream_.mode == DUPLEX && handle->triggered == false ) {
-      int trig = 0;
-      ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
-      result = write( handle->id[0], buffer, samples * formatBytes(format) );
-      trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
-      ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
-      handle->triggered = true;
-    }
-    else
-      // Write samples to device.
-      result = write( handle->id[0], buffer, samples * formatBytes(format) );
-
-    if ( result == -1 ) {
-      // We'll assume this is an underrun, though there isn't a
-      // specific means for determining that.
-      handle->xrun[0] = true;
-      errorText_ = "RtApiOss::callbackEvent: audio write error.";
-      error( RtError::WARNING );
-      // Continue on to input section.
-    }
-  }
-
-  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-
-    // Setup parameters.
-    if ( stream_.doConvertBuffer[1] ) {
-      buffer = stream_.deviceBuffer;
-      samples = stream_.bufferSize * stream_.nDeviceChannels[1];
-      format = stream_.deviceFormat[1];
-    }
-    else {
-      buffer = stream_.userBuffer[1];
-      samples = stream_.bufferSize * stream_.nUserChannels[1];
-      format = stream_.userFormat;
-    }
-
-    // Read samples from device.
-    result = read( handle->id[1], buffer, samples * formatBytes(format) );
-
-    if ( result == -1 ) {
-      // We'll assume this is an overrun, though there isn't a
-      // specific means for determining that.
-      handle->xrun[1] = true;
-      errorText_ = "RtApiOss::callbackEvent: audio read error.";
-      error( RtError::WARNING );
-      goto unlock;
-    }
-
-    // Do byte swapping if necessary.
-    if ( stream_.doByteSwap[1] )
-      byteSwapBuffer( buffer, samples, format );
-
-    // Do buffer conversion if necessary.
-    if ( stream_.doConvertBuffer[1] )
-      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
-  }
-
- unlock:
-  MUTEX_UNLOCK( &stream_.mutex );
-
-  RtApi::tickStreamTime();
-  if ( doStopStream == 1 ) this->stopStream();
-}
-
-extern "C" void *ossCallbackHandler( void *ptr )
-{
-  CallbackInfo *info = (CallbackInfo *) ptr;
-  RtApiOss *object = (RtApiOss *) info->object;
-  bool *isRunning = &info->isRunning;
-
-  while ( *isRunning == true ) {
-    pthread_testcancel();
-    object->callbackEvent();
-  }
-
-  pthread_exit( NULL );
-}
-
-//******************** End of __LINUX_OSS__ *********************//
-#endif
-
-
-// *************************************************** //
-//
-// Protected common (OS-independent) RtAudio methods.
-//
-// *************************************************** //
-
-// This method can be modified to control the behavior of error
-// message printing.
-void RtApi :: error( RtError::Type type )
-{
-  errorStream_.str(""); // clear the ostringstream
-  if ( type == RtError::WARNING && showWarnings_ == true )
-    std::cerr << '\n' << errorText_ << "\n\n";
-  else
-    throw( RtError( errorText_, type ) );
-}
-
-void RtApi :: verifyStream()
-{
-  if ( stream_.state == STREAM_CLOSED ) {
-    errorText_ = "RtApi:: a stream is not open!";
-    error( RtError::INVALID_USE );
-  }
-}
-
-void RtApi :: clearStreamInfo()
-{
-  stream_.mode = UNINITIALIZED;
-  stream_.state = STREAM_CLOSED;
-  stream_.sampleRate = 0;
-  stream_.bufferSize = 0;
-  stream_.nBuffers = 0;
-  stream_.userFormat = 0;
-  stream_.userInterleaved = true;
-  stream_.streamTime = 0.0;
-  stream_.apiHandle = 0;
-  stream_.deviceBuffer = 0;
-  stream_.callbackInfo.callback = 0;
-  stream_.callbackInfo.userData = 0;
-  stream_.callbackInfo.isRunning = false;
-  for ( int i=0; i<2; i++ ) {
-    stream_.device[i] = 11111;
-    stream_.doConvertBuffer[i] = false;
-    stream_.deviceInterleaved[i] = true;
-    stream_.doByteSwap[i] = false;
-    stream_.nUserChannels[i] = 0;
-    stream_.nDeviceChannels[i] = 0;
-    stream_.channelOffset[i] = 0;
-    stream_.deviceFormat[i] = 0;
-    stream_.latency[i] = 0;
-    stream_.userBuffer[i] = 0;
-    stream_.convertInfo[i].channels = 0;
-    stream_.convertInfo[i].inJump = 0;
-    stream_.convertInfo[i].outJump = 0;
-    stream_.convertInfo[i].inFormat = 0;
-    stream_.convertInfo[i].outFormat = 0;
-    stream_.convertInfo[i].inOffset.clear();
-    stream_.convertInfo[i].outOffset.clear();
-  }
-}
-
-unsigned int RtApi :: formatBytes( RtAudioFormat format )
-{
-  if ( format == RTAUDIO_SINT16 )
-    return 2;
-  else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 ||
-            format == RTAUDIO_FLOAT32 )
-    return 4;
-  else if ( format == RTAUDIO_FLOAT64 )
-    return 8;
-  else if ( format == RTAUDIO_SINT8 )
-    return 1;
-
-  errorText_ = "RtApi::formatBytes: undefined format.";
-  error( RtError::WARNING );
-
-  return 0;
-}
-
-void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
-{
-  if ( mode == INPUT ) { // convert device to user buffer
-    stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
-    stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
-    stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
-    stream_.convertInfo[mode].outFormat = stream_.userFormat;
-  }
-  else { // convert user to device buffer
-    stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
-    stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
-    stream_.convertInfo[mode].inFormat = stream_.userFormat;
-    stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
-  }
-
-  if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
-    stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
-  else
-    stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
-
-  // Set up the interleave/deinterleave offsets.
-  if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
-    if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
-         ( mode == INPUT && stream_.userInterleaved ) ) {
-      for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
-        stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
-        stream_.convertInfo[mode].outOffset.push_back( k );
-        stream_.convertInfo[mode].inJump = 1;
-      }
-    }
-    else {
-      for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
-        stream_.convertInfo[mode].inOffset.push_back( k );
-        stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
-        stream_.convertInfo[mode].outJump = 1;
-      }
-    }
-  }
-  else { // no (de)interleaving
-    if ( stream_.userInterleaved ) {
-      for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
-        stream_.convertInfo[mode].inOffset.push_back( k );
-        stream_.convertInfo[mode].outOffset.push_back( k );
-      }
-    }
-    else {
-      for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
-        stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
-        stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
-        stream_.convertInfo[mode].inJump = 1;
-        stream_.convertInfo[mode].outJump = 1;
-      }
-    }
-  }
-
-  // Add channel offset.
-  if ( firstChannel > 0 ) {
-    if ( stream_.deviceInterleaved[mode] ) {
-      if ( mode == OUTPUT ) {
-        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
-          stream_.convertInfo[mode].outOffset[k] += firstChannel;
-      }
-      else {
-        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
-          stream_.convertInfo[mode].inOffset[k] += firstChannel;
-      }
-    }
-    else {
-      if ( mode == OUTPUT ) {
-        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
-          stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
-      }
-      else {
-        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
-          stream_.convertInfo[mode].inOffset[k] += ( firstChannel  * stream_.bufferSize );
-      }
-    }
-  }
-}
-
-void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
-{
-  // This function does format conversion, input/output channel compensation, and
-  // data interleaving/deinterleaving.  24-bit integers are assumed to occupy
-  // the lower three bytes of a 32-bit integer.
-
-  // Clear our device buffer when in/out duplex device channels are different
-  if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
-       ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
-    memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
-
-  int j;
-  if (info.outFormat == RTAUDIO_FLOAT64) {
-    Float64 scale;
-    Float64 *out = (Float64 *)outBuffer;
-
-    if (info.inFormat == RTAUDIO_SINT8) {
-      signed char *in = (signed char *)inBuffer;
-      scale = 1.0 / 127.5;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
-          out[info.outOffset[j]] += 0.5;
-          out[info.outOffset[j]] *= scale;
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT16) {
-      Int16 *in = (Int16 *)inBuffer;
-      scale = 1.0 / 32767.5;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
-          out[info.outOffset[j]] += 0.5;
-          out[info.outOffset[j]] *= scale;
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT24) {
-      Int32 *in = (Int32 *)inBuffer;
-      scale = 1.0 / 8388607.5;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]] & 0x00ffffff);
-          out[info.outOffset[j]] += 0.5;
-          out[info.outOffset[j]] *= scale;
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT32) {
-      Int32 *in = (Int32 *)inBuffer;
-      scale = 1.0 / 2147483647.5;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
-          out[info.outOffset[j]] += 0.5;
-          out[info.outOffset[j]] *= scale;
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT32) {
-      Float32 *in = (Float32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT64) {
-      // Channel compensation and/or (de)interleaving only.
-      Float64 *in = (Float64 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = in[info.inOffset[j]];
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-  }
-  else if (info.outFormat == RTAUDIO_FLOAT32) {
-    Float32 scale;
-    Float32 *out = (Float32 *)outBuffer;
-
-    if (info.inFormat == RTAUDIO_SINT8) {
-      signed char *in = (signed char *)inBuffer;
-      scale = (Float32) ( 1.0 / 127.5 );
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
-          out[info.outOffset[j]] += 0.5;
-          out[info.outOffset[j]] *= scale;
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT16) {
-      Int16 *in = (Int16 *)inBuffer;
-      scale = (Float32) ( 1.0 / 32767.5 );
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
-          out[info.outOffset[j]] += 0.5;
-          out[info.outOffset[j]] *= scale;
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT24) {
-      Int32 *in = (Int32 *)inBuffer;
-      scale = (Float32) ( 1.0 / 8388607.5 );
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]] & 0x00ffffff);
-          out[info.outOffset[j]] += 0.5;
-          out[info.outOffset[j]] *= scale;
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT32) {
-      Int32 *in = (Int32 *)inBuffer;
-      scale = (Float32) ( 1.0 / 2147483647.5 );
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
-          out[info.outOffset[j]] += 0.5;
-          out[info.outOffset[j]] *= scale;
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT32) {
-      // Channel compensation and/or (de)interleaving only.
-      Float32 *in = (Float32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = in[info.inOffset[j]];
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT64) {
-      Float64 *in = (Float64 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-  }
-  else if (info.outFormat == RTAUDIO_SINT32) {
-    Int32 *out = (Int32 *)outBuffer;
-    if (info.inFormat == RTAUDIO_SINT8) {
-      signed char *in = (signed char *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
-          out[info.outOffset[j]] <<= 24;
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT16) {
-      Int16 *in = (Int16 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
-          out[info.outOffset[j]] <<= 16;
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT24) { // Hmmm ... we could just leave it in the lower 3 bytes
-      Int32 *in = (Int32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
-          out[info.outOffset[j]] <<= 8;
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT32) {
-      // Channel compensation and/or (de)interleaving only.
-      Int32 *in = (Int32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = in[info.inOffset[j]];
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT32) {
-      Float32 *in = (Float32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT64) {
-      Float64 *in = (Float64 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-  }
-  else if (info.outFormat == RTAUDIO_SINT24) {
-    Int32 *out = (Int32 *)outBuffer;
-    if (info.inFormat == RTAUDIO_SINT8) {
-      signed char *in = (signed char *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
-          out[info.outOffset[j]] <<= 16;
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT16) {
-      Int16 *in = (Int16 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
-          out[info.outOffset[j]] <<= 8;
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT24) {
-      // Channel compensation and/or (de)interleaving only.
-      Int32 *in = (Int32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = in[info.inOffset[j]];
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT32) {
-      Int32 *in = (Int32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
-          out[info.outOffset[j]] >>= 8;
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT32) {
-      Float32 *in = (Float32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT64) {
-      Float64 *in = (Float64 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-  }
-  else if (info.outFormat == RTAUDIO_SINT16) {
-    Int16 *out = (Int16 *)outBuffer;
-    if (info.inFormat == RTAUDIO_SINT8) {
-      signed char *in = (signed char *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
-          out[info.outOffset[j]] <<= 8;
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT16) {
-      // Channel compensation and/or (de)interleaving only.
-      Int16 *in = (Int16 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = in[info.inOffset[j]];
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT24) {
-      Int32 *in = (Int32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 8) & 0x0000ffff);
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT32) {
-      Int32 *in = (Int32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT32) {
-      Float32 *in = (Float32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT64) {
-      Float64 *in = (Float64 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-  }
-  else if (info.outFormat == RTAUDIO_SINT8) {
-    signed char *out = (signed char *)outBuffer;
-    if (info.inFormat == RTAUDIO_SINT8) {
-      // Channel compensation and/or (de)interleaving only.
-      signed char *in = (signed char *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = in[info.inOffset[j]];
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    if (info.inFormat == RTAUDIO_SINT16) {
-      Int16 *in = (Int16 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT24) {
-      Int32 *in = (Int32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 16) & 0x000000ff);
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_SINT32) {
-      Int32 *in = (Int32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT32) {
-      Float32 *in = (Float32 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-    else if (info.inFormat == RTAUDIO_FLOAT64) {
-      Float64 *in = (Float64 *)inBuffer;
-      for (unsigned int i=0; i<stream_.bufferSize; i++) {
-        for (j=0; j<info.channels; j++) {
-          out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
-        }
-        in += info.inJump;
-        out += info.outJump;
-      }
-    }
-  }
-}
-
-  //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
-  //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
-  //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
-
-void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
-{
-  register char val;
-  register char *ptr;
-
-  ptr = buffer;
-  if ( format == RTAUDIO_SINT16 ) {
-    for ( unsigned int i=0; i<samples; i++ ) {
-      // Swap 1st and 2nd bytes.
-      val = *(ptr);
-      *(ptr) = *(ptr+1);
-      *(ptr+1) = val;
-
-      // Increment 2 bytes.
-      ptr += 2;
-    }
-  }
-  else if ( format == RTAUDIO_SINT24 ||
-            format == RTAUDIO_SINT32 ||
-            format == RTAUDIO_FLOAT32 ) {
-    for ( unsigned int i=0; i<samples; i++ ) {
-      // Swap 1st and 4th bytes.
-      val = *(ptr);
-      *(ptr) = *(ptr+3);
-      *(ptr+3) = val;
-
-      // Swap 2nd and 3rd bytes.
-      ptr += 1;
-      val = *(ptr);
-      *(ptr) = *(ptr+1);
-      *(ptr+1) = val;
-
-      // Increment 3 more bytes.
-      ptr += 3;
-    }
-  }
-  else if ( format == RTAUDIO_FLOAT64 ) {
-    for ( unsigned int i=0; i<samples; i++ ) {
-      // Swap 1st and 8th bytes
-      val = *(ptr);
-      *(ptr) = *(ptr+7);
-      *(ptr+7) = val;
-
-      // Swap 2nd and 7th bytes
-      ptr += 1;
-      val = *(ptr);
-      *(ptr) = *(ptr+5);
-      *(ptr+5) = val;
-
-      // Swap 3rd and 6th bytes
-      ptr += 1;
-      val = *(ptr);
-      *(ptr) = *(ptr+3);
-      *(ptr+3) = val;
-
-      // Swap 4th and 5th bytes
-      ptr += 1;
-      val = *(ptr);
-      *(ptr) = *(ptr+1);
-      *(ptr+1) = val;
-
-      // Increment 5 more bytes.
-      ptr += 5;
-    }
-  }
-}
-
-  // Indentation settings for Vim and Emacs
-  //
-  // Local Variables:
-  // c-basic-offset: 2
-  // indent-tabs-mode: nil
-  // End:
-  //
-  // vim: et sts=2 sw=2
-
+/************************************************************************/

+/*! \class RtAudio

+    \brief Realtime audio i/o C++ classes.

+

+    RtAudio provides a common API (Application Programming Interface)

+    for realtime audio input/output across Linux (native ALSA, Jack,

+    and OSS), Macintosh OS X (CoreAudio and Jack), and Windows

+    (DirectSound and ASIO) operating systems.

+

+    RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/

+

+    RtAudio: realtime audio i/o C++ classes

+    Copyright (c) 2001-2011 Gary P. Scavone

+

+    Permission is hereby granted, free of charge, to any person

+    obtaining a copy of this software and associated documentation files

+    (the "Software"), to deal in the Software without restriction,

+    including without limitation the rights to use, copy, modify, merge,

+    publish, distribute, sublicense, and/or sell copies of the Software,

+    and to permit persons to whom the Software is furnished to do so,

+    subject to the following conditions:

+

+    The above copyright notice and this permission notice shall be

+    included in all copies or substantial portions of the Software.

+

+    Any person wishing to distribute modifications to the Software is

+    asked to send the modifications to the original developer so that

+    they can be incorporated into the canonical version.  This is,

+    however, not a binding provision of this license.

+

+    THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,

+    EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF

+    MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.

+    IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR

+    ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF

+    CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION

+    WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.

+*/

+/************************************************************************/

+

+// RtAudio: Version 4.0.8

+

+#include "RtAudio.h"

+#include <iostream>

+#include <cstdlib>

+#include <cstring>

+#include <climits>

+

+// Static variable definitions.

+const unsigned int RtApi::MAX_SAMPLE_RATES = 14;

+const unsigned int RtApi::SAMPLE_RATES[] = {

+  4000, 5512, 8000, 9600, 11025, 16000, 22050,

+  32000, 44100, 48000, 88200, 96000, 176400, 192000

+};

+

+#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__)

+  #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)

+  #define MUTEX_DESTROY(A)    DeleteCriticalSection(A)

+  #define MUTEX_LOCK(A)       EnterCriticalSection(A)

+  #define MUTEX_UNLOCK(A)     LeaveCriticalSection(A)

+#elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)

+  // pthread API

+  #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)

+  #define MUTEX_DESTROY(A)    pthread_mutex_destroy(A)

+  #define MUTEX_LOCK(A)       pthread_mutex_lock(A)

+  #define MUTEX_UNLOCK(A)     pthread_mutex_unlock(A)

+#else

+  #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions

+  #define MUTEX_DESTROY(A)    abs(*A) // dummy definitions

+#endif

+

+// *************************************************** //

+//

+// RtAudio definitions.

+//

+// *************************************************** //

+

+void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()

+{

+  apis.clear();

+

+  // The order here will control the order of RtAudio's API search in

+  // the constructor.

+#if defined(__UNIX_JACK__)

+  apis.push_back( UNIX_JACK );

+#endif

+#if defined(__LINUX_ALSA__)

+  apis.push_back( LINUX_ALSA );

+#endif

+#if defined(__LINUX_OSS__)

+  apis.push_back( LINUX_OSS );

+#endif

+#if defined(__WINDOWS_ASIO__)

+  apis.push_back( WINDOWS_ASIO );

+#endif

+#if defined(__WINDOWS_DS__)

+  apis.push_back( WINDOWS_DS );

+#endif

+#if defined(__MACOSX_CORE__)

+  apis.push_back( MACOSX_CORE );

+#endif

+#if defined(__RTAUDIO_DUMMY__)

+  apis.push_back( RTAUDIO_DUMMY );

+#endif

+}

+

+void RtAudio :: openRtApi( RtAudio::Api api )

+{

+#if defined(__UNIX_JACK__)

+  if ( api == UNIX_JACK )

+    rtapi_ = new RtApiJack();

+#endif

+#if defined(__LINUX_ALSA__)

+  if ( api == LINUX_ALSA )

+    rtapi_ = new RtApiAlsa();

+#endif

+#if defined(__LINUX_OSS__)

+  if ( api == LINUX_OSS )

+    rtapi_ = new RtApiOss();

+#endif

+#if defined(__WINDOWS_ASIO__)

+  if ( api == WINDOWS_ASIO )

+    rtapi_ = new RtApiAsio();

+#endif

+#if defined(__WINDOWS_DS__)

+  if ( api == WINDOWS_DS )

+    rtapi_ = new RtApiDs();

+#endif

+#if defined(__MACOSX_CORE__)

+  if ( api == MACOSX_CORE )

+    rtapi_ = new RtApiCore();

+#endif

+#if defined(__RTAUDIO_DUMMY__)

+  if ( api == RTAUDIO_DUMMY )

+    rtapi_ = new RtApiDummy();

+#endif

+}

+

+RtAudio :: RtAudio( RtAudio::Api api ) throw()

+{

+  rtapi_ = 0;

+

+  if ( api != UNSPECIFIED ) {

+    // Attempt to open the specified API.

+    openRtApi( api );

+    if ( rtapi_ ) return;

+

+    // No compiled support for specified API value.  Issue a debug

+    // warning and continue as if no API was specified.

+    std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;

+  }

+

+  // Iterate through the compiled APIs and return as soon as we find

+  // one with at least one device or we reach the end of the list.

+  std::vector< RtAudio::Api > apis;

+  getCompiledApi( apis );

+  for ( unsigned int i=0; i<apis.size(); i++ ) {

+    openRtApi( apis[i] );

+    if ( rtapi_->getDeviceCount() ) break;

+  }

+

+  if ( rtapi_ ) return;

+

+  // It should not be possible to get here because the preprocessor

+  // definition __RTAUDIO_DUMMY__ is automatically defined if no

+  // API-specific definitions are passed to the compiler. But just in

+  // case something weird happens, we'll print out an error message.

+  std::cerr << "\nRtAudio: no compiled API support found ... critical error!!\n\n";

+}

+

+RtAudio :: ~RtAudio() throw()

+{

+  delete rtapi_;

+}

+

+void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,

+                            RtAudio::StreamParameters *inputParameters,

+                            RtAudioFormat format, unsigned int sampleRate,

+                            unsigned int *bufferFrames,

+                            RtAudioCallback callback, void *userData,

+                            RtAudio::StreamOptions *options )

+{

+  return rtapi_->openStream( outputParameters, inputParameters, format,

+                             sampleRate, bufferFrames, callback,

+                             userData, options );

+}

+

+// *************************************************** //

+//

+// Public RtApi definitions (see end of file for

+// private or protected utility functions).

+//

+// *************************************************** //

+

+RtApi :: RtApi()

+{

+  stream_.state = STREAM_CLOSED;

+  stream_.mode = UNINITIALIZED;

+  stream_.apiHandle = 0;

+  stream_.userBuffer[0] = 0;

+  stream_.userBuffer[1] = 0;

+  MUTEX_INITIALIZE( &stream_.mutex );

+  showWarnings_ = true;

+}

+

+RtApi :: ~RtApi()

+{

+  MUTEX_DESTROY( &stream_.mutex );

+}

+

+void RtApi :: openStream( RtAudio::StreamParameters *oParams,

+                          RtAudio::StreamParameters *iParams,

+                          RtAudioFormat format, unsigned int sampleRate,

+                          unsigned int *bufferFrames,

+                          RtAudioCallback callback, void *userData,

+                          RtAudio::StreamOptions *options )

+{

+  if ( stream_.state != STREAM_CLOSED ) {

+    errorText_ = "RtApi::openStream: a stream is already open!";

+    error( RtError::INVALID_USE );

+  }

+

+  if ( oParams && oParams->nChannels < 1 ) {

+    errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";

+    error( RtError::INVALID_USE );

+  }

+

+  if ( iParams && iParams->nChannels < 1 ) {

+    errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";

+    error( RtError::INVALID_USE );

+  }

+

+  if ( oParams == NULL && iParams == NULL ) {

+    errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";

+    error( RtError::INVALID_USE );

+  }

+

+  if ( formatBytes(format) == 0 ) {

+    errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";

+    error( RtError::INVALID_USE );

+  }

+

+  unsigned int nDevices = getDeviceCount();

+  unsigned int oChannels = 0;

+  if ( oParams ) {

+    oChannels = oParams->nChannels;

+    if ( oParams->deviceId >= nDevices ) {

+      errorText_ = "RtApi::openStream: output device parameter value is invalid.";

+      error( RtError::INVALID_USE );

+    }

+  }

+

+  unsigned int iChannels = 0;

+  if ( iParams ) {

+    iChannels = iParams->nChannels;

+    if ( iParams->deviceId >= nDevices ) {

+      errorText_ = "RtApi::openStream: input device parameter value is invalid.";

+      error( RtError::INVALID_USE );

+    }

+  }

+

+  clearStreamInfo();

+  bool result;

+

+  if ( oChannels > 0 ) {

+

+    result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,

+                              sampleRate, format, bufferFrames, options );

+    if ( result == false ) error( RtError::SYSTEM_ERROR );

+  }

+

+  if ( iChannels > 0 ) {

+

+    result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,

+                              sampleRate, format, bufferFrames, options );

+    if ( result == false ) {

+      if ( oChannels > 0 ) closeStream();

+      error( RtError::SYSTEM_ERROR );

+    }

+  }

+

+  stream_.callbackInfo.callback = (void *) callback;

+  stream_.callbackInfo.userData = userData;

+

+  if ( options ) options->numberOfBuffers = stream_.nBuffers;

+  stream_.state = STREAM_STOPPED;

+}

+

+unsigned int RtApi :: getDefaultInputDevice( void )

+{

+  // Should be implemented in subclasses if possible.

+  return 0;

+}

+

+unsigned int RtApi :: getDefaultOutputDevice( void )

+{

+  // Should be implemented in subclasses if possible.

+  return 0;

+}

+

+void RtApi :: closeStream( void )

+{

+  // MUST be implemented in subclasses!

+  return;

+}

+

+bool RtApi :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,

+                               unsigned int firstChannel, unsigned int sampleRate,

+                               RtAudioFormat format, unsigned int *bufferSize,

+                               RtAudio::StreamOptions *options )

+{

+  // MUST be implemented in subclasses!

+  return FAILURE;

+}

+

+void RtApi :: tickStreamTime( void )

+{

+  // Subclasses that do not provide their own implementation of

+  // getStreamTime should call this function once per buffer I/O to

+  // provide basic stream time support.

+

+  stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );

+

+#if defined( HAVE_GETTIMEOFDAY )

+  gettimeofday( &stream_.lastTickTimestamp, NULL );

+#endif

+}

+

+long RtApi :: getStreamLatency( void )

+{

+  verifyStream();

+

+  long totalLatency = 0;

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )

+    totalLatency = stream_.latency[0];

+  if ( stream_.mode == INPUT || stream_.mode == DUPLEX )

+    totalLatency += stream_.latency[1];

+

+  return totalLatency;

+}

+

+double RtApi :: getStreamTime( void )

+{

+  verifyStream();

+

+#if defined( HAVE_GETTIMEOFDAY )

+  // Return a very accurate estimate of the stream time by

+  // adding in the elapsed time since the last tick.

+  struct timeval then;

+  struct timeval now;

+

+  if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )

+    return stream_.streamTime;

+

+  gettimeofday( &now, NULL );

+  then = stream_.lastTickTimestamp;

+  return stream_.streamTime +

+    ((now.tv_sec + 0.000001 * now.tv_usec) -

+     (then.tv_sec + 0.000001 * then.tv_usec));     

+#else

+  return stream_.streamTime;

+#endif

+}

+

+unsigned int RtApi :: getStreamSampleRate( void )

+{

+ verifyStream();

+

+ return stream_.sampleRate;

+}

+

+

+// *************************************************** //

+//

+// OS/API-specific methods.

+//

+// *************************************************** //

+

+#if defined(__MACOSX_CORE__)

+

+// The OS X CoreAudio API is designed to use a separate callback

+// procedure for each of its audio devices.  A single RtAudio duplex

+// stream using two different devices is supported here, though it

+// cannot be guaranteed to always behave correctly because we cannot

+// synchronize these two callbacks.

+//

+// A property listener is installed for over/underrun information.

+// However, no functionality is currently provided to allow property

+// listeners to trigger user handlers because it is unclear what could

+// be done if a critical stream parameter (buffer size, sample rate,

+// device disconnect) notification arrived.  The listeners entail

+// quite a bit of extra code and most likely, a user program wouldn't

+// be prepared for the result anyway.  However, we do provide a flag

+// to the client callback function to inform of an over/underrun.

+//

+// The mechanism for querying and setting system parameters was

+// updated (and perhaps simplified) in OS-X version 10.4.  However,

+// since 10.4 support is not necessarily available to all users, I've

+// decided not to update the respective code at this time.  Perhaps

+// this will happen when Apple makes 10.4 free for everyone. :-)

+

+// A structure to hold various information related to the CoreAudio API

+// implementation.

+struct CoreHandle {

+  AudioDeviceID id[2];    // device ids

+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )

+  AudioDeviceIOProcID procId[2];

+#endif

+  UInt32 iStream[2];      // device stream index (or first if using multiple)

+  UInt32 nStreams[2];     // number of streams to use

+  bool xrun[2];

+  char *deviceBuffer;

+  pthread_cond_t condition;

+  int drainCounter;       // Tracks callback counts when draining

+  bool internalDrain;     // Indicates if stop is initiated from callback or not.

+

+  CoreHandle()

+    :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }

+};

+

+RtApiCore :: RtApiCore()

+{

+  // Nothing to do here.

+}

+

+RtApiCore :: ~RtApiCore()

+{

+  // The subclass destructor gets called before the base class

+  // destructor, so close an existing stream before deallocating

+  // apiDeviceId memory.

+  if ( stream_.state != STREAM_CLOSED ) closeStream();

+}

+

+unsigned int RtApiCore :: getDeviceCount( void )

+{

+  // Find out how many audio devices there are, if any.

+  UInt32 dataSize;

+  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( RtError::WARNING );

+    return 0;

+  }

+

+  return dataSize / sizeof( AudioDeviceID );

+}

+

+unsigned int RtApiCore :: getDefaultInputDevice( void )

+{

+  unsigned int nDevices = getDeviceCount();

+  if ( nDevices <= 1 ) return 0;

+

+  AudioDeviceID id;

+  UInt32 dataSize = sizeof( AudioDeviceID );

+  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( RtError::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( RtError::WARNING );

+    return 0;

+  }

+

+  for ( unsigned int i=0; i<nDevices; i++ )

+    if ( id == deviceList[i] ) return i;

+

+  errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";

+  error( RtError::WARNING );

+  return 0;

+}

+

+unsigned int RtApiCore :: getDefaultOutputDevice( void )

+{

+  unsigned int nDevices = getDeviceCount();

+  if ( nDevices <= 1 ) return 0;

+

+  AudioDeviceID id;

+  UInt32 dataSize = sizeof( AudioDeviceID );

+  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( RtError::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( RtError::WARNING );

+    return 0;

+  }

+

+  for ( unsigned int i=0; i<nDevices; i++ )

+    if ( id == deviceList[i] ) return i;

+

+  errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";

+  error( RtError::WARNING );

+  return 0;

+}

+

+RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )

+{

+  RtAudio::DeviceInfo info;

+  info.probed = false;

+

+  // Get device ID

+  unsigned int nDevices = getDeviceCount();

+  if ( nDevices == 0 ) {

+    errorText_ = "RtApiCore::getDeviceInfo: no devices found!";

+    error( RtError::INVALID_USE );

+  }

+

+  if ( device >= nDevices ) {

+    errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";

+    error( RtError::INVALID_USE );

+  }

+

+  AudioDeviceID deviceList[ nDevices ];

+  UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;

+  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( RtError::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( RtError::WARNING );

+    return info;

+  }

+

+  //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );

+  int length = CFStringGetLength(cfname);

+  char *mname = (char *)malloc(length * 3 + 1);

+  CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());

+  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( RtError::WARNING );

+    return info;

+  }

+

+  //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );

+  length = CFStringGetLength(cfname);

+  char *name = (char *)malloc(length * 3 + 1);

+  CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());

+  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( RtError::WARNING );

+    return info;

+  }

+

+  // Allocate the AudioBufferList.

+  bufferList = (AudioBufferList *) malloc( dataSize );

+  if ( bufferList == NULL ) {

+    errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";

+    error( RtError::WARNING );

+    return info;

+  }

+

+  result = 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( RtError::WARNING );

+    return info;

+  }

+

+  // Get output channel information.

+  unsigned int i, nStreams = bufferList->mNumberBuffers;

+  for ( i=0; i<nStreams; i++ )

+    info.outputChannels += bufferList->mBuffers[i].mNumberChannels;

+  free( bufferList );

+

+  // Get the input stream "configuration".

+  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( RtError::WARNING );

+    return info;

+  }

+

+  // Allocate the AudioBufferList.

+  bufferList = (AudioBufferList *) malloc( dataSize );

+  if ( bufferList == NULL ) {

+    errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";

+    error( RtError::WARNING );

+    return info;

+  }

+

+  result = 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( RtError::WARNING );

+    return info;

+  }

+

+  // Get input channel information.

+  nStreams = bufferList->mNumberBuffers;

+  for ( i=0; i<nStreams; i++ )

+    info.inputChannels += bufferList->mBuffers[i].mNumberChannels;

+  free( bufferList );

+

+  // If device opens for both playback and capture, we determine the channels.

+  if ( info.outputChannels > 0 && info.inputChannels > 0 )

+    info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;

+

+  // Probe the device sample rates.

+  bool isInput = false;

+  if ( info.outputChannels == 0 ) isInput = true;

+

+  // Determine the supported sample rates.

+  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( RtError::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( RtError::WARNING );

+    return info;

+  }

+

+  Float64 minimumRate = 100000000.0, maximumRate = 0.0;

+  for ( UInt32 i=0; i<nRanges; i++ ) {

+    if ( rangeList[i].mMinimum < minimumRate ) minimumRate = rangeList[i].mMinimum;

+    if ( rangeList[i].mMaximum > maximumRate ) maximumRate = rangeList[i].mMaximum;

+  }

+

+  info.sampleRates.clear();

+  for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {

+    if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate )

+      info.sampleRates.push_back( SAMPLE_RATES[k] );

+  }

+

+  if ( info.sampleRates.size() == 0 ) {

+    errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // CoreAudio always uses 32-bit floating point data for PCM streams.

+  // Thus, any other "physical" formats supported by the device are of

+  // no interest to the client.

+  info.nativeFormats = RTAUDIO_FLOAT32;

+

+  if ( info.outputChannels > 0 )

+    if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;

+  if ( info.inputChannels > 0 )

+    if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;

+

+  info.probed = true;

+  return info;

+}

+

+OSStatus callbackHandler( AudioDeviceID inDevice,

+                          const AudioTimeStamp* inNow,

+                          const AudioBufferList* inInputData,

+                          const AudioTimeStamp* inInputTime,

+                          AudioBufferList* outOutputData,

+                          const AudioTimeStamp* inOutputTime, 

+                          void* infoPointer )

+{

+  CallbackInfo *info = (CallbackInfo *) infoPointer;

+

+  RtApiCore *object = (RtApiCore *) info->object;

+  if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )

+    return kAudioHardwareUnspecifiedError;

+  else

+    return kAudioHardwareNoError;

+}

+

+OSStatus deviceListener( 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;

+}

+

+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) {

+    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;

+

+  // 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 ) {

+    nominalRate = (Float64) sampleRate;

+    result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );

+

+    if ( result != noErr ) {

+      errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+  }

+

+  // 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;

+      }

+    }

+  }

+

+  // Get the stream ID(s) so we can set the stream format.

+  AudioStreamID streamIDs[ nStreams ];

+  dataSize = nStreams * sizeof( AudioStreamID );

+  property.mSelector = kAudioDevicePropertyStreams;

+  result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &streamIDs );

+

+  if ( result != noErr ) {

+    errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream ID(s) for device (" << device << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Now set the stream format for each stream.  Also, check the

+  // physical format of the device and change that if necessary.

+  AudioStreamBasicDescription	description;

+  dataSize = sizeof( AudioStreamBasicDescription );

+

+  bool updateFormat;

+  for ( UInt32 i=0; i<streamCount; i++ ) {

+

+    property.mSelector = kAudioStreamPropertyVirtualFormat;

+    result = AudioObjectGetPropertyData( streamIDs[firstStream+i], &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.

+    updateFormat = false;

+    if ( fabs( description.mSampleRate - (double)sampleRate ) > 1.0 ) {

+      description.mSampleRate = (double) sampleRate;

+      updateFormat = true;

+    }

+

+    if ( description.mFormatID != kAudioFormatLinearPCM ) {

+      description.mFormatID = kAudioFormatLinearPCM;

+      updateFormat = true;

+    }

+

+    if ( updateFormat ) {

+      result = AudioObjectSetPropertyData( streamIDs[firstStream+i], &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( streamIDs[firstStream+i], &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;

+    }

+

+    if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 24 ) {

+      description.mFormatID = kAudioFormatLinearPCM;

+      AudioStreamBasicDescription	testDescription = description;

+      unsigned long formatFlags;

+

+      // We'll try higher bit rates first and then work our way down.

+      testDescription.mBitsPerChannel = 32;

+      testDescription.mBytesPerFrame =  testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;

+      testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;

+      formatFlags = description.mFormatFlags | kLinearPCMFormatFlagIsFloat & ~kLinearPCMFormatFlagIsSignedInteger;

+      testDescription.mFormatFlags = formatFlags;

+      result = AudioObjectSetPropertyData( streamIDs[firstStream+i], &property, 0, NULL, dataSize, &testDescription );

+      if ( result == noErr ) continue;

+

+      testDescription = description;

+      testDescription.mBitsPerChannel = 32;

+      testDescription.mBytesPerFrame =  testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;

+      testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;

+      formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger) & ~kLinearPCMFormatFlagIsFloat;

+      testDescription.mFormatFlags = formatFlags;

+      result = AudioObjectSetPropertyData( streamIDs[firstStream+i], &property, 0, NULL, dataSize, &testDescription );

+      if ( result == noErr ) continue;

+

+      testDescription = description;

+      testDescription.mBitsPerChannel = 24;

+      testDescription.mBytesPerFrame =  testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;

+      testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;

+      testDescription.mFormatFlags = formatFlags;

+      result = AudioObjectSetPropertyData( streamIDs[firstStream+i], &property, 0, NULL, dataSize, &testDescription );

+      if ( result == noErr ) continue;

+

+      testDescription = description;

+      testDescription.mBitsPerChannel = 16;

+      testDescription.mBytesPerFrame =  testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;

+      testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;

+      testDescription.mFormatFlags = formatFlags;

+      result = AudioObjectSetPropertyData( streamIDs[firstStream+i], &property, 0, NULL, dataSize, &testDescription );

+      if ( result == noErr ) continue;

+

+      testDescription = description;

+      testDescription.mBitsPerChannel = 8;

+      testDescription.mBytesPerFrame =  testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;

+      testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;

+      testDescription.mFormatFlags = formatFlags;

+      result = AudioObjectSetPropertyData( streamIDs[firstStream+i], &property, 0, NULL, dataSize, &testDescription );

+      if ( result != noErr ) {

+        errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";

+        errorText_ = errorStream_.str();

+        return FAILURE;

+      }

+    }

+  }

+

+

+  // Get the stream latency.  There can be latency in both the device

+  // and the stream.  First, attempt to get the device latency on the

+  // master channel or the first open channel.  Errors that might

+  // occur here are not deemed critical.

+

+  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( RtError::WARNING );

+    }

+  }

+

+  // Now try to get the stream latency.  For multiple streams, I assume the

+  // latency is equal for each.

+  result = AudioObjectGetPropertyData( streamIDs[firstStream], &property, 0, NULL, &dataSize, &latency );

+  if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] += latency;

+  else {

+    errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream latency for device (" << device << ").";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+  }

+

+  // Byte-swapping: According to AudioHardware.h, the stream data will

+  // always be presented in native-endian format, so we should never

+  // need to byte swap.

+  stream_.doByteSwap[mode] = false;

+

+  // From the CoreAudio documentation, PCM data must be supplied as

+  // 32-bit floats.

+  stream_.userFormat = format;

+  stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;

+

+  if ( streamCount == 1 )

+    stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;

+  else // multiple streams

+    stream_.nDeviceChannels[mode] = channels;

+  stream_.nUserChannels[mode] = channels;

+  stream_.channelOffset[mode] = channelOffset;  // offset within a CoreAudio stream

+  if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;

+  else stream_.userInterleaved = true;

+  stream_.deviceInterleaved[mode] = true;

+  if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;

+

+  // Set flags for buffer conversion.

+  stream_.doConvertBuffer[mode] = false;

+  if ( stream_.userFormat != stream_.deviceFormat[mode] )

+    stream_.doConvertBuffer[mode] = true;

+  if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )

+    stream_.doConvertBuffer[mode] = true;

+  if ( streamCount == 1 ) {

+    if ( stream_.nUserChannels[mode] > 1 &&

+         stream_.userInterleaved != stream_.deviceInterleaved[mode] )

+      stream_.doConvertBuffer[mode] = true;

+  }

+  else if ( monoMode && stream_.userInterleaved )

+    stream_.doConvertBuffer[mode] = true;

+

+  // Allocate our CoreHandle structure for the stream.

+  CoreHandle *handle = 0;

+  if ( stream_.apiHandle == 0 ) {

+    try {

+      handle = new CoreHandle;

+    }

+    catch ( std::bad_alloc& ) {

+      errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";

+      goto error;

+    }

+

+    if ( pthread_cond_init( &handle->condition, NULL ) ) {

+      errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";

+      goto error;

+    }

+    stream_.apiHandle = (void *) handle;

+  }

+  else

+    handle = (CoreHandle *) stream_.apiHandle;

+  handle->iStream[mode] = firstStream;

+  handle->nStreams[mode] = streamCount;

+  handle->id[mode] = id;

+

+  // Allocate necessary internal buffers.

+  unsigned long bufferBytes;

+  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );

+  //  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );

+  stream_.userBuffer[mode] = (char *) malloc( bufferBytes );

+  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;

+  result = AudioObjectAddPropertyListener( id, &property, deviceListener, (void *) handle );

+

+  return SUCCESS;

+

+ error:

+  if ( handle ) {

+    pthread_cond_destroy( &handle->condition );

+    delete handle;

+    stream_.apiHandle = 0;

+  }

+

+  for ( int i=0; i<2; i++ ) {

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  return FAILURE;

+}

+

+void RtApiCore :: closeStream( void )

+{

+  if ( stream_.state == STREAM_CLOSED ) {

+    errorText_ = "RtApiCore::closeStream(): no open stream to close!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+    if ( stream_.state == STREAM_RUNNING )

+      AudioDeviceStop( handle->id[0], callbackHandler );

+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )

+    AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );

+#else

+    // deprecated in favor of AudioDeviceDestroyIOProcID()

+    AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );

+#endif

+  }

+

+  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {

+    if ( stream_.state == STREAM_RUNNING )

+      AudioDeviceStop( handle->id[1], callbackHandler );

+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )

+    AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );

+#else

+    // deprecated in favor of AudioDeviceDestroyIOProcID()

+    AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );

+#endif

+  }

+

+  for ( int i=0; i<2; i++ ) {

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  // Destroy pthread condition variable.

+  pthread_cond_destroy( &handle->condition );

+  delete handle;

+  stream_.apiHandle = 0;

+

+  stream_.mode = UNINITIALIZED;

+  stream_.state = STREAM_CLOSED;

+}

+

+void RtApiCore :: startStream( void )

+{

+  verifyStream();

+  if ( stream_.state == STREAM_RUNNING ) {

+    errorText_ = "RtApiCore::startStream(): the stream is already running!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  OSStatus result = noErr;

+  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+

+    result = AudioDeviceStart( handle->id[0], callbackHandler );

+    if ( result != noErr ) {

+      errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+  if ( stream_.mode == INPUT ||

+       ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {

+

+    result = AudioDeviceStart( handle->id[1], callbackHandler );

+    if ( result != noErr ) {

+      errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+  handle->drainCounter = 0;

+  handle->internalDrain = false;

+  stream_.state = STREAM_RUNNING;

+

+ unlock:

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  if ( result == noErr ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiCore :: stopStream( void )

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

+    errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_UNLOCK( &stream_.mutex );

+    return;

+  }

+

+  OSStatus result = noErr;

+  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+

+    if ( handle->drainCounter == 0 ) {

+      handle->drainCounter = 2;

+      pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled

+    }

+

+    MUTEX_UNLOCK( &stream_.mutex );

+    result = AudioDeviceStop( handle->id[0], callbackHandler );

+    MUTEX_LOCK( &stream_.mutex );

+    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] ) ) {

+

+    MUTEX_UNLOCK( &stream_.mutex );

+    result = AudioDeviceStop( handle->id[1], callbackHandler );

+    MUTEX_LOCK( &stream_.mutex );

+    if ( result != noErr ) {

+      errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+  stream_.state = STREAM_STOPPED;

+

+ unlock:

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  if ( result == noErr ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiCore :: abortStream( void )

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

+    errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;

+  handle->drainCounter = 2;

+

+  stopStream();

+}

+

+bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,

+                                 const AudioBufferList *inBufferList,

+                                 const AudioBufferList *outBufferList )

+{

+  if ( stream_.state == STREAM_STOPPED ) return SUCCESS;

+  if ( stream_.state == STREAM_CLOSED ) {

+    errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";

+    error( RtError::WARNING );

+    return FAILURE;

+  }

+

+  CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;

+  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;

+

+  // Check if we were draining the stream and signal is finished.

+  if ( handle->drainCounter > 3 ) {

+    if ( handle->internalDrain == true )

+      stopStream();

+    else // external call to stopStream()

+      pthread_cond_signal( &handle->condition );

+    return SUCCESS;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  // The state might change while waiting on a mutex.

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_UNLOCK( &stream_.mutex );

+    return SUCCESS;

+  }

+

+  AudioDeviceID outputDevice = handle->id[0];

+

+  // Invoke user callback to get fresh output data UNLESS we are

+  // draining stream or duplex mode AND the input/output devices are

+  // different AND this function is called for the input device.

+  if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {

+    RtAudioCallback callback = (RtAudioCallback) info->callback;

+    double streamTime = getStreamTime();

+    RtAudioStreamStatus status = 0;

+    if ( stream_.mode != INPUT && handle->xrun[0] == true ) {

+      status |= RTAUDIO_OUTPUT_UNDERFLOW;

+      handle->xrun[0] = false;

+    }

+    if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {

+      status |= RTAUDIO_INPUT_OVERFLOW;

+      handle->xrun[1] = false;

+    }

+

+    handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],

+                                     stream_.bufferSize, streamTime, status, info->userData );

+    if ( handle->drainCounter == 2 ) {

+      MUTEX_UNLOCK( &stream_.mutex );

+      abortStream();

+      return SUCCESS;

+    }

+    else if ( handle->drainCounter == 1 )

+      handle->internalDrain = true;

+  }

+

+  if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {

+

+    if ( handle->drainCounter > 1 ) { // write zeros to the output stream

+

+      if ( handle->nStreams[0] == 1 ) {

+        memset( outBufferList->mBuffers[handle->iStream[0]].mData,

+                0,

+                outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );

+      }

+      else { // fill multiple streams with zeros

+        for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {

+          memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,

+                  0,

+                  outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );

+        }

+      }

+    }

+    else if ( handle->nStreams[0] == 1 ) {

+      if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer

+        convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,

+                       stream_.userBuffer[0], stream_.convertInfo[0] );

+      }

+      else { // copy from user buffer

+        memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,

+                stream_.userBuffer[0],

+                outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );

+      }

+    }

+    else { // fill multiple streams

+      Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];

+      if ( stream_.doConvertBuffer[0] ) {

+        convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );

+        inBuffer = (Float32 *) stream_.deviceBuffer;

+      }

+

+      if ( stream_.deviceInterleaved[0] == false ) { // mono mode

+        UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;

+        for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {

+          memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,

+                  (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );

+        }

+      }

+      else { // fill multiple multi-channel streams with interleaved data

+        UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;

+        Float32 *out, *in;

+

+        bool inInterleaved = ( stream_.userInterleaved ) ? true : false;

+        UInt32 inChannels = stream_.nUserChannels[0];

+        if ( stream_.doConvertBuffer[0] ) {

+          inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode

+          inChannels = stream_.nDeviceChannels[0];

+        }

+

+        if ( inInterleaved ) inOffset = 1;

+        else inOffset = stream_.bufferSize;

+

+        channelsLeft = inChannels;

+        for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {

+          in = inBuffer;

+          out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;

+          streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;

+

+          outJump = 0;

+          // Account for possible channel offset in first stream

+          if ( i == 0 && stream_.channelOffset[0] > 0 ) {

+            streamChannels -= stream_.channelOffset[0];

+            outJump = stream_.channelOffset[0];

+            out += outJump;

+          }

+

+          // Account for possible unfilled channels at end of the last stream

+          if ( streamChannels > channelsLeft ) {

+            outJump = streamChannels - channelsLeft;

+            streamChannels = channelsLeft;

+          }

+

+          // Determine input buffer offsets and skips

+          if ( inInterleaved ) {

+            inJump = inChannels;

+            in += inChannels - channelsLeft;

+          }

+          else {

+            inJump = 1;

+            in += (inChannels - channelsLeft) * inOffset;

+          }

+

+          for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {

+            for ( unsigned int j=0; j<streamChannels; j++ ) {

+              *out++ = in[j*inOffset];

+            }

+            out += outJump;

+            in += inJump;

+          }

+          channelsLeft -= streamChannels;

+        }

+      }

+    }

+

+    if ( handle->drainCounter ) {

+      handle->drainCounter++;

+      goto unlock;

+    }

+  }

+

+  AudioDeviceID inputDevice;

+  inputDevice = handle->id[1];

+  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {

+

+    if ( handle->nStreams[1] == 1 ) {

+      if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer

+        convertBuffer( stream_.userBuffer[1],

+                       (char *) inBufferList->mBuffers[handle->iStream[1]].mData,

+                       stream_.convertInfo[1] );

+      }

+      else { // copy to user buffer

+        memcpy( stream_.userBuffer[1],

+                inBufferList->mBuffers[handle->iStream[1]].mData,

+                inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );

+      }

+    }

+    else { // read from multiple streams

+      Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];

+      if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;

+

+      if ( stream_.deviceInterleaved[1] == false ) { // mono mode

+        UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;

+        for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {

+          memcpy( (void *)&outBuffer[i*stream_.bufferSize],

+                  inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );

+        }

+      }

+      else { // read from multiple multi-channel streams

+        UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;

+        Float32 *out, *in;

+

+        bool outInterleaved = ( stream_.userInterleaved ) ? true : false;

+        UInt32 outChannels = stream_.nUserChannels[1];

+        if ( stream_.doConvertBuffer[1] ) {

+          outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode

+          outChannels = stream_.nDeviceChannels[1];

+        }

+

+        if ( outInterleaved ) outOffset = 1;

+        else outOffset = stream_.bufferSize;

+

+        channelsLeft = outChannels;

+        for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {

+          out = outBuffer;

+          in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;

+          streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;

+

+          inJump = 0;

+          // Account for possible channel offset in first stream

+          if ( i == 0 && stream_.channelOffset[1] > 0 ) {

+            streamChannels -= stream_.channelOffset[1];

+            inJump = stream_.channelOffset[1];

+            in += inJump;

+          }

+

+          // Account for possible unread channels at end of the last stream

+          if ( streamChannels > channelsLeft ) {

+            inJump = streamChannels - channelsLeft;

+            streamChannels = channelsLeft;

+          }

+

+          // Determine output buffer offsets and skips

+          if ( outInterleaved ) {

+            outJump = outChannels;

+            out += outChannels - channelsLeft;

+          }

+          else {

+            outJump = 1;

+            out += (outChannels - channelsLeft) * outOffset;

+          }

+

+          for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {

+            for ( unsigned int j=0; j<streamChannels; j++ ) {

+              out[j*outOffset] = *in++;

+            }

+            out += outJump;

+            in += inJump;

+          }

+          channelsLeft -= streamChannels;

+        }

+      }

+      

+      if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer

+        convertBuffer( stream_.userBuffer[1],

+                       stream_.deviceBuffer,

+                       stream_.convertInfo[1] );

+      }

+    }

+  }

+

+ unlock:

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  RtApi::tickStreamTime();

+  return SUCCESS;

+}

+

+const char* RtApiCore :: getErrorCode( OSStatus code )

+{

+  switch( code ) {

+

+  case kAudioHardwareNotRunningError:

+    return "kAudioHardwareNotRunningError";

+

+  case kAudioHardwareUnspecifiedError:

+    return "kAudioHardwareUnspecifiedError";

+

+  case kAudioHardwareUnknownPropertyError:

+    return "kAudioHardwareUnknownPropertyError";

+

+  case kAudioHardwareBadPropertySizeError:

+    return "kAudioHardwareBadPropertySizeError";

+

+  case kAudioHardwareIllegalOperationError:

+    return "kAudioHardwareIllegalOperationError";

+

+  case kAudioHardwareBadObjectError:

+    return "kAudioHardwareBadObjectError";

+

+  case kAudioHardwareBadDeviceError:

+    return "kAudioHardwareBadDeviceError";

+

+  case kAudioHardwareBadStreamError:

+    return "kAudioHardwareBadStreamError";

+

+  case kAudioHardwareUnsupportedOperationError:

+    return "kAudioHardwareUnsupportedOperationError";

+

+  case kAudioDeviceUnsupportedFormatError:

+    return "kAudioDeviceUnsupportedFormatError";

+

+  case kAudioDevicePermissionsError:

+    return "kAudioDevicePermissionsError";

+

+  default:

+    return "CoreAudio unknown error";

+  }

+}

+

+  //******************** End of __MACOSX_CORE__ *********************//

+#endif

+

+#if defined(__UNIX_JACK__)

+

+// JACK is a low-latency audio server, originally written for the

+// GNU/Linux operating system and now also ported to OS-X. It can

+// connect a number of different applications to an audio device, as

+// well as allowing them to share audio between themselves.

+//

+// When using JACK with RtAudio, "devices" refer to JACK clients that

+// have ports connected to the server.  The JACK server is typically

+// started in a terminal as follows:

+//

+// .jackd -d alsa -d hw:0

+//

+// or through an interface program such as qjackctl.  Many of the

+// parameters normally set for a stream are fixed by the JACK server

+// and can be specified when the JACK server is started.  In

+// particular,

+//

+// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4

+//

+// specifies a sample rate of 44100 Hz, a buffer size of 512 sample

+// frames, and number of buffers = 4.  Once the server is running, it

+// is not possible to override these values.  If the values are not

+// specified in the command-line, the JACK server uses default values.

+//

+// The JACK server does not have to be running when an instance of

+// RtApiJack is created, though the function getDeviceCount() will

+// report 0 devices found until JACK has been started.  When no

+// devices are available (i.e., the JACK server is not running), a

+// stream cannot be opened.

+

+#include <jack/jack.h>

+#include <unistd.h>

+#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; }

+};

+

+ThreadHandle threadId;

+void jackSilentError( const char * ) {};

+

+RtApiJack :: RtApiJack()

+{

+  // Nothing to do here.

+#if !defined(__RTAUDIO_DEBUG__)

+  // Turn off Jack's internal error reporting.

+  jack_set_error_function( &jackSilentError );

+#endif

+}

+

+RtApiJack :: ~RtApiJack()

+{

+  if ( stream_.state != STREAM_CLOSED ) closeStream();

+}

+

+unsigned int RtApiJack :: getDeviceCount( void )

+{

+  // See if we can become a jack client.

+  jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption;

+  jack_status_t *status = NULL;

+  jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );

+  if ( client == 0 ) return 0;

+

+  const char **ports;

+  std::string port, previousPort;

+  unsigned int nChannels = 0, nDevices = 0;

+  ports = jack_get_ports( client, NULL, NULL, 0 );

+  if ( ports ) {

+    // Parse the port names up to the first colon (:).

+    size_t iColon = 0;

+    do {

+      port = (char *) ports[ nChannels ];

+      iColon = port.find(":");

+      if ( iColon != std::string::npos ) {

+        port = port.substr( 0, iColon + 1 );

+        if ( port != previousPort ) {

+          nDevices++;

+          previousPort = port;

+        }

+      }

+    } while ( ports[++nChannels] );

+    free( ports );

+  }

+

+  jack_client_close( client );

+  return nDevices;

+}

+

+RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )

+{

+  RtAudio::DeviceInfo info;

+  info.probed = false;

+

+  jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption

+  jack_status_t *status = NULL;

+  jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );

+  if ( client == 0 ) {

+    errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";

+    error( RtError::WARNING );

+    return info;

+  }

+

+  const char **ports;

+  std::string port, previousPort;

+  unsigned int nPorts = 0, nDevices = 0;

+  ports = jack_get_ports( client, NULL, NULL, 0 );

+  if ( ports ) {

+    // Parse the port names up to the first colon (:).

+    size_t iColon = 0;

+    do {

+      port = (char *) ports[ nPorts ];

+      iColon = port.find(":");

+      if ( iColon != std::string::npos ) {

+        port = port.substr( 0, iColon );

+        if ( port != previousPort ) {

+          if ( nDevices == device ) info.name = port;

+          nDevices++;

+          previousPort = port;

+        }

+      }

+    } while ( ports[++nPorts] );

+    free( ports );

+  }

+

+  if ( device >= nDevices ) {

+    errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";

+    error( RtError::INVALID_USE );

+  }

+

+  // Get the current jack server sample rate.

+  info.sampleRates.clear();

+  info.sampleRates.push_back( jack_get_sample_rate( client ) );

+

+  // Count the available ports containing the client name as device

+  // channels.  Jack "input ports" equal RtAudio output channels.

+  unsigned int nChannels = 0;

+  ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );

+  if ( ports ) {

+    while ( ports[ nChannels ] ) nChannels++;

+    free( ports );

+    info.outputChannels = nChannels;

+  }

+

+  // Jack "output ports" equal RtAudio input channels.

+  nChannels = 0;

+  ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );

+  if ( ports ) {

+    while ( ports[ nChannels ] ) nChannels++;

+    free( ports );

+    info.inputChannels = nChannels;

+  }

+

+  if ( info.outputChannels == 0 && info.inputChannels == 0 ) {

+    jack_client_close(client);

+    errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // If device opens for both playback and capture, we determine the channels.

+  if ( info.outputChannels > 0 && info.inputChannels > 0 )

+    info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;

+

+  // Jack always uses 32-bit floats.

+  info.nativeFormats = RTAUDIO_FLOAT32;

+

+  // Jack doesn't provide default devices so we'll use the first available one.

+  if ( device == 0 && info.outputChannels > 0 )

+    info.isDefaultOutput = true;

+  if ( device == 0 && info.inputChannels > 0 )

+    info.isDefaultInput = true;

+

+  jack_client_close(client);

+  info.probed = true;

+  return info;

+}

+

+int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )

+{

+  CallbackInfo *info = (CallbackInfo *) infoPointer;

+

+  RtApiJack *object = (RtApiJack *) info->object;

+  if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;

+

+  return 0;

+}

+

+// 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.

+extern "C" void *jackCloseStream( void *ptr )

+{

+  CallbackInfo *info = (CallbackInfo *) ptr;

+  RtApiJack *object = (RtApiJack *) info->object;

+

+  object->closeStream();

+

+  pthread_exit( NULL );

+}

+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;

+

+  pthread_create( &threadId, NULL, jackCloseStream, info );

+  std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;

+}

+

+int jackXrun( void *infoPointer )

+{

+  JackHandle *handle = (JackHandle *) infoPointer;

+

+  if ( handle->ports[0] ) handle->xrun[0] = true;

+  if ( handle->ports[1] ) handle->xrun[1] = true;

+

+  return 0;

+}

+

+bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,

+                                   unsigned int firstChannel, unsigned int sampleRate,

+                                   RtAudioFormat format, unsigned int *bufferSize,

+                                   RtAudio::StreamOptions *options )

+{

+  JackHandle *handle = (JackHandle *) stream_.apiHandle;

+

+  // Look for jack server and try to become a client (only do once per stream).

+  jack_client_t *client = 0;

+  if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {

+    jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;

+    jack_status_t *status = NULL;

+    if ( options && !options->streamName.empty() )

+      client = jack_client_open( options->streamName.c_str(), jackoptions, status );

+    else

+      client = jack_client_open( "RtApiJack", jackoptions, status );

+    if ( client == 0 ) {

+      errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";

+      error( RtError::WARNING );

+      return FAILURE;

+    }

+  }

+  else {

+    // The handle must have been created on an earlier pass.

+    client = handle->client;

+  }

+

+  const char **ports;

+  std::string port, previousPort, deviceName;

+  unsigned int nPorts = 0, nDevices = 0;

+  ports = jack_get_ports( client, NULL, NULL, 0 );

+  if ( ports ) {

+    // Parse the port names up to the first colon (:).

+    size_t iColon = 0;

+    do {

+      port = (char *) ports[ nPorts ];

+      iColon = port.find(":");

+      if ( iColon != std::string::npos ) {

+        port = port.substr( 0, iColon );

+        if ( port != previousPort ) {

+          if ( nDevices == device ) deviceName = port;

+          nDevices++;

+          previousPort = port;

+        }

+      }

+    } while ( ports[++nPorts] );

+    free( ports );

+  }

+

+  if ( device >= nDevices ) {

+    errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";

+    return FAILURE;

+  }

+

+  // Count the available ports containing the client name as device

+  // channels.  Jack "input ports" equal RtAudio output channels.

+  unsigned int nChannels = 0;

+  unsigned long flag = JackPortIsInput;

+  if ( mode == INPUT ) flag = JackPortIsOutput;

+  ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );

+  if ( ports ) {

+    while ( ports[ nChannels ] ) nChannels++;

+    free( ports );

+  }

+

+  // Compare the jack ports for specified client to the requested number of channels.

+  if ( nChannels < (channels + firstChannel) ) {

+    errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Check the jack server sample rate.

+  unsigned int jackRate = jack_get_sample_rate( client );

+  if ( sampleRate != jackRate ) {

+    jack_client_close( client );

+    errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+  stream_.sampleRate = jackRate;

+

+  // Get the latency of the JACK port.

+  ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );

+  if ( ports[ firstChannel ] )

+    stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );

+  free( ports );

+

+  // The jack server always uses 32-bit floating-point data.

+  stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;

+  stream_.userFormat = format;

+

+  if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;

+  else stream_.userInterleaved = true;

+

+  // Jack always uses non-interleaved buffers.

+  stream_.deviceInterleaved[mode] = false;

+

+  // Jack always provides host byte-ordered data.

+  stream_.doByteSwap[mode] = false;

+

+  // Get the buffer size.  The buffer size and number of buffers

+  // (periods) is set when the jack server is started.

+  stream_.bufferSize = (int) jack_get_buffer_size( client );

+  *bufferSize = stream_.bufferSize;

+

+  stream_.nDeviceChannels[mode] = channels;

+  stream_.nUserChannels[mode] = channels;

+

+  // Set flags for buffer conversion.

+  stream_.doConvertBuffer[mode] = false;

+  if ( stream_.userFormat != stream_.deviceFormat[mode] )

+    stream_.doConvertBuffer[mode] = true;

+  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&

+       stream_.nUserChannels[mode] > 1 )

+    stream_.doConvertBuffer[mode] = true;

+

+  // Allocate our JackHandle structure for the stream.

+  if ( handle == 0 ) {

+    try {

+      handle = new JackHandle;

+    }

+    catch ( std::bad_alloc& ) {

+      errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";

+      goto error;

+    }

+

+    if ( pthread_cond_init(&handle->condition, NULL) ) {

+      errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";

+      goto error;

+    }

+    stream_.apiHandle = (void *) handle;

+    handle->client = client;

+  }

+  handle->deviceName[mode] = deviceName;

+

+  // Allocate necessary internal buffers.

+  unsigned long bufferBytes;

+  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );

+  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );

+  if ( stream_.userBuffer[mode] == NULL ) {

+    errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";

+    goto error;

+  }

+

+  if ( stream_.doConvertBuffer[mode] ) {

+

+    bool makeBuffer = true;

+    if ( mode == OUTPUT )

+      bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );

+    else { // mode == INPUT

+      bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );

+      if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {

+        unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);

+        if ( bufferBytes < bytesOut ) makeBuffer = false;

+      }

+    }

+

+    if ( makeBuffer ) {

+      bufferBytes *= *bufferSize;

+      if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );

+      stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );

+      if ( stream_.deviceBuffer == NULL ) {

+        errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";

+        goto error;

+      }

+    }

+  }

+

+  // Allocate memory for the Jack ports (channels) identifiers.

+  handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );

+  if ( handle->ports[mode] == NULL )  {

+    errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";

+    goto error;

+  }

+

+  stream_.device[mode] = device;

+  stream_.channelOffset[mode] = firstChannel;

+  stream_.state = STREAM_STOPPED;

+  stream_.callbackInfo.object = (void *) this;

+

+  if ( stream_.mode == OUTPUT && mode == INPUT )

+    // We had already set up the stream for output.

+    stream_.mode = DUPLEX;

+  else {

+    stream_.mode = mode;

+    jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );

+    jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );

+    jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );

+  }

+

+  // Register our ports.

+  char label[64];

+  if ( mode == OUTPUT ) {

+    for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {

+      snprintf( label, 64, "outport %d", i );

+      handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,

+                                                JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );

+    }

+  }

+  else {

+    for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {

+      snprintf( label, 64, "inport %d", i );

+      handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,

+                                                JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );

+    }

+  }

+

+  // Setup the buffer conversion information structure.  We don't use

+  // buffers to do channel offsets, so we override that parameter

+  // here.

+  if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );

+

+  return SUCCESS;

+

+ error:

+  if ( handle ) {

+    pthread_cond_destroy( &handle->condition );

+    jack_client_close( handle->client );

+

+    if ( handle->ports[0] ) free( handle->ports[0] );

+    if ( handle->ports[1] ) free( handle->ports[1] );

+

+    delete handle;

+    stream_.apiHandle = 0;

+  }

+

+  for ( int i=0; i<2; i++ ) {

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  return FAILURE;

+}

+

+void RtApiJack :: closeStream( void )

+{

+  if ( stream_.state == STREAM_CLOSED ) {

+    errorText_ = "RtApiJack::closeStream(): no open stream to close!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  JackHandle *handle = (JackHandle *) stream_.apiHandle;

+  if ( handle ) {

+

+    if ( stream_.state == STREAM_RUNNING )

+      jack_deactivate( handle->client );

+

+    jack_client_close( handle->client );

+  }

+

+  if ( handle ) {

+    if ( handle->ports[0] ) free( handle->ports[0] );

+    if ( handle->ports[1] ) free( handle->ports[1] );

+    pthread_cond_destroy( &handle->condition );

+    delete handle;

+    stream_.apiHandle = 0;

+  }

+

+  for ( int i=0; i<2; i++ ) {

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  stream_.mode = UNINITIALIZED;

+  stream_.state = STREAM_CLOSED;

+}

+

+void RtApiJack :: startStream( void )

+{

+  verifyStream();

+  if ( stream_.state == STREAM_RUNNING ) {

+    errorText_ = "RtApiJack::startStream(): the stream is already running!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  MUTEX_LOCK(&stream_.mutex);

+

+  JackHandle *handle = (JackHandle *) stream_.apiHandle;

+  int result = jack_activate( handle->client );

+  if ( result ) {

+    errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";

+    goto unlock;

+  }

+

+  const char **ports;

+

+  // Get the list of available ports.

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+    result = 1;

+    ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);

+    if ( ports == NULL) {

+      errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";

+      goto unlock;

+    }

+

+    // Now make the port connections.  Since RtAudio wasn't designed to

+    // allow the user to select particular channels of a device, we'll

+    // just open the first "nChannels" ports with offset.

+    for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {

+      result = 1;

+      if ( ports[ stream_.channelOffset[0] + i ] )

+        result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );

+      if ( result ) {

+        free( ports );

+        errorText_ = "RtApiJack::startStream(): error connecting output ports!";

+        goto unlock;

+      }

+    }

+    free(ports);

+  }

+

+  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {

+    result = 1;

+    ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );

+    if ( ports == NULL) {

+      errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";

+      goto unlock;

+    }

+

+    // Now make the port connections.  See note above.

+    for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {

+      result = 1;

+      if ( ports[ stream_.channelOffset[1] + i ] )

+        result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );

+      if ( result ) {

+        free( ports );

+        errorText_ = "RtApiJack::startStream(): error connecting input ports!";

+        goto unlock;

+      }

+    }

+    free(ports);

+  }

+

+  handle->drainCounter = 0;

+  handle->internalDrain = false;

+  stream_.state = STREAM_RUNNING;

+

+ unlock:

+  MUTEX_UNLOCK(&stream_.mutex);

+

+  if ( result == 0 ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiJack :: stopStream( void )

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

+    errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_UNLOCK( &stream_.mutex );

+    return;

+  }

+

+  JackHandle *handle = (JackHandle *) stream_.apiHandle;

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+

+    if ( handle->drainCounter == 0 ) {

+      handle->drainCounter = 2;

+      pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled

+    }

+  }

+

+  jack_deactivate( handle->client );

+  stream_.state = STREAM_STOPPED;

+

+  MUTEX_UNLOCK( &stream_.mutex );

+}

+

+void RtApiJack :: abortStream( void )

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

+    errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  JackHandle *handle = (JackHandle *) stream_.apiHandle;

+  handle->drainCounter = 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.

+extern "C" void *jackStopStream( void *ptr )

+{

+  CallbackInfo *info = (CallbackInfo *) ptr;

+  RtApiJack *object = (RtApiJack *) info->object;

+

+  object->stopStream();

+

+  pthread_exit( NULL );

+}

+

+bool RtApiJack :: callbackEvent( unsigned long nframes )

+{

+  if ( stream_.state == STREAM_STOPPED ) return SUCCESS;

+  if ( stream_.state == STREAM_CLOSED ) {

+    errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";

+    error( RtError::WARNING );

+    return FAILURE;

+  }

+  if ( stream_.bufferSize != nframes ) {

+    errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";

+    error( RtError::WARNING );

+    return FAILURE;

+  }

+

+  CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;

+  JackHandle *handle = (JackHandle *) stream_.apiHandle;

+

+  // Check if we were draining the stream and signal is finished.

+  if ( handle->drainCounter > 3 ) {

+    if ( handle->internalDrain == true )

+      pthread_create( &threadId, NULL, jackStopStream, info );

+    else

+      pthread_cond_signal( &handle->condition );

+    return SUCCESS;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  // The state might change while waiting on a mutex.

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_UNLOCK( &stream_.mutex );

+    return SUCCESS;

+  }

+

+  // Invoke user callback first, to get fresh output data.

+  if ( handle->drainCounter == 0 ) {

+    RtAudioCallback callback = (RtAudioCallback) info->callback;

+    double streamTime = getStreamTime();

+    RtAudioStreamStatus status = 0;

+    if ( stream_.mode != INPUT && handle->xrun[0] == true ) {

+      status |= RTAUDIO_OUTPUT_UNDERFLOW;

+      handle->xrun[0] = false;

+    }

+    if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {

+      status |= RTAUDIO_INPUT_OVERFLOW;

+      handle->xrun[1] = false;

+    }

+    handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],

+                                     stream_.bufferSize, streamTime, status, info->userData );

+    if ( handle->drainCounter == 2 ) {

+      MUTEX_UNLOCK( &stream_.mutex );

+      ThreadHandle id;

+      pthread_create( &id, NULL, jackStopStream, info );

+      return SUCCESS;

+    }

+    else if ( handle->drainCounter == 1 )

+      handle->internalDrain = true;

+  }

+

+  jack_default_audio_sample_t *jackbuffer;

+  unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+

+    if ( handle->drainCounter > 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 );

+      }

+    }

+

+    if ( handle->drainCounter ) {

+      handle->drainCounter++;

+      goto unlock;

+    }

+  }

+

+  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {

+

+    if ( stream_.doConvertBuffer[1] ) {

+      for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {

+        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );

+        memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );

+      }

+      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );

+    }

+    else { // no buffer conversion

+      for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {

+        jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );

+        memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );

+      }

+    }

+  }

+

+ unlock:

+  MUTEX_UNLOCK(&stream_.mutex);

+

+  RtApi::tickStreamTime();

+  return SUCCESS;

+}

+  //******************** End of __UNIX_JACK__ *********************//

+#endif

+

+#if defined(__WINDOWS_ASIO__) // ASIO API on Windows

+

+// The ASIO API is designed around a callback scheme, so this

+// implementation is similar to that used for OS-X CoreAudio and Linux

+// Jack.  The primary constraint with ASIO is that it only allows

+// access to a single driver at a time.  Thus, it is not possible to

+// have more than one simultaneous RtAudio stream.

+//

+// This implementation also requires a number of external ASIO files

+// and a few global variables.  The ASIO callback scheme does not

+// allow for the passing of user data, so we must create a global

+// pointer to our callbackInfo structure.

+//

+// On unix systems, we make use of a pthread condition variable.

+// Since there is no equivalent in Windows, I hacked something based

+// on information found in

+// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.

+

+#include "asiosys.h"

+#include "asio.h"

+#include "iasiothiscallresolver.h"

+#include "asiodrivers.h"

+#include <cmath>

+

+AsioDrivers drivers;

+ASIOCallbacks asioCallbacks;

+ASIODriverInfo driverInfo;

+CallbackInfo *asioCallbackInfo;

+bool asioXRun;

+

+struct AsioHandle {

+  int drainCounter;       // Tracks callback counts when draining

+  bool internalDrain;     // Indicates if stop is initiated from callback or not.

+  ASIOBufferInfo *bufferInfos;

+  HANDLE condition;

+

+  AsioHandle()

+    :drainCounter(0), internalDrain(false), bufferInfos(0) {}

+};

+

+// Function declarations (definitions at end of section)

+static const char* getAsioErrorString( ASIOError result );

+void sampleRateChanged( ASIOSampleRate sRate );

+long asioMessages( long selector, long value, void* message, double* opt );

+

+RtApiAsio :: RtApiAsio()

+{

+  // ASIO cannot run on a multi-threaded appartment. You can call

+  // CoInitialize beforehand, but it must be for appartment threading

+  // (in which case, CoInitilialize will return S_FALSE here).

+  coInitialized_ = false;

+  HRESULT hr = CoInitialize( NULL ); 

+  if ( FAILED(hr) ) {

+    errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";

+    error( RtError::WARNING );

+  }

+  coInitialized_ = true;

+

+  drivers.removeCurrentDriver();

+  driverInfo.asioVersion = 2;

+

+  // See note in DirectSound implementation about GetDesktopWindow().

+  driverInfo.sysRef = GetForegroundWindow();

+}

+

+RtApiAsio :: ~RtApiAsio()

+{

+  if ( stream_.state != STREAM_CLOSED ) closeStream();

+  if ( coInitialized_ ) CoUninitialize();

+}

+

+unsigned int RtApiAsio :: getDeviceCount( void )

+{

+  return (unsigned int) drivers.asioGetNumDev();

+}

+

+RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )

+{

+  RtAudio::DeviceInfo info;

+  info.probed = false;

+

+  // Get device ID

+  unsigned int nDevices = getDeviceCount();

+  if ( nDevices == 0 ) {

+    errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";

+    error( RtError::INVALID_USE );

+  }

+

+  if ( device >= nDevices ) {

+    errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";

+    error( RtError::INVALID_USE );

+  }

+

+  // If a stream is already open, we cannot probe other devices.  Thus, use the saved results.

+  if ( stream_.state != STREAM_CLOSED ) {

+    if ( device >= devices_.size() ) {

+      errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";

+      error( RtError::WARNING );

+      return info;

+    }

+    return devices_[ device ];

+  }

+

+  char driverName[32];

+  ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );

+  if ( result != ASE_OK ) {

+    errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  info.name = driverName;

+

+  if ( !drivers.loadDriver( driverName ) ) {

+    errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  result = ASIOInit( &driverInfo );

+  if ( result != ASE_OK ) {

+    errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // Determine the device channel information.

+  long inputChannels, outputChannels;

+  result = ASIOGetChannels( &inputChannels, &outputChannels );

+  if ( result != ASE_OK ) {

+    drivers.removeCurrentDriver();

+    errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  info.outputChannels = outputChannels;

+  info.inputChannels = inputChannels;

+  if ( info.outputChannels > 0 && info.inputChannels > 0 )

+    info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;

+

+  // Determine the supported sample rates.

+  info.sampleRates.clear();

+  for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {

+    result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );

+    if ( result == ASE_OK )

+      info.sampleRates.push_back( SAMPLE_RATES[i] );

+  }

+

+  // Determine supported data types ... just check first channel and assume rest are the same.

+  ASIOChannelInfo channelInfo;

+  channelInfo.channel = 0;

+  channelInfo.isInput = true;

+  if ( info.inputChannels <= 0 ) channelInfo.isInput = false;

+  result = ASIOGetChannelInfo( &channelInfo );

+  if ( result != ASE_OK ) {

+    drivers.removeCurrentDriver();

+    errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  info.nativeFormats = 0;

+  if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )

+    info.nativeFormats |= RTAUDIO_SINT16;

+  else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )

+    info.nativeFormats |= RTAUDIO_SINT32;

+  else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )

+    info.nativeFormats |= RTAUDIO_FLOAT32;

+  else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )

+    info.nativeFormats |= RTAUDIO_FLOAT64;

+

+  if ( 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;

+}

+

+void bufferSwitch( long index, ASIOBool processNow )

+{

+  RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;

+  object->callbackEvent( index );

+}

+

+void RtApiAsio :: saveDeviceInfo( void )

+{

+  devices_.clear();

+

+  unsigned int nDevices = getDeviceCount();

+  devices_.resize( nDevices );

+  for ( unsigned int i=0; i<nDevices; i++ )

+    devices_[i] = getDeviceInfo( i );

+}

+

+bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,

+                                   unsigned int firstChannel, unsigned int sampleRate,

+                                   RtAudioFormat format, unsigned int *bufferSize,

+                                   RtAudio::StreamOptions *options )

+{

+  // For ASIO, a duplex stream MUST use the same driver.

+  if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) {

+    errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";

+    return FAILURE;

+  }

+

+  char driverName[32];

+  ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );

+  if ( result != ASE_OK ) {

+    errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // The getDeviceInfo() function will not work when a stream is open

+  // because ASIO does not allow multiple devices to run at the same

+  // time.  Thus, we'll probe the system before opening a stream and

+  // save the results for use by getDeviceInfo().

+  this->saveDeviceInfo();

+

+  // Only load the driver once for duplex stream.

+  if ( mode != INPUT || stream_.mode != OUTPUT ) {

+    if ( !drivers.loadDriver( driverName ) ) {

+      errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+

+    result = ASIOInit( &driverInfo );

+    if ( result != ASE_OK ) {

+      errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+  }

+

+  // Check the device channel count.

+  long inputChannels, outputChannels;

+  result = ASIOGetChannels( &inputChannels, &outputChannels );

+  if ( result != ASE_OK ) {

+    drivers.removeCurrentDriver();

+    errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||

+       ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {

+    drivers.removeCurrentDriver();

+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+  stream_.nDeviceChannels[mode] = channels;

+  stream_.nUserChannels[mode] = channels;

+  stream_.channelOffset[mode] = firstChannel;

+

+  // Verify the sample rate is supported.

+  result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );

+  if ( result != ASE_OK ) {

+    drivers.removeCurrentDriver();

+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Get the current sample rate

+  ASIOSampleRate currentRate;

+  result = ASIOGetSampleRate( &currentRate );

+  if ( result != ASE_OK ) {

+    drivers.removeCurrentDriver();

+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Set the sample rate only if necessary

+  if ( currentRate != sampleRate ) {

+    result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );

+    if ( result != ASE_OK ) {

+      drivers.removeCurrentDriver();

+      errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+  }

+

+  // Determine the driver data type.

+  ASIOChannelInfo channelInfo;

+  channelInfo.channel = 0;

+  if ( mode == OUTPUT ) channelInfo.isInput = false;

+  else channelInfo.isInput = true;

+  result = ASIOGetChannelInfo( &channelInfo );

+  if ( result != ASE_OK ) {

+    drivers.removeCurrentDriver();

+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Assuming WINDOWS host is always little-endian.

+  stream_.doByteSwap[mode] = false;

+  stream_.userFormat = format;

+  stream_.deviceFormat[mode] = 0;

+  if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {

+    stream_.deviceFormat[mode] = RTAUDIO_SINT16;

+    if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;

+  }

+  else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {

+    stream_.deviceFormat[mode] = RTAUDIO_SINT32;

+    if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;

+  }

+  else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {

+    stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;

+    if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;

+  }

+  else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {

+    stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;

+    if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;

+  }

+

+  if ( stream_.deviceFormat[mode] == 0 ) {

+    drivers.removeCurrentDriver();

+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Set the buffer size.  For a duplex stream, this will end up

+  // setting the buffer size based on the input constraints, which

+  // should be ok.

+  long minSize, maxSize, preferSize, granularity;

+  result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );

+  if ( result != ASE_OK ) {

+    drivers.removeCurrentDriver();

+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;

+  else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;

+  else if ( granularity == -1 ) {

+    // Make sure bufferSize is a power of two.

+    int log2_of_min_size = 0;

+    int log2_of_max_size = 0;

+

+    for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {

+      if ( minSize & ((long)1 << i) ) log2_of_min_size = i;

+      if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;

+    }

+

+    long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );

+    int min_delta_num = log2_of_min_size;

+

+    for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {

+      long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );

+      if (current_delta < min_delta) {

+        min_delta = current_delta;

+        min_delta_num = i;

+      }

+    }

+

+    *bufferSize = ( (unsigned int)1 << min_delta_num );

+    if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;

+    else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;

+  }

+  else if ( granularity != 0 ) {

+    // Set to an even multiple of granularity, rounding up.

+    *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;

+  }

+

+  if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) {

+    drivers.removeCurrentDriver();

+    errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";

+    return FAILURE;

+  }

+

+  stream_.bufferSize = *bufferSize;

+  stream_.nBuffers = 2;

+

+  if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;

+  else stream_.userInterleaved = true;

+

+  // ASIO always uses non-interleaved buffers.

+  stream_.deviceInterleaved[mode] = false;

+

+  // Allocate, if necessary, our AsioHandle structure for the stream.

+  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;

+  if ( handle == 0 ) {

+    try {

+      handle = new AsioHandle;

+    }

+    catch ( std::bad_alloc& ) {

+      //if ( handle == NULL ) {    

+      drivers.removeCurrentDriver();

+      errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";

+      return FAILURE;

+    }

+    handle->bufferInfos = 0;

+

+    // Create a manual-reset event.

+    handle->condition = CreateEvent( NULL,   // no security

+                                     TRUE,   // manual-reset

+                                     FALSE,  // non-signaled initially

+                                     NULL ); // unnamed

+    stream_.apiHandle = (void *) handle;

+  }

+

+  // Create the ASIO internal buffers.  Since RtAudio sets up input

+  // and output separately, we'll have to dispose of previously

+  // created output buffers for a duplex stream.

+  long inputLatency, outputLatency;

+  if ( mode == INPUT && stream_.mode == OUTPUT ) {

+    ASIODisposeBuffers();

+    if ( handle->bufferInfos ) free( handle->bufferInfos );

+  }

+

+  // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.

+  bool buffersAllocated = false;

+  unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];

+  handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );

+  if ( handle->bufferInfos == NULL ) {

+    errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";

+    errorText_ = errorStream_.str();

+    goto error;

+  }

+

+  ASIOBufferInfo *infos;

+  infos = handle->bufferInfos;

+  for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {

+    infos->isInput = ASIOFalse;

+    infos->channelNum = i + stream_.channelOffset[0];

+    infos->buffers[0] = infos->buffers[1] = 0;

+  }

+  for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {

+    infos->isInput = ASIOTrue;

+    infos->channelNum = i + stream_.channelOffset[1];

+    infos->buffers[0] = infos->buffers[1] = 0;

+  }

+

+  // Set up the ASIO callback structure and create the ASIO data buffers.

+  asioCallbacks.bufferSwitch = &bufferSwitch;

+  asioCallbacks.sampleRateDidChange = &sampleRateChanged;

+  asioCallbacks.asioMessage = &asioMessages;

+  asioCallbacks.bufferSwitchTimeInfo = NULL;

+  result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );

+  if ( result != ASE_OK ) {

+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";

+    errorText_ = errorStream_.str();

+    goto error;

+  }

+  buffersAllocated = true;

+

+  // Set flags for buffer conversion.

+  stream_.doConvertBuffer[mode] = false;

+  if ( stream_.userFormat != stream_.deviceFormat[mode] )

+    stream_.doConvertBuffer[mode] = true;

+  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&

+       stream_.nUserChannels[mode] > 1 )

+    stream_.doConvertBuffer[mode] = true;

+

+  // Allocate necessary internal buffers

+  unsigned long bufferBytes;

+  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );

+  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );

+  if ( stream_.userBuffer[mode] == NULL ) {

+    errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";

+    goto error;

+  }

+

+  if ( stream_.doConvertBuffer[mode] ) {

+

+    bool makeBuffer = true;

+    bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );

+    if ( mode == INPUT ) {

+      if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {

+        unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );

+        if ( bufferBytes <= bytesOut ) makeBuffer = false;

+      }

+    }

+

+    if ( makeBuffer ) {

+      bufferBytes *= *bufferSize;

+      if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );

+      stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );

+      if ( stream_.deviceBuffer == NULL ) {

+        errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";

+        goto error;

+      }

+    }

+  }

+

+  stream_.sampleRate = sampleRate;

+  stream_.device[mode] = device;

+  stream_.state = STREAM_STOPPED;

+  asioCallbackInfo = &stream_.callbackInfo;

+  stream_.callbackInfo.object = (void *) this;

+  if ( stream_.mode == OUTPUT && mode == INPUT )

+    // We had already set up an output stream.

+    stream_.mode = DUPLEX;

+  else

+    stream_.mode = mode;

+

+  // Determine device latencies

+  result = ASIOGetLatencies( &inputLatency, &outputLatency );

+  if ( result != ASE_OK ) {

+    errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING); // warn but don't fail

+  }

+  else {

+    stream_.latency[0] = outputLatency;

+    stream_.latency[1] = inputLatency;

+  }

+

+  // Setup the buffer conversion information structure.  We don't use

+  // buffers to do channel offsets, so we override that parameter

+  // here.

+  if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );

+

+  return SUCCESS;

+

+ error:

+  if ( buffersAllocated )

+    ASIODisposeBuffers();

+  drivers.removeCurrentDriver();

+

+  if ( handle ) {

+    CloseHandle( handle->condition );

+    if ( handle->bufferInfos )

+      free( handle->bufferInfos );

+    delete handle;

+    stream_.apiHandle = 0;

+  }

+

+  for ( int i=0; i<2; i++ ) {

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  return FAILURE;

+}

+

+void RtApiAsio :: closeStream()

+{

+  if ( stream_.state == STREAM_CLOSED ) {

+    errorText_ = "RtApiAsio::closeStream(): no open stream to close!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  if ( stream_.state == STREAM_RUNNING ) {

+    stream_.state = STREAM_STOPPED;

+    ASIOStop();

+  }

+  ASIODisposeBuffers();

+  drivers.removeCurrentDriver();

+

+  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;

+  if ( handle ) {

+    CloseHandle( handle->condition );

+    if ( handle->bufferInfos )

+      free( handle->bufferInfos );

+    delete handle;

+    stream_.apiHandle = 0;

+  }

+

+  for ( int i=0; i<2; i++ ) {

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  stream_.mode = UNINITIALIZED;

+  stream_.state = STREAM_CLOSED;

+}

+

+bool stopThreadCalled = false;

+

+void RtApiAsio :: startStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_RUNNING ) {

+    errorText_ = "RtApiAsio::startStream(): the stream is already running!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  //MUTEX_LOCK( &stream_.mutex );

+

+  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;

+  ASIOError result = ASIOStart();

+  if ( result != ASE_OK ) {

+    errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";

+    errorText_ = errorStream_.str();

+    goto unlock;

+  }

+

+  handle->drainCounter = 0;

+  handle->internalDrain = false;

+  ResetEvent( handle->condition );

+  stream_.state = STREAM_RUNNING;

+  asioXRun = false;

+

+ unlock:

+  //MUTEX_UNLOCK( &stream_.mutex );

+

+  stopThreadCalled = false;

+

+  if ( result == ASE_OK ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiAsio :: stopStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

+    errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  /*

+  MUTEX_LOCK( &stream_.mutex );

+

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_UNLOCK( &stream_.mutex );

+    return;

+  }

+  */

+

+  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+    if ( handle->drainCounter == 0 ) {

+      handle->drainCounter = 2;

+      //      MUTEX_UNLOCK( &stream_.mutex );

+      WaitForSingleObject( handle->condition, INFINITE );  // block until signaled

+      //ResetEvent( handle->condition );

+      //      MUTEX_LOCK( &stream_.mutex );

+    }

+  }

+

+  stream_.state = STREAM_STOPPED;

+

+  ASIOError result = ASIOStop();

+  if ( result != ASE_OK ) {

+    errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";

+    errorText_ = errorStream_.str();

+  }

+

+  //  MUTEX_UNLOCK( &stream_.mutex );

+

+  if ( result == ASE_OK ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiAsio :: abortStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

+    errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  // The following lines were commented-out because some behavior was

+  // noted where the device buffers need to be zeroed to avoid

+  // continuing sound, even when the device buffers are completely

+  // disposed.  So now, calling abort is the same as calling stop.

+  // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;

+  // handle->drainCounter = 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.

+extern "C" 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 ) return SUCCESS;

+  if ( stopThreadCalled ) return SUCCESS;

+  if ( stream_.state == STREAM_CLOSED ) {

+    errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";

+    error( RtError::WARNING );

+    return FAILURE;

+  }

+

+  CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;

+  AsioHandle *handle = (AsioHandle *) stream_.apiHandle;

+

+  // Check if we were draining the stream and signal if finished.

+  if ( handle->drainCounter > 3 ) {

+    if ( handle->internalDrain == false )

+      SetEvent( handle->condition );

+    else { // spawn a thread to stop the stream

+      unsigned threadId;

+      stopThreadCalled = true;

+      stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,

+                                                    &stream_.callbackInfo, 0, &threadId );

+    }

+    return SUCCESS;

+  }

+

+  /*MUTEX_LOCK( &stream_.mutex );

+

+  // The state might change while waiting on a mutex.

+  if ( stream_.state == STREAM_STOPPED ) goto unlock; */

+

+  // Invoke user callback to get fresh output data UNLESS we are

+  // draining stream.

+  if ( handle->drainCounter == 0 ) {

+    RtAudioCallback callback = (RtAudioCallback) info->callback;

+    double streamTime = getStreamTime();

+    RtAudioStreamStatus status = 0;

+    if ( stream_.mode != INPUT && asioXRun == true ) {

+      status |= RTAUDIO_OUTPUT_UNDERFLOW;

+      asioXRun = false;

+    }

+    if ( stream_.mode != OUTPUT && asioXRun == true ) {

+      status |= RTAUDIO_INPUT_OVERFLOW;

+      asioXRun = false;

+    }

+    handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],

+                                     stream_.bufferSize, streamTime, status, info->userData );

+    if ( handle->drainCounter == 2 ) {

+      //      MUTEX_UNLOCK( &stream_.mutex );

+      //      abortStream();

+      unsigned threadId;

+      stopThreadCalled = true;

+      stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,

+                                                    &stream_.callbackInfo, 0, &threadId );

+      return SUCCESS;

+    }

+    else if ( handle->drainCounter == 1 )

+      handle->internalDrain = true;

+  }

+

+  unsigned int nChannels, bufferBytes, i, j;

+  nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+

+    bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );

+

+    if ( handle->drainCounter > 1 ) { // write zeros to the output stream

+

+      for ( i=0, j=0; i<nChannels; i++ ) {

+        if ( handle->bufferInfos[i].isInput != ASIOTrue )

+          memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );

+      }

+

+    }

+    else if ( stream_.doConvertBuffer[0] ) {

+

+      convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );

+      if ( stream_.doByteSwap[0] )

+        byteSwapBuffer( stream_.deviceBuffer,

+                        stream_.bufferSize * stream_.nDeviceChannels[0],

+                        stream_.deviceFormat[0] );

+

+      for ( i=0, j=0; i<nChannels; i++ ) {

+        if ( handle->bufferInfos[i].isInput != ASIOTrue )

+          memcpy( handle->bufferInfos[i].buffers[bufferIndex],

+                  &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );

+      }

+

+    }

+    else {

+

+      if ( stream_.doByteSwap[0] )

+        byteSwapBuffer( stream_.userBuffer[0],

+                        stream_.bufferSize * stream_.nUserChannels[0],

+                        stream_.userFormat );

+

+      for ( i=0, j=0; i<nChannels; i++ ) {

+        if ( handle->bufferInfos[i].isInput != ASIOTrue )

+          memcpy( handle->bufferInfos[i].buffers[bufferIndex],

+                  &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );

+      }

+

+    }

+

+    if ( handle->drainCounter ) {

+      handle->drainCounter++;

+      goto unlock;

+    }

+  }

+

+  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {

+

+    bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);

+

+    if (stream_.doConvertBuffer[1]) {

+

+      // Always interleave ASIO input data.

+      for ( i=0, j=0; i<nChannels; i++ ) {

+        if ( handle->bufferInfos[i].isInput == ASIOTrue )

+          memcpy( &stream_.deviceBuffer[j++*bufferBytes],

+                  handle->bufferInfos[i].buffers[bufferIndex],

+                  bufferBytes );

+      }

+

+      if ( stream_.doByteSwap[1] )

+        byteSwapBuffer( stream_.deviceBuffer,

+                        stream_.bufferSize * stream_.nDeviceChannels[1],

+                        stream_.deviceFormat[1] );

+      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );

+

+    }

+    else {

+      for ( i=0, j=0; i<nChannels; i++ ) {

+        if ( handle->bufferInfos[i].isInput == ASIOTrue ) {

+          memcpy( &stream_.userBuffer[1][bufferBytes*j++],

+                  handle->bufferInfos[i].buffers[bufferIndex],

+                  bufferBytes );

+        }

+      }

+

+      if ( stream_.doByteSwap[1] )

+        byteSwapBuffer( stream_.userBuffer[1],

+                        stream_.bufferSize * stream_.nUserChannels[1],

+                        stream_.userFormat );

+    }

+  }

+

+ unlock:

+  // The following call was suggested by Malte Clasen.  While the API

+  // documentation indicates it should not be required, some device

+  // drivers apparently do not function correctly without it.

+  ASIOOutputReady();

+

+  //  MUTEX_UNLOCK( &stream_.mutex );

+

+  RtApi::tickStreamTime();

+  return SUCCESS;

+}

+

+void sampleRateChanged( ASIOSampleRate sRate )

+{

+  // The ASIO documentation says that this usually only happens during

+  // external sync.  Audio processing is not stopped by the driver,

+  // actual sample rate might not have even changed, maybe only the

+  // sample rate status of an AES/EBU or S/PDIF digital input at the

+  // audio device.

+

+  RtApi *object = (RtApi *) asioCallbackInfo->object;

+  try {

+    object->stopStream();

+  }

+  catch ( RtError &exception ) {

+    std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;

+    return;

+  }

+

+  std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;

+}

+

+long asioMessages( long selector, long value, void* message, double* opt )

+{

+  long ret = 0;

+

+  switch( selector ) {

+  case kAsioSelectorSupported:

+    if ( value == kAsioResetRequest

+         || value == kAsioEngineVersion

+         || value == kAsioResyncRequest

+         || value == kAsioLatenciesChanged

+         // The following three were added for ASIO 2.0, you don't

+         // necessarily have to support them.

+         || value == kAsioSupportsTimeInfo

+         || value == kAsioSupportsTimeCode

+         || value == kAsioSupportsInputMonitor)

+      ret = 1L;

+    break;

+  case kAsioResetRequest:

+    // Defer the task and perform the reset of the driver during the

+    // next "safe" situation.  You cannot reset the driver right now,

+    // as this code is called from the driver.  Reset the driver is

+    // done by completely destruct is. I.e. ASIOStop(),

+    // ASIODisposeBuffers(), Destruction Afterwards you initialize the

+    // driver again.

+    std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;

+    ret = 1L;

+    break;

+  case kAsioResyncRequest:

+    // This informs the application that the driver encountered some

+    // non-fatal data loss.  It is used for synchronization purposes

+    // of different media.  Added mainly to work around the Win16Mutex

+    // problems in Windows 95/98 with the Windows Multimedia system,

+    // which could lose data because the Mutex was held too long by

+    // another thread.  However a driver can issue it in other

+    // situations, too.

+    // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;

+    asioXRun = true;

+    ret = 1L;

+    break;

+  case kAsioLatenciesChanged:

+    // This will inform the host application that the drivers were

+    // latencies changed.  Beware, it this does not mean that the

+    // buffer sizes have changed!  You might need to update internal

+    // delay data.

+    std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;

+    ret = 1L;

+    break;

+  case kAsioEngineVersion:

+    // Return the supported ASIO version of the host application.  If

+    // a host application does not implement this selector, ASIO 1.0

+    // is assumed by the driver.

+    ret = 2L;

+    break;

+  case kAsioSupportsTimeInfo:

+    // Informs the driver whether the

+    // asioCallbacks.bufferSwitchTimeInfo() callback is supported.

+    // For compatibility with ASIO 1.0 drivers the host application

+    // should always support the "old" bufferSwitch method, too.

+    ret = 0;

+    break;

+  case kAsioSupportsTimeCode:

+    // Informs the driver whether application is interested in time

+    // code info.  If an application does not need to know about time

+    // code, the driver has less work to do.

+    ret = 0;

+    break;

+  }

+  return ret;

+}

+

+static const char* getAsioErrorString( ASIOError result )

+{

+  struct Messages 

+  {

+    ASIOError value;

+    const char*message;

+  };

+

+  static Messages m[] = 

+    {

+      {   ASE_NotPresent,    "Hardware input or output is not present or available." },

+      {   ASE_HWMalfunction,  "Hardware is malfunctioning." },

+      {   ASE_InvalidParameter, "Invalid input parameter." },

+      {   ASE_InvalidMode,      "Invalid mode." },

+      {   ASE_SPNotAdvancing,     "Sample position not advancing." },

+      {   ASE_NoClock,            "Sample clock or rate cannot be determined or is not present." },

+      {   ASE_NoMemory,           "Not enough memory to complete the request." }

+    };

+

+  for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )

+    if ( m[i].value == result ) return m[i].message;

+

+  return "Unknown error.";

+}

+//******************** End of __WINDOWS_ASIO__ *********************//

+#endif

+

+

+#if defined(__WINDOWS_DS__) // Windows DirectSound API

+

+// Modified by Robin Davies, October 2005

+// - Improvements to DirectX pointer chasing. 

+// - 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 <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 );

+

+extern "C" 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; }

+};

+

+std::vector< 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 ( coInitialized_ ) CoUninitialize(); // balanced call.

+  if ( stream_.state != STREAM_CLOSED ) closeStream();

+}

+

+// 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.

+  bool isInput = false;

+  HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &isInput );

+  if ( FAILED( result ) ) {

+    errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+  }

+

+  // Query DirectSoundCapture devices.

+  isInput = true;

+  result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &isInput );

+  if ( FAILED( result ) ) {

+    errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+  }

+

+  // Clean out any devices that may have disappeared.

+  std::vector< DsDevice > :: iterator it;

+  for ( it=dsDevices.begin(); it < dsDevices.end(); it++ )

+    if ( it->found == false ) dsDevices.erase( it );

+

+  return 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( RtError::INVALID_USE );

+    }

+  }

+

+  if ( device >= dsDevices.size() ) {

+    errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";

+    error( RtError::INVALID_USE );

+  }

+

+  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( RtError::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( RtError::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] );

+  }

+

+  // 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( RtError::WARNING );

+    return info;

+  }

+

+  DSCCAPS inCaps;

+  inCaps.dwSize = sizeof( inCaps );

+  result = input->GetCaps( &inCaps );

+  if ( FAILED( result ) ) {

+    input->Release();

+    errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";

+    errorText_ = errorStream_.str();

+    error( RtError::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] );

+  }

+  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;

+  }

+

+  unsigned int 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;

+  }

+

+  return FAILURE;

+}

+

+void RtApiDs :: closeStream()

+{

+  if ( stream_.state == STREAM_CLOSED ) {

+    errorText_ = "RtApiDs::closeStream(): no open stream to close!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  // Stop the callback thread.

+  stream_.callbackInfo.isRunning = false;

+  WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );

+  CloseHandle( (HANDLE) stream_.callbackInfo.thread );

+

+  DsHandle *handle = (DsHandle *) stream_.apiHandle;

+  if ( handle ) {

+    if ( handle->buffer[0] ) { // the object pointer can be NULL and valid

+      LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];

+      LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];

+      if ( buffer ) {

+        buffer->Stop();

+        buffer->Release();

+      }

+      object->Release();

+    }

+    if ( handle->buffer[1] ) {

+      LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];

+      LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];

+      if ( buffer ) {

+        buffer->Stop();

+        buffer->Release();

+      }

+      object->Release();

+    }

+    CloseHandle( handle->condition );

+    delete handle;

+    stream_.apiHandle = 0;

+  }

+

+  for ( int i=0; i<2; i++ ) {

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  stream_.mode = UNINITIALIZED;

+  stream_.state = STREAM_CLOSED;

+}

+

+void RtApiDs :: startStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_RUNNING ) {

+    errorText_ = "RtApiDs::startStream(): the stream is already running!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  //MUTEX_LOCK( &stream_.mutex );

+  

+  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:

+  //  MUTEX_UNLOCK( &stream_.mutex );

+

+  if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiDs :: stopStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

+    errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  /*

+  MUTEX_LOCK( &stream_.mutex );

+

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_UNLOCK( &stream_.mutex );

+    return;

+  }

+  */

+

+  HRESULT result = 0;

+  LPVOID audioPtr;

+  DWORD dataLen;

+  DsHandle *handle = (DsHandle *) stream_.apiHandle;

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+    if ( handle->drainCounter == 0 ) {

+      handle->drainCounter = 2;

+      //      MUTEX_UNLOCK( &stream_.mutex );

+      WaitForSingleObject( handle->condition, INFINITE );  // block until signaled

+      //ResetEvent( handle->condition );

+      //      MUTEX_LOCK( &stream_.mutex );

+    }

+

+    stream_.state = STREAM_STOPPED;

+

+    // 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;

+

+    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( RtError::SYSTEM_ERROR );

+}

+

+void RtApiDs :: abortStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

+    errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  DsHandle *handle = (DsHandle *) stream_.apiHandle;

+  handle->drainCounter = 2;

+

+  stopStream();

+}

+

+void RtApiDs :: callbackEvent()

+{

+  if ( stream_.state == STREAM_STOPPED ) {

+    Sleep( 50 ); // sleep 50 milliseconds

+    return;

+  }

+

+  if ( stream_.state == STREAM_CLOSED ) {

+    errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;

+  DsHandle *handle = (DsHandle *) stream_.apiHandle;

+

+  // Check if we were draining the stream and signal is finished.

+  if ( handle->drainCounter > stream_.nBuffers + 2 ) {

+    if ( handle->internalDrain == false )

+      SetEvent( handle->condition );

+    else

+      stopStream();

+    return;

+  }

+

+  /*

+  MUTEX_LOCK( &stream_.mutex );

+

+  // The state might change while waiting on a mutex.

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_UNLOCK( &stream_.mutex );

+    return;

+  }

+  */

+

+  // Invoke user callback to get fresh output data UNLESS we are

+  // draining stream.

+  if ( handle->drainCounter == 0 ) {

+    RtAudioCallback callback = (RtAudioCallback) info->callback;

+    double streamTime = getStreamTime();

+    RtAudioStreamStatus status = 0;

+    if ( stream_.mode != INPUT && handle->xrun[0] == true ) {

+      status |= RTAUDIO_OUTPUT_UNDERFLOW;

+      handle->xrun[0] = false;

+    }

+    if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {

+      status |= RTAUDIO_INPUT_OVERFLOW;

+      handle->xrun[1] = false;

+    }

+    handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],

+                                     stream_.bufferSize, streamTime, status, info->userData );

+    if ( handle->drainCounter == 2 ) {

+      //      MUTEX_UNLOCK( &stream_.mutex );

+      abortStream();

+      return;

+    }

+    else if ( handle->drainCounter == 1 )

+      handle->internalDrain = true;

+  }

+

+  HRESULT result;

+  DWORD currentWritePointer, safeWritePointer;

+  DWORD currentReadPointer, safeReadPointer;

+  UINT nextWritePointer;

+

+  LPVOID buffer1 = NULL;

+  LPVOID buffer2 = NULL;

+  DWORD bufferSize1 = 0;

+  DWORD bufferSize2 = 0;

+

+  char *buffer;

+  long bufferBytes;

+

+  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();

+        error( RtError::SYSTEM_ERROR );

+      }

+      result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );

+      if ( FAILED( result ) ) {

+        errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";

+        errorText_ = errorStream_.str();

+        error( RtError::SYSTEM_ERROR );

+      }

+      while ( true ) {

+        result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );

+        if ( FAILED( result ) ) {

+          errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";

+          errorText_ = errorStream_.str();

+          error( RtError::SYSTEM_ERROR );

+        }

+        result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );

+        if ( FAILED( result ) ) {

+          errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";

+          errorText_ = errorStream_.str();

+          error( RtError::SYSTEM_ERROR );

+        }

+        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();

+        error( RtError::SYSTEM_ERROR );

+      }

+      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();

+        error( RtError::SYSTEM_ERROR );

+      }

+

+      // 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();

+      error( RtError::SYSTEM_ERROR );

+    }

+

+    // Copy our buffer into the DS buffer

+    CopyMemory( buffer1, buffer, bufferSize1 );

+    if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );

+

+    // Update our buffer offset and unlock sound buffer

+    dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );

+    if ( FAILED( result ) ) {

+      errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";

+      errorText_ = errorStream_.str();

+      error( RtError::SYSTEM_ERROR );

+    }

+    nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;

+    handle->bufferPointer[0] = nextWritePointer;

+

+    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();

+      error( RtError::SYSTEM_ERROR );

+    }

+

+    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 ) {

+        // 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();

+          error( RtError::SYSTEM_ERROR );

+        }

+      

+        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();

+      error( RtError::SYSTEM_ERROR );

+    }

+

+    if ( duplexPrerollBytes <= 0 ) {

+      // Copy our buffer into the DS buffer

+      CopyMemory( buffer, buffer1, bufferSize1 );

+      if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );

+    }

+    else {

+      memset( buffer, 0, bufferSize1 );

+      if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );

+      duplexPrerollBytes -= bufferSize1 + bufferSize2;

+    }

+

+    // Update our buffer offset and unlock sound buffer

+    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();

+      error( RtError::SYSTEM_ERROR );

+    }

+    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.

+

+extern "C" unsigned __stdcall callbackHandler( void *ptr )

+{

+  CallbackInfo *info = (CallbackInfo *) ptr;

+  RtApiDs *object = (RtApiDs *) info->object;

+  bool* isRunning = &info->isRunning;

+

+  while ( *isRunning == true ) {

+    object->callbackEvent();

+  }

+

+  _endthreadex( 0 );

+  return 0;

+}

+

+#include "tchar.h"

+

+std::string convertTChar( LPCTSTR name )

+{

+  std::string s;

+

+#if defined( UNICODE ) || defined( _UNICODE )

+  // Yes, this conversion doesn't make sense for two-byte characters

+  // but RtAudio is currently written to return an std::string of

+  // one-byte chars for the device name.

+  for ( unsigned int i=0; i<wcslen( name ); i++ )

+    s.push_back( name[i] );

+#else

+  s.append( std::string( name ) );

+#endif

+

+  return s;

+}

+

+static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,

+                                          LPCTSTR description,

+                                          LPCTSTR module,

+                                          LPVOID lpContext )

+{

+  bool *isInput = (bool *) lpContext;

+

+  HRESULT hr;

+  bool validDevice = false;

+  if ( *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 = convertTChar( description );

+  if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )

+    name = "Default Device";

+  if ( validDevice ) {

+    for ( unsigned int i=0; i<dsDevices.size(); i++ ) {

+      if ( dsDevices[i].name == name ) {

+        dsDevices[i].found = true;

+        if ( *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 ( *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;

+

+  AlsaHandle()

+    :synchronized(false) { xrun[0] = false; xrun[1] = false; }

+};

+

+extern "C" void *alsaCallbackHandler( void * ptr );

+

+RtApiAlsa :: RtApiAlsa()

+{

+  // Nothing to do here.

+}

+

+RtApiAlsa :: ~RtApiAlsa()

+{

+  if ( stream_.state != STREAM_CLOSED ) closeStream();

+}

+

+unsigned int RtApiAlsa :: getDeviceCount( void )

+{

+  unsigned nDevices = 0;

+  int result, subdevice, card;

+  char name[64];

+  snd_ctl_t *handle;

+

+  // Count cards and devices

+  card = -1;

+  snd_card_next( &card );

+  while ( card >= 0 ) {

+    sprintf( name, "hw:%d", card );

+    result = snd_ctl_open( &handle, name, 0 );

+    if ( result < 0 ) {

+      errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";

+      errorText_ = errorStream_.str();

+      error( RtError::WARNING );

+      goto nextcard;

+    }

+    subdevice = -1;

+    while( 1 ) {

+      result = snd_ctl_pcm_next_device( handle, &subdevice );

+      if ( result < 0 ) {

+        errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";

+        errorText_ = errorStream_.str();

+        error( RtError::WARNING );

+        break;

+      }

+      if ( subdevice < 0 )

+        break;

+      nDevices++;

+    }

+  nextcard:

+    snd_ctl_close( handle );

+    snd_card_next( &card );

+  }

+

+  return nDevices;

+}

+

+RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )

+{

+  RtAudio::DeviceInfo info;

+  info.probed = false;

+

+  unsigned nDevices = 0;

+  int result, subdevice, card;

+  char name[64];

+  snd_ctl_t *chandle;

+

+  // Count cards and devices

+  card = -1;

+  snd_card_next( &card );

+  while ( card >= 0 ) {

+    sprintf( name, "hw:%d", card );

+    result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );

+    if ( result < 0 ) {

+      errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";

+      errorText_ = errorStream_.str();

+      error( RtError::WARNING );

+      goto nextcard;

+    }

+    subdevice = -1;

+    while( 1 ) {

+      result = snd_ctl_pcm_next_device( chandle, &subdevice );

+      if ( result < 0 ) {

+        errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";

+        errorText_ = errorStream_.str();

+        error( RtError::WARNING );

+        break;

+      }

+      if ( subdevice < 0 ) break;

+      if ( nDevices == device ) {

+        sprintf( name, "hw:%d,%d", card, subdevice );

+        goto foundDevice;

+      }

+      nDevices++;

+    }

+  nextcard:

+    snd_ctl_close( chandle );

+    snd_card_next( &card );

+  }

+

+  if ( nDevices == 0 ) {

+    errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";

+    error( RtError::INVALID_USE );

+  }

+

+  if ( device >= nDevices ) {

+    errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";

+    error( RtError::INVALID_USE );

+  }

+

+ foundDevice:

+

+  // If a stream is already open, we cannot probe the stream devices.

+  // Thus, use the saved results.

+  if ( stream_.state != STREAM_CLOSED &&

+       ( stream_.device[0] == device || stream_.device[1] == device ) ) {

+    if ( device >= devices_.size() ) {

+      errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";

+      error( RtError::WARNING );

+      return info;

+    }

+    return devices_[ device ];

+  }

+

+  int openMode = SND_PCM_ASYNC;

+  snd_pcm_stream_t stream;

+  snd_pcm_info_t *pcminfo;

+  snd_pcm_info_alloca( &pcminfo );

+  snd_pcm_t *phandle;

+  snd_pcm_hw_params_t *params;

+  snd_pcm_hw_params_alloca( &params );

+

+  // First try for playback

+  stream = SND_PCM_STREAM_PLAYBACK;

+  snd_pcm_info_set_device( pcminfo, subdevice );

+  snd_pcm_info_set_subdevice( pcminfo, 0 );

+  snd_pcm_info_set_stream( pcminfo, stream );

+

+  result = snd_ctl_pcm_info( chandle, pcminfo );

+  if ( result < 0 ) {

+    // Device probably doesn't support playback.

+    goto captureProbe;

+  }

+

+  result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );

+  if ( result < 0 ) {

+    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    goto captureProbe;

+  }

+

+  // The device is open ... fill the parameter structure.

+  result = snd_pcm_hw_params_any( phandle, params );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    goto captureProbe;

+  }

+

+  // Get output channel information.

+  unsigned int value;

+  result = snd_pcm_hw_params_get_channels_max( params, &value );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    goto captureProbe;

+  }

+  info.outputChannels = value;

+  snd_pcm_close( phandle );

+

+ captureProbe:

+  // Now try for capture

+  stream = SND_PCM_STREAM_CAPTURE;

+  snd_pcm_info_set_stream( pcminfo, stream );

+

+  result = snd_ctl_pcm_info( chandle, pcminfo );

+  snd_ctl_close( chandle );

+  if ( result < 0 ) {

+    // Device probably doesn't support capture.

+    if ( info.outputChannels == 0 ) return info;

+    goto probeParameters;

+  }

+

+  result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);

+  if ( result < 0 ) {

+    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    if ( info.outputChannels == 0 ) return info;

+    goto probeParameters;

+  }

+

+  // The device is open ... fill the parameter structure.

+  result = snd_pcm_hw_params_any( phandle, params );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    if ( info.outputChannels == 0 ) return info;

+    goto probeParameters;

+  }

+

+  result = snd_pcm_hw_params_get_channels_max( params, &value );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    if ( info.outputChannels == 0 ) return info;

+    goto probeParameters;

+  }

+  info.inputChannels = value;

+  snd_pcm_close( phandle );

+

+  // If device opens for both playback and capture, we determine the channels.

+  if ( info.outputChannels > 0 && info.inputChannels > 0 )

+    info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;

+

+  // ALSA doesn't provide default devices so we'll use the first available one.

+  if ( device == 0 && info.outputChannels > 0 )

+    info.isDefaultOutput = true;

+  if ( device == 0 && info.inputChannels > 0 )

+    info.isDefaultInput = true;

+

+ probeParameters:

+  // At this point, we just need to figure out the supported data

+  // formats and sample rates.  We'll proceed by opening the device in

+  // the direction with the maximum number of channels, or playback if

+  // they are equal.  This might limit our sample rate options, but so

+  // be it.

+

+  if ( info.outputChannels >= info.inputChannels )

+    stream = SND_PCM_STREAM_PLAYBACK;

+  else

+    stream = SND_PCM_STREAM_CAPTURE;

+  snd_pcm_info_set_stream( pcminfo, stream );

+

+  result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);

+  if ( result < 0 ) {

+    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // The device is open ... fill the parameter structure.

+  result = snd_pcm_hw_params_any( phandle, params );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // Test our discrete set of sample rate values.

+  info.sampleRates.clear();

+  for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {

+    if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 )

+      info.sampleRates.push_back( SAMPLE_RATES[i] );

+  }

+  if ( info.sampleRates.size() == 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // Probe the supported data formats ... we don't care about endian-ness just yet

+  snd_pcm_format_t format;

+  info.nativeFormats = 0;

+  format = SND_PCM_FORMAT_S8;

+  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )

+    info.nativeFormats |= RTAUDIO_SINT8;

+  format = SND_PCM_FORMAT_S16;

+  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )

+    info.nativeFormats |= RTAUDIO_SINT16;

+  format = SND_PCM_FORMAT_S24;

+  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )

+    info.nativeFormats |= RTAUDIO_SINT24;

+  format = SND_PCM_FORMAT_S32;

+  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )

+    info.nativeFormats |= RTAUDIO_SINT32;

+  format = SND_PCM_FORMAT_FLOAT;

+  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )

+    info.nativeFormats |= RTAUDIO_FLOAT32;

+  format = SND_PCM_FORMAT_FLOAT64;

+  if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )

+    info.nativeFormats |= RTAUDIO_FLOAT64;

+

+  // Check that we have at least one supported format

+  if ( info.nativeFormats == 0 ) {

+    errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // Get the device name

+  char *cardname;

+  result = snd_card_get_name( card, &cardname );

+  if ( result >= 0 )

+    sprintf( name, "hw:%s,%d", cardname, subdevice );

+  info.name = name;

+

+  // That's all ... close the device and return

+  snd_pcm_close( phandle );

+  info.probed = true;

+  return info;

+}

+

+void RtApiAlsa :: saveDeviceInfo( void )

+{

+  devices_.clear();

+

+  unsigned int nDevices = getDeviceCount();

+  devices_.resize( nDevices );

+  for ( unsigned int i=0; i<nDevices; i++ )

+    devices_[i] = getDeviceInfo( i );

+}

+

+bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,

+                                   unsigned int firstChannel, unsigned int sampleRate,

+                                   RtAudioFormat format, unsigned int *bufferSize,

+                                   RtAudio::StreamOptions *options )

+

+{

+#if defined(__RTAUDIO_DEBUG__)

+  snd_output_t *out;

+  snd_output_stdio_attach(&out, stderr, 0);

+#endif

+

+  // I'm not using the "plug" interface ... too much inconsistent behavior.

+

+  unsigned nDevices = 0;

+  int result, subdevice, card;

+  char name[64];

+  snd_ctl_t *chandle;

+

+  if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )

+    snprintf(name, sizeof(name), "%s", "default");

+  else {

+    // Count cards and devices

+    card = -1;

+    snd_card_next( &card );

+    while ( card >= 0 ) {

+      sprintf( name, "hw:%d", card );

+      result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );

+      if ( result < 0 ) {

+        errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";

+        errorText_ = errorStream_.str();

+        return FAILURE;

+      }

+      subdevice = -1;

+      while( 1 ) {

+        result = snd_ctl_pcm_next_device( chandle, &subdevice );

+        if ( result < 0 ) break;

+        if ( subdevice < 0 ) break;

+        if ( nDevices == device ) {

+          sprintf( name, "hw:%d,%d", card, subdevice );

+          snd_ctl_close( chandle );

+          goto foundDevice;

+        }

+        nDevices++;

+      }

+      snd_ctl_close( chandle );

+      snd_card_next( &card );

+    }

+

+    if ( nDevices == 0 ) {

+      // This should not happen because a check is made before this function is called.

+      errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";

+      return FAILURE;

+    }

+

+    if ( device >= nDevices ) {

+      // This should not happen because a check is made before this function is called.

+      errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";

+      return FAILURE;

+    }

+  }

+

+ foundDevice:

+

+  // The getDeviceInfo() function will not work for a device that is

+  // already open.  Thus, we'll probe the system before opening a

+  // stream and save the results for use by getDeviceInfo().

+  if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once

+    this->saveDeviceInfo();

+

+  snd_pcm_stream_t stream;

+  if ( mode == OUTPUT )

+    stream = SND_PCM_STREAM_PLAYBACK;

+  else

+    stream = SND_PCM_STREAM_CAPTURE;

+

+  snd_pcm_t *phandle;

+  int openMode = SND_PCM_ASYNC;

+  result = snd_pcm_open( &phandle, name, stream, openMode );

+  if ( result < 0 ) {

+    if ( mode == OUTPUT )

+      errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";

+    else

+      errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Fill the parameter structure.

+  snd_pcm_hw_params_t *hw_params;

+  snd_pcm_hw_params_alloca( &hw_params );

+  result = snd_pcm_hw_params_any( phandle, hw_params );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+#if defined(__RTAUDIO_DEBUG__)

+  fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );

+  snd_pcm_hw_params_dump( hw_params, out );

+#endif

+

+  // Set access ... check user preference.

+  if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {

+    stream_.userInterleaved = false;

+    result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );

+    if ( result < 0 ) {

+      result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );

+      stream_.deviceInterleaved[mode] =  true;

+    }

+    else

+      stream_.deviceInterleaved[mode] = false;

+  }

+  else {

+    stream_.userInterleaved = true;

+    result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );

+    if ( result < 0 ) {

+      result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );

+      stream_.deviceInterleaved[mode] =  false;

+    }

+    else

+      stream_.deviceInterleaved[mode] =  true;

+  }

+

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Determine how to set the device format.

+  stream_.userFormat = format;

+  snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;

+

+  if ( format == RTAUDIO_SINT8 )

+    deviceFormat = SND_PCM_FORMAT_S8;

+  else if ( format == RTAUDIO_SINT16 )

+    deviceFormat = SND_PCM_FORMAT_S16;

+  else if ( format == RTAUDIO_SINT24 )

+    deviceFormat = SND_PCM_FORMAT_S24;

+  else if ( format == RTAUDIO_SINT32 )

+    deviceFormat = SND_PCM_FORMAT_S32;

+  else if ( format == RTAUDIO_FLOAT32 )

+    deviceFormat = SND_PCM_FORMAT_FLOAT;

+  else if ( format == RTAUDIO_FLOAT64 )

+    deviceFormat = SND_PCM_FORMAT_FLOAT64;

+

+  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {

+    stream_.deviceFormat[mode] = format;

+    goto setFormat;

+  }

+

+  // The user requested format is not natively supported by the device.

+  deviceFormat = SND_PCM_FORMAT_FLOAT64;

+  if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {

+    stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;

+    goto setFormat;

+  }

+

+  deviceFormat = SND_PCM_FORMAT_FLOAT;

+  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {

+    stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;

+    goto setFormat;

+  }

+

+  deviceFormat = SND_PCM_FORMAT_S32;

+  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {

+    stream_.deviceFormat[mode] = RTAUDIO_SINT32;

+    goto setFormat;

+  }

+

+  deviceFormat = SND_PCM_FORMAT_S24;

+  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {

+    stream_.deviceFormat[mode] = RTAUDIO_SINT24;

+    goto setFormat;

+  }

+

+  deviceFormat = SND_PCM_FORMAT_S16;

+  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {

+    stream_.deviceFormat[mode] = RTAUDIO_SINT16;

+    goto setFormat;

+  }

+

+  deviceFormat = SND_PCM_FORMAT_S8;

+  if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {

+    stream_.deviceFormat[mode] = RTAUDIO_SINT8;

+    goto setFormat;

+  }

+

+  // If we get here, no supported format was found.

+  errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";

+  errorText_ = errorStream_.str();

+  return FAILURE;

+

+ setFormat:

+  result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Determine whether byte-swaping is necessary.

+  stream_.doByteSwap[mode] = false;

+  if ( deviceFormat != SND_PCM_FORMAT_S8 ) {

+    result = snd_pcm_format_cpu_endian( deviceFormat );

+    if ( result == 0 )

+      stream_.doByteSwap[mode] = true;

+    else if (result < 0) {

+      snd_pcm_close( phandle );

+      errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";

+      errorText_ = errorStream_.str();

+      return FAILURE;

+    }

+  }

+

+  // Set the sample rate.

+  result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Determine the number of channels for this device.  We support a possible

+  // minimum device channel number > than the value requested by the user.

+  stream_.nUserChannels[mode] = channels;

+  unsigned int value;

+  result = snd_pcm_hw_params_get_channels_max( hw_params, &value );

+  unsigned int deviceChannels = value;

+  if ( result < 0 || deviceChannels < channels + firstChannel ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  result = snd_pcm_hw_params_get_channels_min( hw_params, &value );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+  deviceChannels = value;

+  if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;

+  stream_.nDeviceChannels[mode] = deviceChannels;

+

+  // Set the device channels.

+  result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Set the buffer (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 ) {

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  stream_.bufferSize = *bufferSize;

+

+  // Install the hardware configuration

+  result = snd_pcm_hw_params( phandle, hw_params );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+#if defined(__RTAUDIO_DEBUG__)

+  fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");

+  snd_pcm_hw_params_dump( hw_params, out );

+#endif

+

+  // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.

+  snd_pcm_sw_params_t *sw_params = NULL;

+  snd_pcm_sw_params_alloca( &sw_params );

+  snd_pcm_sw_params_current( phandle, sw_params );

+  snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );

+  snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );

+  snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );

+

+  // The following two settings were suggested by Theo Veenker

+  //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );

+  //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );

+

+  // here are two options for a fix

+  //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );

+  snd_pcm_uframes_t val;

+  snd_pcm_sw_params_get_boundary( sw_params, &val );

+  snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );

+

+  result = snd_pcm_sw_params( phandle, sw_params );

+  if ( result < 0 ) {

+    snd_pcm_close( phandle );

+    errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+#if defined(__RTAUDIO_DEBUG__)

+  fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");

+  snd_pcm_sw_params_dump( sw_params, out );

+#endif

+

+  // Set flags for buffer conversion

+  stream_.doConvertBuffer[mode] = false;

+  if ( stream_.userFormat != stream_.deviceFormat[mode] )

+    stream_.doConvertBuffer[mode] = true;

+  if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )

+    stream_.doConvertBuffer[mode] = true;

+  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&

+       stream_.nUserChannels[mode] > 1 )

+    stream_.doConvertBuffer[mode] = true;

+

+  // Allocate the ApiHandle if necessary and then save.

+  AlsaHandle *apiInfo = 0;

+  if ( stream_.apiHandle == 0 ) {

+    try {

+      apiInfo = (AlsaHandle *) new AlsaHandle;

+    }

+    catch ( std::bad_alloc& ) {

+      errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";

+      goto error;

+    }

+

+    if ( pthread_cond_init( &apiInfo->runnable, NULL ) ) {

+      errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";

+      goto error;

+    }

+

+    stream_.apiHandle = (void *) apiInfo;

+    apiInfo->handles[0] = 0;

+    apiInfo->handles[1] = 0;

+  }

+  else {

+    apiInfo = (AlsaHandle *) stream_.apiHandle;

+  }

+  apiInfo->handles[mode] = phandle;

+

+  // Allocate necessary internal buffers.

+  unsigned long bufferBytes;

+  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );

+  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );

+  if ( stream_.userBuffer[mode] == NULL ) {

+    errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";

+    goto error;

+  }

+

+  if ( stream_.doConvertBuffer[mode] ) {

+

+    bool makeBuffer = true;

+    bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );

+    if ( mode == INPUT ) {

+      if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {

+        unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );

+        if ( bufferBytes <= bytesOut ) makeBuffer = false;

+      }

+    }

+

+    if ( makeBuffer ) {

+      bufferBytes *= *bufferSize;

+      if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );

+      stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );

+      if ( stream_.deviceBuffer == NULL ) {

+        errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";

+        goto error;

+      }

+    }

+  }

+

+  stream_.sampleRate = sampleRate;

+  stream_.nBuffers = periods;

+  stream_.device[mode] = device;

+  stream_.state = STREAM_STOPPED;

+

+  // Setup the buffer conversion information structure.

+  if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );

+

+  // Setup thread if necessary.

+  if ( stream_.mode == OUTPUT && mode == INPUT ) {

+    // We had already set up an output stream.

+    stream_.mode = DUPLEX;

+    // Link the streams if possible.

+    apiInfo->synchronized = false;

+    if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )

+      apiInfo->synchronized = true;

+    else {

+      errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";

+      error( RtError::WARNING );

+    }

+  }

+  else {

+    stream_.mode = mode;

+

+    // Setup callback thread.

+    stream_.callbackInfo.object = (void *) this;

+

+    // Set the thread attributes for joinable and realtime scheduling

+    // priority (optional).  The higher priority will only take affect

+    // if the program is run as root or suid. Note, under Linux

+    // processes with CAP_SYS_NICE privilege, a user can change

+    // scheduling policy and priority (thus need not be root). See

+    // POSIX "capabilities".

+    pthread_attr_t attr;

+    pthread_attr_init( &attr );

+    pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );

+#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)

+    if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {

+      struct sched_param param;

+      int priority = options->priority;

+      int min = sched_get_priority_min( SCHED_RR );

+      int max = sched_get_priority_max( SCHED_RR );

+      if ( priority < min ) priority = min;

+      else if ( priority > max ) priority = max;

+      param.sched_priority = priority;

+      pthread_attr_setschedparam( &attr, &param );

+      pthread_attr_setschedpolicy( &attr, SCHED_RR );

+    }

+    else

+      pthread_attr_setschedpolicy( &attr, SCHED_OTHER );

+#else

+    pthread_attr_setschedpolicy( &attr, SCHED_OTHER );

+#endif

+

+    stream_.callbackInfo.isRunning = true;

+    result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );

+    pthread_attr_destroy( &attr );

+    if ( result ) {

+      stream_.callbackInfo.isRunning = false;

+      errorText_ = "RtApiAlsa::error creating callback thread!";

+      goto error;

+    }

+  }

+

+  return SUCCESS;

+

+ error:

+  if ( apiInfo ) {

+    pthread_cond_destroy( &apiInfo->runnable );

+    if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );

+    if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );

+    delete apiInfo;

+    stream_.apiHandle = 0;

+  }

+

+  for ( int i=0; i<2; i++ ) {

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  return FAILURE;

+}

+

+void RtApiAlsa :: closeStream()

+{

+  if ( stream_.state == STREAM_CLOSED ) {

+    errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;

+  stream_.callbackInfo.isRunning = false;

+  MUTEX_LOCK( &stream_.mutex );

+  if ( stream_.state == STREAM_STOPPED )

+    pthread_cond_signal( &apiInfo->runnable );

+  MUTEX_UNLOCK( &stream_.mutex );

+  pthread_join( stream_.callbackInfo.thread, NULL );

+

+  if ( stream_.state == STREAM_RUNNING ) {

+    stream_.state = STREAM_STOPPED;

+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )

+      snd_pcm_drop( apiInfo->handles[0] );

+    if ( stream_.mode == INPUT || stream_.mode == DUPLEX )

+      snd_pcm_drop( apiInfo->handles[1] );

+  }

+

+  if ( apiInfo ) {

+    pthread_cond_destroy( &apiInfo->runnable );

+    if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );

+    if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );

+    delete apiInfo;

+    stream_.apiHandle = 0;

+  }

+

+  for ( int i=0; i<2; i++ ) {

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  stream_.mode = UNINITIALIZED;

+  stream_.state = STREAM_CLOSED;

+}

+

+void RtApiAlsa :: startStream()

+{

+  // This method calls snd_pcm_prepare if the device isn't already in that state.

+

+  verifyStream();

+  if ( stream_.state == STREAM_RUNNING ) {

+    errorText_ = "RtApiAlsa::startStream(): the stream is already running!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  int result = 0;

+  snd_pcm_state_t state;

+  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;

+  snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+    state = snd_pcm_state( handle[0] );

+    if ( state != SND_PCM_STATE_PREPARED ) {

+      result = snd_pcm_prepare( handle[0] );

+      if ( result < 0 ) {

+        errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";

+        errorText_ = errorStream_.str();

+        goto unlock;

+      }

+    }

+  }

+

+  if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {

+    state = snd_pcm_state( handle[1] );

+    if ( state != SND_PCM_STATE_PREPARED ) {

+      result = snd_pcm_prepare( handle[1] );

+      if ( result < 0 ) {

+        errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";

+        errorText_ = errorStream_.str();

+        goto unlock;

+      }

+    }

+  }

+

+  stream_.state = STREAM_RUNNING;

+

+ unlock:

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  pthread_cond_signal( &apiInfo->runnable );

+

+  if ( result >= 0 ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiAlsa :: stopStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

+    errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  stream_.state = STREAM_STOPPED;

+  MUTEX_LOCK( &stream_.mutex );

+

+  //if ( stream_.state == STREAM_STOPPED ) {

+  //  MUTEX_UNLOCK( &stream_.mutex );

+  //  return;

+  //}

+

+  int result = 0;

+  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;

+  snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+    if ( apiInfo->synchronized ) 

+      result = snd_pcm_drop( handle[0] );

+    else

+      result = snd_pcm_drain( handle[0] );

+    if ( result < 0 ) {

+      errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+  if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {

+    result = snd_pcm_drop( handle[1] );

+    if ( result < 0 ) {

+      errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+ unlock:

+  stream_.state = STREAM_STOPPED;

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  if ( result >= 0 ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiAlsa :: abortStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

+    errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  stream_.state = STREAM_STOPPED;

+  MUTEX_LOCK( &stream_.mutex );

+

+  //if ( stream_.state == STREAM_STOPPED ) {

+  //  MUTEX_UNLOCK( &stream_.mutex );

+  //  return;

+  //}

+

+  int result = 0;

+  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;

+  snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+    result = snd_pcm_drop( handle[0] );

+    if ( result < 0 ) {

+      errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+  if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {

+    result = snd_pcm_drop( handle[1] );

+    if ( result < 0 ) {

+      errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+ unlock:

+  stream_.state = STREAM_STOPPED;

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  if ( result >= 0 ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiAlsa :: callbackEvent()

+{

+  AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_LOCK( &stream_.mutex );

+    pthread_cond_wait( &apiInfo->runnable, &stream_.mutex );

+    if ( stream_.state != STREAM_RUNNING ) {

+      MUTEX_UNLOCK( &stream_.mutex );

+      return;

+    }

+    MUTEX_UNLOCK( &stream_.mutex );

+  }

+

+  if ( stream_.state == STREAM_CLOSED ) {

+    errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  int doStopStream = 0;

+  RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;

+  double streamTime = getStreamTime();

+  RtAudioStreamStatus status = 0;

+  if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {

+    status |= RTAUDIO_OUTPUT_UNDERFLOW;

+    apiInfo->xrun[0] = false;

+  }

+  if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {

+    status |= RTAUDIO_INPUT_OVERFLOW;

+    apiInfo->xrun[1] = false;

+  }

+  doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],

+                           stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );

+

+  if ( doStopStream == 2 ) {

+    abortStream();

+    return;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  // The state might change while waiting on a mutex.

+  if ( stream_.state == STREAM_STOPPED ) goto unlock;

+

+  int result;

+  char *buffer;

+  int channels;

+  snd_pcm_t **handle;

+  snd_pcm_sframes_t frames;

+  RtAudioFormat format;

+  handle = (snd_pcm_t **) apiInfo->handles;

+

+  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {

+

+    // Setup parameters.

+    if ( stream_.doConvertBuffer[1] ) {

+      buffer = stream_.deviceBuffer;

+      channels = stream_.nDeviceChannels[1];

+      format = stream_.deviceFormat[1];

+    }

+    else {

+      buffer = stream_.userBuffer[1];

+      channels = stream_.nUserChannels[1];

+      format = stream_.userFormat;

+    }

+

+    // Read samples from device in interleaved/non-interleaved format.

+    if ( stream_.deviceInterleaved[1] )

+      result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );

+    else {

+      void *bufs[channels];

+      size_t offset = stream_.bufferSize * formatBytes( format );

+      for ( int i=0; i<channels; i++ )

+        bufs[i] = (void *) (buffer + (i * offset));

+      result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );

+    }

+

+    if ( result < (int) stream_.bufferSize ) {

+      // Either an error or overrun occured.

+      if ( result == -EPIPE ) {

+        snd_pcm_state_t state = snd_pcm_state( handle[1] );

+        if ( state == SND_PCM_STATE_XRUN ) {

+          apiInfo->xrun[1] = true;

+          result = snd_pcm_prepare( handle[1] );

+          if ( result < 0 ) {

+            errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";

+            errorText_ = errorStream_.str();

+          }

+        }

+        else {

+          errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";

+          errorText_ = errorStream_.str();

+        }

+      }

+      else {

+        errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";

+        errorText_ = errorStream_.str();

+      }

+      error( RtError::WARNING );

+      goto tryOutput;

+    }

+

+    // Do byte swapping if necessary.

+    if ( stream_.doByteSwap[1] )

+      byteSwapBuffer( buffer, stream_.bufferSize * channels, format );

+

+    // Do buffer conversion if necessary.

+    if ( stream_.doConvertBuffer[1] )

+      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );

+

+    // Check stream latency

+    result = snd_pcm_delay( handle[1], &frames );

+    if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;

+  }

+

+ tryOutput:

+

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+

+    // Setup parameters and do buffer conversion if necessary.

+    if ( stream_.doConvertBuffer[0] ) {

+      buffer = stream_.deviceBuffer;

+      convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );

+      channels = stream_.nDeviceChannels[0];

+      format = stream_.deviceFormat[0];

+    }

+    else {

+      buffer = stream_.userBuffer[0];

+      channels = stream_.nUserChannels[0];

+      format = stream_.userFormat;

+    }

+

+    // Do byte swapping if necessary.

+    if ( stream_.doByteSwap[0] )

+      byteSwapBuffer(buffer, stream_.bufferSize * channels, format);

+

+    // Write samples to device in interleaved/non-interleaved format.

+    if ( stream_.deviceInterleaved[0] )

+      result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );

+    else {

+      void *bufs[channels];

+      size_t offset = stream_.bufferSize * formatBytes( format );

+      for ( int i=0; i<channels; i++ )

+        bufs[i] = (void *) (buffer + (i * offset));

+      result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );

+    }

+

+    if ( result < (int) stream_.bufferSize ) {

+      // Either an error or underrun occured.

+      if ( result == -EPIPE ) {

+        snd_pcm_state_t state = snd_pcm_state( handle[0] );

+        if ( state == SND_PCM_STATE_XRUN ) {

+          apiInfo->xrun[0] = true;

+          result = snd_pcm_prepare( handle[0] );

+          if ( result < 0 ) {

+            errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";

+            errorText_ = errorStream_.str();

+          }

+        }

+        else {

+          errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";

+          errorText_ = errorStream_.str();

+        }

+      }

+      else {

+        errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";

+        errorText_ = errorStream_.str();

+      }

+      error( RtError::WARNING );

+      goto unlock;

+    }

+

+    // Check stream latency

+    result = snd_pcm_delay( handle[0], &frames );

+    if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;

+  }

+

+ unlock:

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  RtApi::tickStreamTime();

+  if ( doStopStream == 1 ) this->stopStream();

+}

+

+extern "C" void *alsaCallbackHandler( void *ptr )

+{

+  CallbackInfo *info = (CallbackInfo *) ptr;

+  RtApiAlsa *object = (RtApiAlsa *) info->object;

+  bool *isRunning = &info->isRunning;

+

+  while ( *isRunning == true ) {

+    pthread_testcancel();

+    object->callbackEvent();

+  }

+

+  pthread_exit( NULL );

+}

+

+//******************** End of __LINUX_ALSA__ *********************//

+#endif

+

+

+#if defined(__LINUX_OSS__)

+

+#include <unistd.h>

+#include <sys/ioctl.h>

+#include <unistd.h>

+#include <fcntl.h>

+#include "soundcard.h"

+#include <errno.h>

+#include <math.h>

+

+extern "C" void *ossCallbackHandler(void * ptr);

+

+// A structure to hold various information related to the OSS API

+// implementation.

+struct OssHandle {

+  int id[2];    // device ids

+  bool xrun[2];

+  bool triggered;

+  pthread_cond_t runnable;

+

+  OssHandle()

+    :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }

+};

+

+RtApiOss :: RtApiOss()

+{

+  // Nothing to do here.

+}

+

+RtApiOss :: ~RtApiOss()

+{

+  if ( stream_.state != STREAM_CLOSED ) closeStream();

+}

+

+unsigned int RtApiOss :: getDeviceCount( void )

+{

+  int mixerfd = open( "/dev/mixer", O_RDWR, 0 );

+  if ( mixerfd == -1 ) {

+    errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";

+    error( RtError::WARNING );

+    return 0;

+  }

+

+  oss_sysinfo sysinfo;

+  if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {

+    close( mixerfd );

+    errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";

+    error( RtError::WARNING );

+    return 0;

+  }

+

+  close( mixerfd );

+  return sysinfo.numaudios;

+}

+

+RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )

+{

+  RtAudio::DeviceInfo info;

+  info.probed = false;

+

+  int mixerfd = open( "/dev/mixer", O_RDWR, 0 );

+  if ( mixerfd == -1 ) {

+    errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";

+    error( RtError::WARNING );

+    return info;

+  }

+

+  oss_sysinfo sysinfo;

+  int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );

+  if ( result == -1 ) {

+    close( mixerfd );

+    errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";

+    error( RtError::WARNING );

+    return info;

+  }

+

+  unsigned nDevices = sysinfo.numaudios;

+  if ( nDevices == 0 ) {

+    close( mixerfd );

+    errorText_ = "RtApiOss::getDeviceInfo: no devices found!";

+    error( RtError::INVALID_USE );

+  }

+

+  if ( device >= nDevices ) {

+    close( mixerfd );

+    errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";

+    error( RtError::INVALID_USE );

+  }

+

+  oss_audioinfo ainfo;

+  ainfo.dev = device;

+  result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );

+  close( mixerfd );

+  if ( result == -1 ) {

+    errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // Probe channels

+  if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;

+  if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;

+  if ( ainfo.caps & PCM_CAP_DUPLEX ) {

+    if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )

+      info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;

+  }

+

+  // Probe data formats ... do for input

+  unsigned long mask = ainfo.iformats;

+  if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )

+    info.nativeFormats |= RTAUDIO_SINT16;

+  if ( mask & AFMT_S8 )

+    info.nativeFormats |= RTAUDIO_SINT8;

+  if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )

+    info.nativeFormats |= RTAUDIO_SINT32;

+  if ( mask & AFMT_FLOAT )

+    info.nativeFormats |= RTAUDIO_FLOAT32;

+  if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )

+    info.nativeFormats |= RTAUDIO_SINT24;

+

+  // Check that we have at least one supported format

+  if ( info.nativeFormats == 0 ) {

+    errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+    return info;

+  }

+

+  // Probe the supported sample rates.

+  info.sampleRates.clear();

+  if ( ainfo.nrates ) {

+    for ( unsigned int i=0; i<ainfo.nrates; i++ ) {

+      for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {

+        if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {

+          info.sampleRates.push_back( SAMPLE_RATES[k] );

+          break;

+        }

+      }

+    }

+  }

+  else {

+    // Check min and max rate values;

+    for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {

+      if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] )

+        info.sampleRates.push_back( SAMPLE_RATES[k] );

+    }

+  }

+

+  if ( info.sampleRates.size() == 0 ) {

+    errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";

+    errorText_ = errorStream_.str();

+    error( RtError::WARNING );

+  }

+  else {

+    info.probed = true;

+    info.name = ainfo.name;

+  }

+

+  return info;

+}

+

+

+bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,

+                                  unsigned int firstChannel, unsigned int sampleRate,

+                                  RtAudioFormat format, unsigned int *bufferSize,

+                                  RtAudio::StreamOptions *options )

+{

+  int mixerfd = open( "/dev/mixer", O_RDWR, 0 );

+  if ( mixerfd == -1 ) {

+    errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";

+    return FAILURE;

+  }

+

+  oss_sysinfo sysinfo;

+  int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );

+  if ( result == -1 ) {

+    close( mixerfd );

+    errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";

+    return FAILURE;

+  }

+

+  unsigned nDevices = sysinfo.numaudios;

+  if ( nDevices == 0 ) {

+    // This should not happen because a check is made before this function is called.

+    close( mixerfd );

+    errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";

+    return FAILURE;

+  }

+

+  if ( device >= nDevices ) {

+    // This should not happen because a check is made before this function is called.

+    close( mixerfd );

+    errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";

+    return FAILURE;

+  }

+

+  oss_audioinfo ainfo;

+  ainfo.dev = device;

+  result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );

+  close( mixerfd );

+  if ( result == -1 ) {

+    errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Check if device supports input or output

+  if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||

+       ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {

+    if ( mode == OUTPUT )

+      errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";

+    else

+      errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  int flags = 0;

+  OssHandle *handle = (OssHandle *) stream_.apiHandle;

+  if ( mode == OUTPUT )

+    flags |= O_WRONLY;

+  else { // mode == INPUT

+    if (stream_.mode == OUTPUT && stream_.device[0] == device) {

+      // We just set the same device for playback ... close and reopen for duplex (OSS only).

+      close( handle->id[0] );

+      handle->id[0] = 0;

+      if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {

+        errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";

+        errorText_ = errorStream_.str();

+        return FAILURE;

+      }

+      // Check that the number previously set channels is the same.

+      if ( stream_.nUserChannels[0] != channels ) {

+        errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";

+        errorText_ = errorStream_.str();

+        return FAILURE;

+      }

+      flags |= O_RDWR;

+    }

+    else

+      flags |= O_RDONLY;

+  }

+

+  // Set exclusive access if specified.

+  if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;

+

+  // Try to open the device.

+  int fd;

+  fd = open( ainfo.devnode, flags, 0 );

+  if ( fd == -1 ) {

+    if ( errno == EBUSY )

+      errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";

+    else

+      errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // For duplex operation, specifically set this mode (this doesn't seem to work).

+  /*

+    if ( flags | O_RDWR ) {

+    result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );

+    if ( result == -1) {

+    errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+    }

+    }

+  */

+

+  // Check the device channel support.

+  stream_.nUserChannels[mode] = channels;

+  if ( ainfo.max_channels < (int)(channels + firstChannel) ) {

+    close( fd );

+    errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Set the number of channels.

+  int deviceChannels = channels + firstChannel;

+  result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );

+  if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {

+    close( fd );

+    errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+  stream_.nDeviceChannels[mode] = deviceChannels;

+

+  // Get the data format mask

+  int mask;

+  result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );

+  if ( result == -1 ) {

+    close( fd );

+    errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Determine how to set the device format.

+  stream_.userFormat = format;

+  int deviceFormat = -1;

+  stream_.doByteSwap[mode] = false;

+  if ( format == RTAUDIO_SINT8 ) {

+    if ( mask & AFMT_S8 ) {

+      deviceFormat = AFMT_S8;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT8;

+    }

+  }

+  else if ( format == RTAUDIO_SINT16 ) {

+    if ( mask & AFMT_S16_NE ) {

+      deviceFormat = AFMT_S16_NE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT16;

+    }

+    else if ( mask & AFMT_S16_OE ) {

+      deviceFormat = AFMT_S16_OE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT16;

+      stream_.doByteSwap[mode] = true;

+    }

+  }

+  else if ( format == RTAUDIO_SINT24 ) {

+    if ( mask & AFMT_S24_NE ) {

+      deviceFormat = AFMT_S24_NE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT24;

+    }

+    else if ( mask & AFMT_S24_OE ) {

+      deviceFormat = AFMT_S24_OE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT24;

+      stream_.doByteSwap[mode] = true;

+    }

+  }

+  else if ( format == RTAUDIO_SINT32 ) {

+    if ( mask & AFMT_S32_NE ) {

+      deviceFormat = AFMT_S32_NE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT32;

+    }

+    else if ( mask & AFMT_S32_OE ) {

+      deviceFormat = AFMT_S32_OE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT32;

+      stream_.doByteSwap[mode] = true;

+    }

+  }

+

+  if ( deviceFormat == -1 ) {

+    // The user requested format is not natively supported by the device.

+    if ( mask & AFMT_S16_NE ) {

+      deviceFormat = AFMT_S16_NE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT16;

+    }

+    else if ( mask & AFMT_S32_NE ) {

+      deviceFormat = AFMT_S32_NE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT32;

+    }

+    else if ( mask & AFMT_S24_NE ) {

+      deviceFormat = AFMT_S24_NE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT24;

+    }

+    else if ( mask & AFMT_S16_OE ) {

+      deviceFormat = AFMT_S16_OE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT16;

+      stream_.doByteSwap[mode] = true;

+    }

+    else if ( mask & AFMT_S32_OE ) {

+      deviceFormat = AFMT_S32_OE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT32;

+      stream_.doByteSwap[mode] = true;

+    }

+    else if ( mask & AFMT_S24_OE ) {

+      deviceFormat = AFMT_S24_OE;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT24;

+      stream_.doByteSwap[mode] = true;

+    }

+    else if ( mask & AFMT_S8) {

+      deviceFormat = AFMT_S8;

+      stream_.deviceFormat[mode] = RTAUDIO_SINT8;

+    }

+  }

+

+  if ( stream_.deviceFormat[mode] == 0 ) {

+    // This really shouldn't happen ...

+    close( fd );

+    errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Set the data format.

+  int temp = deviceFormat;

+  result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );

+  if ( result == -1 || deviceFormat != temp ) {

+    close( fd );

+    errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Attempt to set the buffer size.  According to OSS, the minimum

+  // number of buffers is two.  The supposed minimum buffer size is 16

+  // bytes, so that will be our lower bound.  The argument to this

+  // call is in the form 0xMMMMSSSS (hex), where the buffer size (in

+  // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.

+  // We'll check the actual value used near the end of the setup

+  // procedure.

+  int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;

+  if ( ossBufferBytes < 16 ) ossBufferBytes = 16;

+  int buffers = 0;

+  if ( options ) buffers = options->numberOfBuffers;

+  if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;

+  if ( buffers < 2 ) buffers = 3;

+  temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );

+  result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );

+  if ( result == -1 ) {

+    close( fd );

+    errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+  stream_.nBuffers = buffers;

+

+  // Save buffer size (in sample frames).

+  *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );

+  stream_.bufferSize = *bufferSize;

+

+  // Set the sample rate.

+  int srate = sampleRate;

+  result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );

+  if ( result == -1 ) {

+    close( fd );

+    errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+

+  // Verify the sample rate setup worked.

+  if ( abs( srate - sampleRate ) > 100 ) {

+    close( fd );

+    errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";

+    errorText_ = errorStream_.str();

+    return FAILURE;

+  }

+  stream_.sampleRate = sampleRate;

+

+  if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {

+    // We're doing duplex setup here.

+    stream_.deviceFormat[0] = stream_.deviceFormat[1];

+    stream_.nDeviceChannels[0] = deviceChannels;

+  }

+

+  // Set interleaving parameters.

+  stream_.userInterleaved = true;

+  stream_.deviceInterleaved[mode] =  true;

+  if ( options && options->flags & RTAUDIO_NONINTERLEAVED )

+    stream_.userInterleaved = false;

+

+  // Set flags for buffer conversion

+  stream_.doConvertBuffer[mode] = false;

+  if ( stream_.userFormat != stream_.deviceFormat[mode] )

+    stream_.doConvertBuffer[mode] = true;

+  if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )

+    stream_.doConvertBuffer[mode] = true;

+  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&

+       stream_.nUserChannels[mode] > 1 )

+    stream_.doConvertBuffer[mode] = true;

+

+  // Allocate the stream handles if necessary and then save.

+  if ( stream_.apiHandle == 0 ) {

+    try {

+      handle = new OssHandle;

+    }

+    catch ( std::bad_alloc& ) {

+      errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";

+      goto error;

+    }

+

+    if ( pthread_cond_init( &handle->runnable, NULL ) ) {

+      errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";

+      goto error;

+    }

+

+    stream_.apiHandle = (void *) handle;

+  }

+  else {

+    handle = (OssHandle *) stream_.apiHandle;

+  }

+  handle->id[mode] = fd;

+

+  // Allocate necessary internal buffers.

+  unsigned long bufferBytes;

+  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );

+  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );

+  if ( stream_.userBuffer[mode] == NULL ) {

+    errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";

+    goto error;

+  }

+

+  if ( stream_.doConvertBuffer[mode] ) {

+

+    bool makeBuffer = true;

+    bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );

+    if ( mode == INPUT ) {

+      if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {

+        unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );

+        if ( bufferBytes <= bytesOut ) makeBuffer = false;

+      }

+    }

+

+    if ( makeBuffer ) {

+      bufferBytes *= *bufferSize;

+      if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );

+      stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );

+      if ( stream_.deviceBuffer == NULL ) {

+        errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";

+        goto error;

+      }

+    }

+  }

+

+  stream_.device[mode] = device;

+  stream_.state = STREAM_STOPPED;

+

+  // Setup the buffer conversion information structure.

+  if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );

+

+  // Setup thread if necessary.

+  if ( stream_.mode == OUTPUT && mode == INPUT ) {

+    // We had already set up an output stream.

+    stream_.mode = DUPLEX;

+    if ( stream_.device[0] == device ) handle->id[0] = fd;

+  }

+  else {

+    stream_.mode = mode;

+

+    // Setup callback thread.

+    stream_.callbackInfo.object = (void *) this;

+

+    // Set the thread attributes for joinable and realtime scheduling

+    // priority.  The higher priority will only take affect if the

+    // program is run as root or suid.

+    pthread_attr_t attr;

+    pthread_attr_init( &attr );

+    pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );

+#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)

+    if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {

+      struct sched_param param;

+      int priority = options->priority;

+      int min = sched_get_priority_min( SCHED_RR );

+      int max = sched_get_priority_max( SCHED_RR );

+      if ( priority < min ) priority = min;

+      else if ( priority > max ) priority = max;

+      param.sched_priority = priority;

+      pthread_attr_setschedparam( &attr, &param );

+      pthread_attr_setschedpolicy( &attr, SCHED_RR );

+    }

+    else

+      pthread_attr_setschedpolicy( &attr, SCHED_OTHER );

+#else

+    pthread_attr_setschedpolicy( &attr, SCHED_OTHER );

+#endif

+

+    stream_.callbackInfo.isRunning = true;

+    result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );

+    pthread_attr_destroy( &attr );

+    if ( result ) {

+      stream_.callbackInfo.isRunning = false;

+      errorText_ = "RtApiOss::error creating callback thread!";

+      goto error;

+    }

+  }

+

+  return SUCCESS;

+

+ error:

+  if ( handle ) {

+    pthread_cond_destroy( &handle->runnable );

+    if ( handle->id[0] ) close( handle->id[0] );

+    if ( handle->id[1] ) close( handle->id[1] );

+    delete handle;

+    stream_.apiHandle = 0;

+  }

+

+  for ( int i=0; i<2; i++ ) {

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  return FAILURE;

+}

+

+void RtApiOss :: closeStream()

+{

+  if ( stream_.state == STREAM_CLOSED ) {

+    errorText_ = "RtApiOss::closeStream(): no open stream to close!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  OssHandle *handle = (OssHandle *) stream_.apiHandle;

+  stream_.callbackInfo.isRunning = false;

+  MUTEX_LOCK( &stream_.mutex );

+  if ( stream_.state == STREAM_STOPPED )

+    pthread_cond_signal( &handle->runnable );

+  MUTEX_UNLOCK( &stream_.mutex );

+  pthread_join( stream_.callbackInfo.thread, NULL );

+

+  if ( stream_.state == STREAM_RUNNING ) {

+    if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )

+      ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );

+    else

+      ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );

+    stream_.state = STREAM_STOPPED;

+  }

+

+  if ( handle ) {

+    pthread_cond_destroy( &handle->runnable );

+    if ( handle->id[0] ) close( handle->id[0] );

+    if ( handle->id[1] ) close( handle->id[1] );

+    delete handle;

+    stream_.apiHandle = 0;

+  }

+

+  for ( int i=0; i<2; i++ ) {

+    if ( stream_.userBuffer[i] ) {

+      free( stream_.userBuffer[i] );

+      stream_.userBuffer[i] = 0;

+    }

+  }

+

+  if ( stream_.deviceBuffer ) {

+    free( stream_.deviceBuffer );

+    stream_.deviceBuffer = 0;

+  }

+

+  stream_.mode = UNINITIALIZED;

+  stream_.state = STREAM_CLOSED;

+}

+

+void RtApiOss :: startStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_RUNNING ) {

+    errorText_ = "RtApiOss::startStream(): the stream is already running!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  stream_.state = STREAM_RUNNING;

+

+  // No need to do anything else here ... OSS automatically starts

+  // when fed samples.

+

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  OssHandle *handle = (OssHandle *) stream_.apiHandle;

+  pthread_cond_signal( &handle->runnable );

+}

+

+void RtApiOss :: stopStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

+    errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  // The state might change while waiting on a mutex.

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_UNLOCK( &stream_.mutex );

+    return;

+  }

+

+  int result = 0;

+  OssHandle *handle = (OssHandle *) stream_.apiHandle;

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+

+    // Flush the output with zeros a few times.

+    char *buffer;

+    int samples;

+    RtAudioFormat format;

+

+    if ( stream_.doConvertBuffer[0] ) {

+      buffer = stream_.deviceBuffer;

+      samples = stream_.bufferSize * stream_.nDeviceChannels[0];

+      format = stream_.deviceFormat[0];

+    }

+    else {

+      buffer = stream_.userBuffer[0];

+      samples = stream_.bufferSize * stream_.nUserChannels[0];

+      format = stream_.userFormat;

+    }

+

+    memset( buffer, 0, samples * formatBytes(format) );

+    for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {

+      result = write( handle->id[0], buffer, samples * formatBytes(format) );

+      if ( result == -1 ) {

+        errorText_ = "RtApiOss::stopStream: audio write error.";

+        error( RtError::WARNING );

+      }

+    }

+

+    result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );

+    if ( result == -1 ) {

+      errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+    handle->triggered = false;

+  }

+

+  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {

+    result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );

+    if ( result == -1 ) {

+      errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+ unlock:

+  stream_.state = STREAM_STOPPED;

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  if ( result != -1 ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiOss :: abortStream()

+{

+  verifyStream();

+  if ( stream_.state == STREAM_STOPPED ) {

+    errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  // The state might change while waiting on a mutex.

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_UNLOCK( &stream_.mutex );

+    return;

+  }

+

+  int result = 0;

+  OssHandle *handle = (OssHandle *) stream_.apiHandle;

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+    result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );

+    if ( result == -1 ) {

+      errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+    handle->triggered = false;

+  }

+

+  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {

+    result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );

+    if ( result == -1 ) {

+      errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";

+      errorText_ = errorStream_.str();

+      goto unlock;

+    }

+  }

+

+ unlock:

+  stream_.state = STREAM_STOPPED;

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  if ( result != -1 ) return;

+  error( RtError::SYSTEM_ERROR );

+}

+

+void RtApiOss :: callbackEvent()

+{

+  OssHandle *handle = (OssHandle *) stream_.apiHandle;

+  if ( stream_.state == STREAM_STOPPED ) {

+    MUTEX_LOCK( &stream_.mutex );

+    pthread_cond_wait( &handle->runnable, &stream_.mutex );

+    if ( stream_.state != STREAM_RUNNING ) {

+      MUTEX_UNLOCK( &stream_.mutex );

+      return;

+    }

+    MUTEX_UNLOCK( &stream_.mutex );

+  }

+

+  if ( stream_.state == STREAM_CLOSED ) {

+    errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";

+    error( RtError::WARNING );

+    return;

+  }

+

+  // Invoke user callback to get fresh output data.

+  int doStopStream = 0;

+  RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;

+  double streamTime = getStreamTime();

+  RtAudioStreamStatus status = 0;

+  if ( stream_.mode != INPUT && handle->xrun[0] == true ) {

+    status |= RTAUDIO_OUTPUT_UNDERFLOW;

+    handle->xrun[0] = false;

+  }

+  if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {

+    status |= RTAUDIO_INPUT_OVERFLOW;

+    handle->xrun[1] = false;

+  }

+  doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],

+                           stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );

+  if ( doStopStream == 2 ) {

+    this->abortStream();

+    return;

+  }

+

+  MUTEX_LOCK( &stream_.mutex );

+

+  // The state might change while waiting on a mutex.

+  if ( stream_.state == STREAM_STOPPED ) goto unlock;

+

+  int result;

+  char *buffer;

+  int samples;

+  RtAudioFormat format;

+

+  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

+

+    // Setup parameters and do buffer conversion if necessary.

+    if ( stream_.doConvertBuffer[0] ) {

+      buffer = stream_.deviceBuffer;

+      convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );

+      samples = stream_.bufferSize * stream_.nDeviceChannels[0];

+      format = stream_.deviceFormat[0];

+    }

+    else {

+      buffer = stream_.userBuffer[0];

+      samples = stream_.bufferSize * stream_.nUserChannels[0];

+      format = stream_.userFormat;

+    }

+

+    // Do byte swapping if necessary.

+    if ( stream_.doByteSwap[0] )

+      byteSwapBuffer( buffer, samples, format );

+

+    if ( stream_.mode == DUPLEX && handle->triggered == false ) {

+      int trig = 0;

+      ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );

+      result = write( handle->id[0], buffer, samples * formatBytes(format) );

+      trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;

+      ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );

+      handle->triggered = true;

+    }

+    else

+      // Write samples to device.

+      result = write( handle->id[0], buffer, samples * formatBytes(format) );

+

+    if ( result == -1 ) {

+      // We'll assume this is an underrun, though there isn't a

+      // specific means for determining that.

+      handle->xrun[0] = true;

+      errorText_ = "RtApiOss::callbackEvent: audio write error.";

+      error( RtError::WARNING );

+      // Continue on to input section.

+    }

+  }

+

+  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {

+

+    // Setup parameters.

+    if ( stream_.doConvertBuffer[1] ) {

+      buffer = stream_.deviceBuffer;

+      samples = stream_.bufferSize * stream_.nDeviceChannels[1];

+      format = stream_.deviceFormat[1];

+    }

+    else {

+      buffer = stream_.userBuffer[1];

+      samples = stream_.bufferSize * stream_.nUserChannels[1];

+      format = stream_.userFormat;

+    }

+

+    // Read samples from device.

+    result = read( handle->id[1], buffer, samples * formatBytes(format) );

+

+    if ( result == -1 ) {

+      // We'll assume this is an overrun, though there isn't a

+      // specific means for determining that.

+      handle->xrun[1] = true;

+      errorText_ = "RtApiOss::callbackEvent: audio read error.";

+      error( RtError::WARNING );

+      goto unlock;

+    }

+

+    // Do byte swapping if necessary.

+    if ( stream_.doByteSwap[1] )

+      byteSwapBuffer( buffer, samples, format );

+

+    // Do buffer conversion if necessary.

+    if ( stream_.doConvertBuffer[1] )

+      convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );

+  }

+

+ unlock:

+  MUTEX_UNLOCK( &stream_.mutex );

+

+  RtApi::tickStreamTime();

+  if ( doStopStream == 1 ) this->stopStream();

+}

+

+extern "C" void *ossCallbackHandler( void *ptr )

+{

+  CallbackInfo *info = (CallbackInfo *) ptr;

+  RtApiOss *object = (RtApiOss *) info->object;

+  bool *isRunning = &info->isRunning;

+

+  while ( *isRunning == true ) {

+    pthread_testcancel();

+    object->callbackEvent();

+  }

+

+  pthread_exit( NULL );

+}

+

+//******************** End of __LINUX_OSS__ *********************//

+#endif

+

+

+// *************************************************** //

+//

+// Protected common (OS-independent) RtAudio methods.

+//

+// *************************************************** //

+

+// This method can be modified to control the behavior of error

+// message printing.

+void RtApi :: error( RtError::Type type )

+{

+  errorStream_.str(""); // clear the ostringstream

+  if ( type == RtError::WARNING && showWarnings_ == true )

+    std::cerr << '\n' << errorText_ << "\n\n";

+  else if ( type != RtError::WARNING )

+    throw( RtError( errorText_, type ) );

+}

+

+void RtApi :: verifyStream()

+{

+  if ( stream_.state == STREAM_CLOSED ) {

+    errorText_ = "RtApi:: a stream is not open!";

+    error( RtError::INVALID_USE );

+  }

+}

+

+void RtApi :: clearStreamInfo()

+{

+  stream_.mode = UNINITIALIZED;

+  stream_.state = STREAM_CLOSED;

+  stream_.sampleRate = 0;

+  stream_.bufferSize = 0;

+  stream_.nBuffers = 0;

+  stream_.userFormat = 0;

+  stream_.userInterleaved = true;

+  stream_.streamTime = 0.0;

+  stream_.apiHandle = 0;

+  stream_.deviceBuffer = 0;

+  stream_.callbackInfo.callback = 0;

+  stream_.callbackInfo.userData = 0;

+  stream_.callbackInfo.isRunning = false;

+  for ( int i=0; i<2; i++ ) {

+    stream_.device[i] = 11111;

+    stream_.doConvertBuffer[i] = false;

+    stream_.deviceInterleaved[i] = true;

+    stream_.doByteSwap[i] = false;

+    stream_.nUserChannels[i] = 0;

+    stream_.nDeviceChannels[i] = 0;

+    stream_.channelOffset[i] = 0;

+    stream_.deviceFormat[i] = 0;

+    stream_.latency[i] = 0;

+    stream_.userBuffer[i] = 0;

+    stream_.convertInfo[i].channels = 0;

+    stream_.convertInfo[i].inJump = 0;

+    stream_.convertInfo[i].outJump = 0;

+    stream_.convertInfo[i].inFormat = 0;

+    stream_.convertInfo[i].outFormat = 0;

+    stream_.convertInfo[i].inOffset.clear();

+    stream_.convertInfo[i].outOffset.clear();

+  }

+}

+

+unsigned int RtApi :: formatBytes( RtAudioFormat format )

+{

+  if ( format == RTAUDIO_SINT16 )

+    return 2;

+  else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 ||

+            format == RTAUDIO_FLOAT32 )

+    return 4;

+  else if ( format == RTAUDIO_FLOAT64 )

+    return 8;

+  else if ( format == RTAUDIO_SINT8 )

+    return 1;

+

+  errorText_ = "RtApi::formatBytes: undefined format.";

+  error( RtError::WARNING );

+

+  return 0;

+}

+

+void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )

+{

+  if ( mode == INPUT ) { // convert device to user buffer

+    stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];

+    stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];

+    stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];

+    stream_.convertInfo[mode].outFormat = stream_.userFormat;

+  }

+  else { // convert user to device buffer

+    stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];

+    stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];

+    stream_.convertInfo[mode].inFormat = stream_.userFormat;

+    stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];

+  }

+

+  if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )

+    stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;

+  else

+    stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;

+

+  // Set up the interleave/deinterleave offsets.

+  if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {

+    if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||

+         ( mode == INPUT && stream_.userInterleaved ) ) {

+      for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {

+        stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );

+        stream_.convertInfo[mode].outOffset.push_back( k );

+        stream_.convertInfo[mode].inJump = 1;

+      }

+    }

+    else {

+      for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {

+        stream_.convertInfo[mode].inOffset.push_back( k );

+        stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );

+        stream_.convertInfo[mode].outJump = 1;

+      }

+    }

+  }

+  else { // no (de)interleaving

+    if ( stream_.userInterleaved ) {

+      for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {

+        stream_.convertInfo[mode].inOffset.push_back( k );

+        stream_.convertInfo[mode].outOffset.push_back( k );

+      }

+    }

+    else {

+      for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {

+        stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );

+        stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );

+        stream_.convertInfo[mode].inJump = 1;

+        stream_.convertInfo[mode].outJump = 1;

+      }

+    }

+  }

+

+  // Add channel offset.

+  if ( firstChannel > 0 ) {

+    if ( stream_.deviceInterleaved[mode] ) {

+      if ( mode == OUTPUT ) {

+        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )

+          stream_.convertInfo[mode].outOffset[k] += firstChannel;

+      }

+      else {

+        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )

+          stream_.convertInfo[mode].inOffset[k] += firstChannel;

+      }

+    }

+    else {

+      if ( mode == OUTPUT ) {

+        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )

+          stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );

+      }

+      else {

+        for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )

+          stream_.convertInfo[mode].inOffset[k] += ( firstChannel  * stream_.bufferSize );

+      }

+    }

+  }

+}

+

+void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )

+{

+  // This function does format conversion, input/output channel compensation, and

+  // data interleaving/deinterleaving.  24-bit integers are assumed to occupy

+  // the lower three bytes of a 32-bit integer.

+

+  // Clear our device buffer when in/out duplex device channels are different

+  if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&

+       ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )

+    memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );

+

+  int j;

+  if (info.outFormat == RTAUDIO_FLOAT64) {

+    Float64 scale;

+    Float64 *out = (Float64 *)outBuffer;

+

+    if (info.inFormat == RTAUDIO_SINT8) {

+      signed char *in = (signed char *)inBuffer;

+      scale = 1.0 / 127.5;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];

+          out[info.outOffset[j]] += 0.5;

+          out[info.outOffset[j]] *= scale;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT16) {

+      Int16 *in = (Int16 *)inBuffer;

+      scale = 1.0 / 32767.5;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];

+          out[info.outOffset[j]] += 0.5;

+          out[info.outOffset[j]] *= scale;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT24) {

+      Int32 *in = (Int32 *)inBuffer;

+      scale = 1.0 / 8388607.5;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]] & 0x00ffffff);

+          out[info.outOffset[j]] += 0.5;

+          out[info.outOffset[j]] *= scale;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT32) {

+      Int32 *in = (Int32 *)inBuffer;

+      scale = 1.0 / 2147483647.5;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];

+          out[info.outOffset[j]] += 0.5;

+          out[info.outOffset[j]] *= scale;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT32) {

+      Float32 *in = (Float32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT64) {

+      // Channel compensation and/or (de)interleaving only.

+      Float64 *in = (Float64 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = in[info.inOffset[j]];

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+  }

+  else if (info.outFormat == RTAUDIO_FLOAT32) {

+    Float32 scale;

+    Float32 *out = (Float32 *)outBuffer;

+

+    if (info.inFormat == RTAUDIO_SINT8) {

+      signed char *in = (signed char *)inBuffer;

+      scale = (Float32) ( 1.0 / 127.5 );

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];

+          out[info.outOffset[j]] += 0.5;

+          out[info.outOffset[j]] *= scale;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT16) {

+      Int16 *in = (Int16 *)inBuffer;

+      scale = (Float32) ( 1.0 / 32767.5 );

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];

+          out[info.outOffset[j]] += 0.5;

+          out[info.outOffset[j]] *= scale;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT24) {

+      Int32 *in = (Int32 *)inBuffer;

+      scale = (Float32) ( 1.0 / 8388607.5 );

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]] & 0x00ffffff);

+          out[info.outOffset[j]] += 0.5;

+          out[info.outOffset[j]] *= scale;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT32) {

+      Int32 *in = (Int32 *)inBuffer;

+      scale = (Float32) ( 1.0 / 2147483647.5 );

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];

+          out[info.outOffset[j]] += 0.5;

+          out[info.outOffset[j]] *= scale;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT32) {

+      // Channel compensation and/or (de)interleaving only.

+      Float32 *in = (Float32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = in[info.inOffset[j]];

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT64) {

+      Float64 *in = (Float64 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+  }

+  else if (info.outFormat == RTAUDIO_SINT32) {

+    Int32 *out = (Int32 *)outBuffer;

+    if (info.inFormat == RTAUDIO_SINT8) {

+      signed char *in = (signed char *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];

+          out[info.outOffset[j]] <<= 24;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT16) {

+      Int16 *in = (Int16 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];

+          out[info.outOffset[j]] <<= 16;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT24) { // Hmmm ... we could just leave it in the lower 3 bytes

+      Int32 *in = (Int32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];

+          out[info.outOffset[j]] <<= 8;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT32) {

+      // Channel compensation and/or (de)interleaving only.

+      Int32 *in = (Int32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = in[info.inOffset[j]];

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT32) {

+      Float32 *in = (Float32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT64) {

+      Float64 *in = (Float64 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+  }

+  else if (info.outFormat == RTAUDIO_SINT24) {

+    Int32 *out = (Int32 *)outBuffer;

+    if (info.inFormat == RTAUDIO_SINT8) {

+      signed char *in = (signed char *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];

+          out[info.outOffset[j]] <<= 16;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT16) {

+      Int16 *in = (Int16 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];

+          out[info.outOffset[j]] <<= 8;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT24) {

+      // Channel compensation and/or (de)interleaving only.

+      Int32 *in = (Int32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = in[info.inOffset[j]];

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT32) {

+      Int32 *in = (Int32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];

+          out[info.outOffset[j]] >>= 8;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT32) {

+      Float32 *in = (Float32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT64) {

+      Float64 *in = (Float64 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+  }

+  else if (info.outFormat == RTAUDIO_SINT16) {

+    Int16 *out = (Int16 *)outBuffer;

+    if (info.inFormat == RTAUDIO_SINT8) {

+      signed char *in = (signed char *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];

+          out[info.outOffset[j]] <<= 8;

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT16) {

+      // Channel compensation and/or (de)interleaving only.

+      Int16 *in = (Int16 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = in[info.inOffset[j]];

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT24) {

+      Int32 *in = (Int32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 8) & 0x0000ffff);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT32) {

+      Int32 *in = (Int32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT32) {

+      Float32 *in = (Float32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT64) {

+      Float64 *in = (Float64 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+  }

+  else if (info.outFormat == RTAUDIO_SINT8) {

+    signed char *out = (signed char *)outBuffer;

+    if (info.inFormat == RTAUDIO_SINT8) {

+      // Channel compensation and/or (de)interleaving only.

+      signed char *in = (signed char *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = in[info.inOffset[j]];

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    if (info.inFormat == RTAUDIO_SINT16) {

+      Int16 *in = (Int16 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT24) {

+      Int32 *in = (Int32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 16) & 0x000000ff);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_SINT32) {

+      Int32 *in = (Int32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT32) {

+      Float32 *in = (Float32 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+    else if (info.inFormat == RTAUDIO_FLOAT64) {

+      Float64 *in = (Float64 *)inBuffer;

+      for (unsigned int i=0; i<stream_.bufferSize; i++) {

+        for (j=0; j<info.channels; j++) {

+          out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);

+        }

+        in += info.inJump;

+        out += info.outJump;

+      }

+    }

+  }

+}

+

+  //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }

+  //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }

+  //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }

+

+void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )

+{

+  register char val;

+  register char *ptr;

+

+  ptr = buffer;

+  if ( format == RTAUDIO_SINT16 ) {

+    for ( unsigned int i=0; i<samples; i++ ) {

+      // Swap 1st and 2nd bytes.

+      val = *(ptr);

+      *(ptr) = *(ptr+1);

+      *(ptr+1) = val;

+

+      // Increment 2 bytes.

+      ptr += 2;

+    }

+  }

+  else if ( format == RTAUDIO_SINT24 ||

+            format == RTAUDIO_SINT32 ||

+            format == RTAUDIO_FLOAT32 ) {

+    for ( unsigned int i=0; i<samples; i++ ) {

+      // Swap 1st and 4th bytes.

+      val = *(ptr);

+      *(ptr) = *(ptr+3);

+      *(ptr+3) = val;

+

+      // Swap 2nd and 3rd bytes.

+      ptr += 1;

+      val = *(ptr);

+      *(ptr) = *(ptr+1);

+      *(ptr+1) = val;

+

+      // Increment 3 more bytes.

+      ptr += 3;

+    }

+  }

+  else if ( format == RTAUDIO_FLOAT64 ) {

+    for ( unsigned int i=0; i<samples; i++ ) {

+      // Swap 1st and 8th bytes

+      val = *(ptr);

+      *(ptr) = *(ptr+7);

+      *(ptr+7) = val;

+

+      // Swap 2nd and 7th bytes

+      ptr += 1;

+      val = *(ptr);

+      *(ptr) = *(ptr+5);

+      *(ptr+5) = val;

+

+      // Swap 3rd and 6th bytes

+      ptr += 1;

+      val = *(ptr);

+      *(ptr) = *(ptr+3);

+      *(ptr+3) = val;

+

+      // Swap 4th and 5th bytes

+      ptr += 1;

+      val = *(ptr);

+      *(ptr) = *(ptr+1);

+      *(ptr+1) = val;

+

+      // Increment 5 more bytes.

+      ptr += 5;

+    }

+  }

+}

+

+  // Indentation settings for Vim and Emacs

+  //

+  // Local Variables:

+  // c-basic-offset: 2

+  // indent-tabs-mode: nil

+  // End:

+  //

+  // vim: et sts=2 sw=2

+