RtAudio.cpp

/*
 * Copyright (c) 2010-2016 Stephane Poirier
 *
 * stephane.poirier@oifii.org
 *
 * Stephane Poirier
 * 3532 rue Ste-Famille, #3
 * Montreal, QC, H2X 2L1
 *
 * This program is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 3 of the License, or
 * (at your option) any later version.
 *
 * This program is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with this program.  If not, see <http://www.gnu.org/licenses/>.
 *
 */
// RtAudio: Version 4.0.7

#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() );
  info.name.append( (const char *)mname, strlen(mname) );
  info.name.append( ": " );
  CFRelease( cfname );

  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() );
  info.name.append( (const char *)name, strlen(name) );
  CFRelease( cfname );

  // 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] ) ) {

    result = AudioDeviceStop( handle->id[1], callbackHandler );
    if ( result != noErr ) {
      errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
      errorText_ = errorStream_.str();
      goto unlock;
    }
  }

  stream_.state = STREAM_STOPPED;

 unlock:
  MUTEX_UNLOCK( &stream_.mutex );

  if ( result == noErr ) return;
  error( RtError::SYSTEM_ERROR );
}

void RtApiCore :: abortStream( void )
{
  verifyStream();
  if ( stream_.state == STREAM_STOPPED ) {
    errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
    error( RtError::WARNING );
    return;
  }

  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
  handle->drainCounter = 1;

  stopStream();
}

bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
                                 const AudioBufferList *inBufferList,
                                 const AudioBufferList *outBufferList )
{
  if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
  if ( stream_.state == STREAM_CLOSED ) {
    errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
    error( RtError::WARNING );
    return FAILURE;
  }

  CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
  CoreHandle *handle = (CoreHandle *) stream_.apiHandle;

  // Check if we were draining the stream and signal is finished.
  if ( handle->drainCounter > 3 ) {
    if ( handle->internalDrain == false )
      pthread_cond_signal( &handle->condition );
    else
      stopStream();
    return SUCCESS;
  }

  MUTEX_LOCK( &stream_.mutex );

  // The state might change while waiting on a mutex.
  if ( stream_.state == STREAM_STOPPED ) {
    MUTEX_UNLOCK( &stream_.mutex );
    return SUCCESS;
  }

  AudioDeviceID outputDevice = handle->id[0];

  // Invoke user callback to get fresh output data UNLESS we are
  // draining stream or duplex mode AND the input/output devices are
  // different AND this function is called for the input device.
  if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
    RtAudioCallback callback = (RtAudioCallback) info->callback;
    double streamTime = getStreamTime();
    RtAudioStreamStatus status = 0;
    if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
      status |= RTAUDIO_OUTPUT_UNDERFLOW;
      handle->xrun[0] = false;
    }
    if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
      status |= RTAUDIO_INPUT_OVERFLOW;
      handle->xrun[1] = false;
    }
    handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
                                     stream_.bufferSize, streamTime, status, info->userData );
    if ( handle->drainCounter == 2 ) {
      MUTEX_UNLOCK( &stream_.mutex );
      abortStream();
      return SUCCESS;
    }
    else if ( handle->drainCounter == 1 )
      handle->internalDrain = true;
  }

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

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

      if ( handle->nStreams[0] == 1 ) {
        memset( outBufferList->mBuffers[handle->iStream[0]].mData,
                0,
                outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
      }
      else { // fill multiple streams with zeros
        for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
          memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
                  0,
                  outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
        }
      }
    }
    else if ( handle->nStreams[0] == 1 ) {
      if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
        convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
                       stream_.userBuffer[0], stream_.convertInfo[0] );
      }
      else { // copy from user buffer
        memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
                stream_.userBuffer[0],
                outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
      }
    }
    else { // fill multiple streams
      Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
      if ( stream_.doConvertBuffer[0] ) {
        convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
        inBuffer = (Float32 *) stream_.deviceBuffer;
      }

      if ( stream_.deviceInterleaved[0] == false ) { // mono mode
        UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
        for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
          memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
                  (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
        }
      }
      else { // fill multiple multi-channel streams with interleaved data
        UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
        Float32 *out, *in;

        bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
        UInt32 inChannels = stream_.nUserChannels[0];
        if ( stream_.doConvertBuffer[0] ) {
          inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
          inChannels = stream_.nDeviceChannels[0];
        }

        if ( inInterleaved ) inOffset = 1;
        else inOffset = stream_.bufferSize;

        channelsLeft = inChannels;
        for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
          in = inBuffer;
          out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
          streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;

          outJump = 0;
          // Account for possible channel offset in first stream
          if ( i == 0 && stream_.channelOffset[0] > 0 ) {
            streamChannels -= stream_.channelOffset[0];
            outJump = stream_.channelOffset[0];
            out += outJump;
          }

          // Account for possible unfilled channels at end of the last stream
          if ( streamChannels > channelsLeft ) {
            outJump = streamChannels - channelsLeft;
            streamChannels = channelsLeft;
          }

          // Determine input buffer offsets and skips
          if ( inInterleaved ) {
            inJump = inChannels;
            in += inChannels - channelsLeft;
          }
          else {
            inJump = 1;
            in += (inChannels - channelsLeft) * inOffset;
          }

          for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
            for ( unsigned int j=0; j<streamChannels; j++ ) {
              *out++ = in[j*inOffset];
            }
            out += outJump;
            in += inJump;
          }
          channelsLeft -= streamChannels;
        }
      }
    }

    if ( handle->drainCounter ) {
      handle->drainCounter++;
      goto unlock;
    }
  }

  AudioDeviceID inputDevice;
  inputDevice = handle->id[1];
  if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {

    if ( handle->nStreams[1] == 1 ) {
      if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer
        convertBuffer( stream_.userBuffer[1],
                       (char *) inBufferList->mBuffers[handle->iStream[1]].mData,
                       stream_.convertInfo[1] );
      }
      else { // copy to user buffer
        memcpy( stream_.userBuffer[1],
                inBufferList->mBuffers[handle->iStream[1]].mData,
                inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
      }
    }
    else { // read from multiple streams
      Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];
      if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;

      if ( stream_.deviceInterleaved[1] == false ) { // mono mode
        UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
        for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
          memcpy( (void *)&outBuffer[i*stream_.bufferSize],
                  inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
        }
      }
      else { // read from multiple multi-channel streams
        UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;
        Float32 *out, *in;

        bool outInterleaved = ( stream_.userInterleaved ) ? true : false;
        UInt32 outChannels = stream_.nUserChannels[1];
        if ( stream_.doConvertBuffer[1] ) {
          outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
          outChannels = stream_.nDeviceChannels[1];
        }

        if ( outInterleaved ) outOffset = 1;
        else outOffset = stream_.bufferSize;

        channelsLeft = outChannels;
        for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {
          out = outBuffer;
          in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;
          streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;

          inJump = 0;
          // Account for possible channel offset in first stream
          if ( i == 0 && stream_.channelOffset[1] > 0 ) {
            streamChannels -= stream_.channelOffset[1];
            inJump = stream_.channelOffset[1];
            in += inJump;
          }

          // Account for possible unread channels at end of the last stream
          if ( streamChannels > channelsLeft ) {
            inJump = streamChannels - channelsLeft;
            streamChannels = channelsLeft;
          }

          // Determine output buffer offsets and skips
          if ( outInterleaved ) {
            outJump = outChannels;
            out += outChannels - channelsLeft;
          }
          else {
            outJump = 1;
            out += (outChannels - channelsLeft) * outOffset;
          }

          for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
            for ( unsigned int j=0; j<streamChannels; j++ ) {
              out[j*outOffset] = *in++;
            }
            out += outJump;
            in += inJump;
          }
          channelsLeft -= streamChannels;
        }
      }
      
      if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
        convertBuffer( stream_.userBuffer[1],
                       stream_.deviceBuffer,
                       stream_.convertInfo[1] );
      }
    }
  }

 unlock:
  MUTEX_UNLOCK( &stream_.mutex );

  RtApi::tickStreamTime();
  return SUCCESS;
}

const char* RtApiCore :: getErrorCode( OSStatus code )
{
  switch( code ) {

  case kAudioHardwareNotRunningError:
    return "kAudioHardwareNotRunningError";

  case kAudioHardwareUnspecifiedError:
    return "kAudioHardwareUnspecifiedError";

  case kAudioHardwareUnknownPropertyError:
    return "kAudioHardwareUnknownPropertyError";

  case kAudioHardwareBadPropertySizeError:
    return "kAudioHardwareBadPropertySizeError";

  case kAudioHardwareIllegalOperationError:
    return "kAudioHardwareIllegalOperationError";

  case kAudioHardwareBadObjectError:
    return "kAudioHardwareBadObjectError";

  case kAudioHardwareBadDeviceError:
    return "kAudioHardwareBadDeviceError";

  case kAudioHardwareBadStreamError:
    return "kAudioHardwareBadStreamError";

  case kAudioHardwareUnsupportedOperationError:
    return "kAudioHardwareUnsupportedOperationError";

  case kAudioDeviceUnsupportedFormatError:
    return "kAudioDeviceUnsupportedFormatError";

  case kAudioDevicePermissionsError:
    return "kAudioDevicePermissionsError";

  default:
    return "CoreAudio unknown error";
  }
}

  //******************** End of __MACOSX_CORE__ *********************//
#endif

#if defined(__UNIX_JACK__)

// JACK is a low-latency audio server, originally written for the
// GNU/Linux operating system and now also ported to OS-X. It can
// connect a number of different applications to an audio device, as
// well as allowing them to share audio between themselves.
//
// When using JACK with RtAudio, "devices" refer to JACK clients that
// have ports connected to the server.  The JACK server is typically
// started in a terminal as follows:
//
// .jackd -d alsa -d hw:0
//
// or through an interface program such as qjackctl.  Many of the
// parameters normally set for a stream are fixed by the JACK server
// and can be specified when the JACK server is started.  In
// particular,
//
// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
//
// specifies a sample rate of 44100 Hz, a buffer size of 512 sample
// frames, and number of buffers = 4.  Once the server is running, it
// is not possible to override these values.  If the values are not
// specified in the command-line, the JACK server uses default values.
//
// The JACK server does not have to be running when an instance of
// RtApiJack is created, though the function getDeviceCount() will
// report 0 devices found until JACK has been started.  When no
// devices are available (i.e., the JACK server is not running), a
// stream cannot be opened.

#include <jack/jack.h>
#include <unistd.h>
#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 | JackUseExactName ); //JackNullOption;
  jack_status_t *status = NULL;
  jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
  if ( client == 0 ) return 0;

  const char **ports;
  std::string port, previousPort;
  unsigned int nChannels = 0, nDevices = 0;
  ports = jack_get_ports( client, NULL, NULL, 0 );
  if ( ports ) {
    // Parse the port names up to the first colon (:).
    size_t iColon = 0;
    do {
      port = (char *) ports[ nChannels ];
      iColon = port.find(":");
      if ( iColon != std::string::npos ) {
        port = port.substr( 0, iColon + 1 );
        if ( port != previousPort ) {
          nDevices++;
          previousPort = port;
        }
      }
    } while ( ports[++nChannels] );
    free( ports );
  }

  jack_client_close( client );
  return nDevices;
}

RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
{
  RtAudio::DeviceInfo info;
  info.probed = false;

  jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption
  jack_status_t *status = NULL;
  jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
  if ( client == 0 ) {
    errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
    error( RtError::WARNING );
    return info;
  }

  const char **ports;
  std::string port, previousPort;
  unsigned int nPorts = 0, nDevices = 0;
  ports = jack_get_ports( client, NULL, NULL, 0 );
  if ( ports ) {
    // Parse the port names up to the first colon (:).
    size_t iColon = 0;
    do {
      port = (char *) ports[ nPorts ];
      iColon = port.find(":");
      if ( iColon != std::string::npos ) {
        port = port.substr( 0, iColon );
        if ( port != previousPort ) {
          if ( nDevices == device ) info.name = port;
          nDevices++;
          previousPort = port;
        }
      }
    } while ( ports[++nPorts] );
    free( ports );
  }

  if ( device >= nDevices ) {
    errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
    error( RtError::INVALID_USE );
  }

  // Get the current jack server sample rate.
  info.sampleRates.clear();
  info.sampleRates.push_back( jack_get_sample_rate( client ) );

  // Count the available ports containing the client name as device
  // channels.  Jack "input ports" equal RtAudio output channels.
  unsigned int nChannels = 0;
  ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
  if ( ports ) {
    while ( ports[ nChannels ] ) nChannels++;
    free( ports );
    info.outputChannels = nChannels;
  }

  // Jack "output ports" equal RtAudio input channels.
  nChannels = 0;
  ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
  if ( ports ) {
    while ( ports[ nChannels ] ) nChannels++;
    free( ports );
    info.inputChannels = nChannels;
  }

  if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
    jack_client_close(client);
    errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
    error( RtError::WARNING );
    return info;
  }

  // If device opens for both playback and capture, we determine the channels.
  if ( info.outputChannels > 0 && info.inputChannels > 0 )
    info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;

  // Jack always uses 32-bit floats.
  info.nativeFormats = RTAUDIO_FLOAT32;

  // Jack doesn't provide default devices so we'll use the first available one.
  if ( device == 0 && info.outputChannels > 0 )
    info.isDefaultOutput = true;
  if ( device == 0 && info.inputChannels > 0 )
    info.isDefaultInput = true;

  jack_client_close(client);
  info.probed = true;
  return info;
}

int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
{
  CallbackInfo *info = (CallbackInfo *) infoPointer;

  RtApiJack *object = (RtApiJack *) info->object;
  if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;

  return 0;
}

// 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 | JackUseExactName ); //JackNullOption;
    jack_status_t *status = NULL;
    if ( options && !options->streamName.empty() )
      client = jack_client_open( options->streamName.c_str(), jackoptions, status );
    else
      client = jack_client_open( "RtApiJack", jackoptions, status );
    if ( client == 0 ) {
      errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
      error( RtError::WARNING );
      return FAILURE;
    }
  }
  else {
    // The handle must have been created on an earlier pass.
    client = handle->client;
  }

  const char **ports;
  std::string port, previousPort, deviceName;
  unsigned int nPorts = 0, nDevices = 0;
  ports = jack_get_ports( client, NULL, NULL, 0 );
  if ( ports ) {
    // Parse the port names up to the first colon (:).
    size_t iColon = 0;
    do {
      port = (char *) ports[ nPorts ];
      iColon = port.find(":");
      if ( iColon != std::string::npos ) {
        port = port.substr( 0, iColon );
        if ( port != previousPort ) {
          if ( nDevices == device ) deviceName = port;
          nDevices++;
          previousPort = port;
        }
      }
    } while ( ports[++nPorts] );
    free( ports );
  }

  if ( device >= nDevices ) {
    errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
    return FAILURE;
  }

  // Count the available ports containing the client name as device
  // channels.  Jack "input ports" equal RtAudio output channels.
  unsigned int nChannels = 0;
  unsigned long flag = JackPortIsInput;
  if ( mode == INPUT ) flag = JackPortIsOutput;
  ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
  if ( ports ) {
    while ( ports[ nChannels ] ) nChannels++;
    free( ports );
  }

  // Compare the jack ports for specified client to the requested number of channels.
  if ( nChannels < (channels + firstChannel) ) {
    errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
    errorText_ = errorStream_.str();
    return FAILURE;
  }

  // Check the jack server sample rate.
  unsigned int jackRate = jack_get_sample_rate( client );
  if ( sampleRate != jackRate ) {
    jack_client_close( client );
    errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
    errorText_ = errorStream_.str();
    return FAILURE;
  }
  stream_.sampleRate = jackRate;

  // Get the latency of the JACK port.
  ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
  if ( ports[ firstChannel ] )
    stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
  free( ports );

  // The jack server always uses 32-bit floating-point data.
  stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
  stream_.userFormat = format;

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

  // Jack always uses non-interleaved buffers.
  stream_.deviceInterleaved[mode] = false;

  // Jack always provides host byte-ordered data.
  stream_.doByteSwap[mode] = false;

  // Get the buffer size.  The buffer size and number of buffers
  // (periods) is set when the jack server is started.
  stream_.bufferSize = (int) jack_get_buffer_size( client );
  *bufferSize = stream_.bufferSize;

  stream_.nDeviceChannels[mode] = channels;
  stream_.nUserChannels[mode] = channels;

  // Set flags for buffer conversion.
  stream_.doConvertBuffer[mode] = false;
  if ( stream_.userFormat != stream_.deviceFormat[mode] )
    stream_.doConvertBuffer[mode] = true;
  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
       stream_.nUserChannels[mode] > 1 )
    stream_.doConvertBuffer[mode] = true;

  // Allocate our JackHandle structure for the stream.
  if ( handle == 0 ) {
    try {
      handle = new JackHandle;
    }
    catch ( std::bad_alloc& ) {
      errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
      goto error;
    }

    if ( pthread_cond_init(&handle->condition, NULL) ) {
      errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
      goto error;
    }
    stream_.apiHandle = (void *) handle;
    handle->client = client;
  }
  handle->deviceName[mode] = deviceName;

  // Allocate necessary internal buffers.
  unsigned long bufferBytes;
  bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
  if ( stream_.userBuffer[mode] == NULL ) {
    errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
    goto error;
  }

  if ( stream_.doConvertBuffer[mode] ) {

    bool makeBuffer = true;
    if ( mode == OUTPUT )
      bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
    else { // mode == INPUT
      bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
      if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
        unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
        if ( bufferBytes < bytesOut ) makeBuffer = false;
      }
    }

    if ( makeBuffer ) {
      bufferBytes *= *bufferSize;
      if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
      stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
      if ( stream_.deviceBuffer == NULL ) {
        errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
        goto error;
      }
    }
  }

  // Allocate memory for the Jack ports (channels) identifiers.
  handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
  if ( handle->ports[mode] == NULL )  {
    errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
    goto error;
  }

  stream_.device[mode] = device;
  stream_.channelOffset[mode] = firstChannel;
  stream_.state = STREAM_STOPPED;
  stream_.callbackInfo.object = (void *) this;

  if ( stream_.mode == OUTPUT && mode == INPUT )
    // We had already set up the stream for output.
    stream_.mode = DUPLEX;
  else {
    stream_.mode = mode;
    jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
    jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );
    jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
  }

  // Register our ports.
  char label[64];
  if ( mode == OUTPUT ) {
    for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
      snprintf( label, 64, "outport %d", i );
      handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
                                                JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
    }
  }
  else {
    for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
      snprintf( label, 64, "inport %d", i );
      handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
                                                JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
    }
  }

  // Setup the buffer conversion information structure.  We don't use
  // buffers to do channel offsets, so we override that parameter
  // here.
  if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );

  return SUCCESS;

 error:
  if ( handle ) {
    pthread_cond_destroy( &handle->condition );
    jack_client_close( handle->client );

    if ( handle->ports[0] ) free( handle->ports[0] );
    if ( handle->ports[1] ) free( handle->ports[1] );

    delete handle;
    stream_.apiHandle = 0;
  }

  for ( int i=0; i<2; i++ ) {
    if ( stream_.userBuffer[i] ) {
      free( stream_.userBuffer[i] );
      stream_.userBuffer[i] = 0;
    }
  }

  if ( stream_.deviceBuffer ) {
    free( stream_.deviceBuffer );
    stream_.deviceBuffer = 0;
  }

  return FAILURE;
}

void RtApiJack :: closeStream( void )
{
  if ( stream_.state == STREAM_CLOSED ) {
    errorText_ = "RtApiJack::closeStream(): no open stream to close!";
    error( RtError::WARNING );
    return;
  }

  JackHandle *handle = (JackHandle *) stream_.apiHandle;
  if ( handle ) {

    if ( stream_.state == STREAM_RUNNING )
      jack_deactivate( handle->client );

    jack_client_close( handle->client );
  }

  if ( handle ) {
    if ( handle->ports[0] ) free( handle->ports[0] );
    if ( handle->ports[1] ) free( handle->ports[1] );
    pthread_cond_destroy( &handle->condition );
    delete handle;
    stream_.apiHandle = 0;
  }

  for ( int i=0; i<2; i++ ) {
    if ( stream_.userBuffer[i] ) {
      free( stream_.userBuffer[i] );
      stream_.userBuffer[i] = 0;
    }
  }

  if ( stream_.deviceBuffer ) {
    free( stream_.deviceBuffer );
    stream_.deviceBuffer = 0;
  }

  stream_.mode = UNINITIALIZED;
  stream_.state = STREAM_CLOSED;
}

void RtApiJack :: startStream( void )
{
  verifyStream();
  if ( stream_.state == STREAM_RUNNING ) {
    errorText_ = "RtApiJack::startStream(): the stream is already running!";
    error( RtError::WARNING );
    return;
  }

  MUTEX_LOCK(&stream_.mutex);

  JackHandle *handle = (JackHandle *) stream_.apiHandle;
  int result = jack_activate( handle->client );
  if ( result ) {
    errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
    goto unlock;
  }

  const char **ports;

  // Get the list of available ports.
  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
    result = 1;
    ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);
    if ( ports == NULL) {
      errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
      goto unlock;
    }

    // Now make the port connections.  Since RtAudio wasn't designed to
    // allow the user to select particular channels of a device, we'll
    // just open the first "nChannels" ports with offset.
    for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
      result = 1;
      if ( ports[ stream_.channelOffset[0] + i ] )
        result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
      if ( result ) {
        free( ports );
        errorText_ = "RtApiJack::startStream(): error connecting output ports!";
        goto unlock;
      }
    }
    free(ports);
  }

  if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
    result = 1;
    ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );
    if ( ports == NULL) {
      errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
      goto unlock;
    }

    // Now make the port connections.  See note above.
    for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
      result = 1;
      if ( ports[ stream_.channelOffset[1] + i ] )
        result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
      if ( result ) {
        free( ports );
        errorText_ = "RtApiJack::startStream(): error connecting input ports!";
        goto unlock;
      }
    }
    free(ports);
  }

  handle->drainCounter = 0;
  handle->internalDrain = false;
  stream_.state = STREAM_RUNNING;

 unlock:
  MUTEX_UNLOCK(&stream_.mutex);

  if ( result == 0 ) return;
  error( RtError::SYSTEM_ERROR );
}

void RtApiJack :: stopStream( void )
{
  verifyStream();
  if ( stream_.state == STREAM_STOPPED ) {
    errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
    error( RtError::WARNING );
    return;
  }

  MUTEX_LOCK( &stream_.mutex );

  if ( stream_.state == STREAM_STOPPED ) {
    MUTEX_UNLOCK( &stream_.mutex );
    return;
  }

  JackHandle *handle = (JackHandle *) stream_.apiHandle;
  if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {

    if ( handle->drainCounter == 0 ) {
      handle->drainCounter = 1;
      pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
    }
  }

  jack_deactivate( handle->client );
  stream_.state = STREAM_STOPPED;

  MUTEX_UNLOCK( &stream_.mutex );
}

void RtApiJack :: abortStream( void )
{
  verifyStream();
  if ( stream_.state == STREAM_STOPPED ) {
    errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
    error( RtError::WARNING );
    return;
  }

  JackHandle *handle = (JackHandle *) stream_.apiHandle;
  handle->drainCounter = 1;

  stopStream();
}

// This function will be called by a spawned thread when the user
// callback function signals that the stream should be stopped or
// aborted.  It is necessary to handle it this way because the
// callbackEvent() function must return before the jack_deactivate()
// function will return.
extern "C" void *jackStopStream( void *ptr )
{
  CallbackInfo *info = (CallbackInfo *) ptr;
  RtApiJack *object = (RtApiJack *) info->object;

  object->stopStream();

  pthread_exit( NULL );
}

bool RtApiJack :: callbackEvent( unsigned long nframes )
{
  if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
  if ( stream_.state == STREAM_CLOSED ) {
    errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
    error( RtError::WARNING );
    return FAILURE;
  }
  if ( stream_.bufferSize != nframes ) {
    errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
    error( RtError::WARNING );
    return FAILURE;
  }

  CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
  JackHandle *handle = (JackHandle *) stream_.apiHandle;

  // Check if we were draining the stream and signal is finished.
  if ( handle->drainCounter > 3 ) {
    if ( handle->internalDrain == true ) {
      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 upper three bytes of a 32-bit integer.

  // Clear our device buffer when in/out duplex device channels are different
  if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
       ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
    memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );

  int j;
  if (info.outFormat == RTAUDIO_FLOAT64) {
    Float64 scale;
    Float64 *out = (Float64 *)outBuffer;

    if (info.inFormat == RTAUDIO_SINT8) {
      signed char *in = (signed char *)inBuffer;
      scale = 1.0 / 127.5;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
          out[info.outOffset[j]] += 0.5;
          out[info.outOffset[j]] *= scale;
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_SINT16) {
      Int16 *in = (Int16 *)inBuffer;
      scale = 1.0 / 32767.5;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
          out[info.outOffset[j]] += 0.5;
          out[info.outOffset[j]] *= scale;
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_SINT24) {
      Int32 *in = (Int32 *)inBuffer;
      scale = 1.0 / 8388607.5;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]] & 0x00ffffff);
          out[info.outOffset[j]] += 0.5;
          out[info.outOffset[j]] *= scale;
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_SINT32) {
      Int32 *in = (Int32 *)inBuffer;
      scale = 1.0 / 2147483647.5;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
          out[info.outOffset[j]] += 0.5;
          out[info.outOffset[j]] *= scale;
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_FLOAT32) {
      Float32 *in = (Float32 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_FLOAT64) {
      // Channel compensation and/or (de)interleaving only.
      Float64 *in = (Float64 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = in[info.inOffset[j]];
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
  }
  else if (info.outFormat == RTAUDIO_FLOAT32) {
    Float32 scale;
    Float32 *out = (Float32 *)outBuffer;

    if (info.inFormat == RTAUDIO_SINT8) {
      signed char *in = (signed char *)inBuffer;
      scale = (Float32) ( 1.0 / 127.5 );
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
          out[info.outOffset[j]] += 0.5;
          out[info.outOffset[j]] *= scale;
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_SINT16) {
      Int16 *in = (Int16 *)inBuffer;
      scale = (Float32) ( 1.0 / 32767.5 );
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
          out[info.outOffset[j]] += 0.5;
          out[info.outOffset[j]] *= scale;
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_SINT24) {
      Int32 *in = (Int32 *)inBuffer;
      scale = (Float32) ( 1.0 / 8388607.5 );
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]] & 0x00ffffff);
          out[info.outOffset[j]] += 0.5;
          out[info.outOffset[j]] *= scale;
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_SINT32) {
      Int32 *in = (Int32 *)inBuffer;
      scale = (Float32) ( 1.0 / 2147483647.5 );
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
          out[info.outOffset[j]] += 0.5;
          out[info.outOffset[j]] *= scale;
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_FLOAT32) {
      // Channel compensation and/or (de)interleaving only.
      Float32 *in = (Float32 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = in[info.inOffset[j]];
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_FLOAT64) {
      Float64 *in = (Float64 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
  }
  else if (info.outFormat == RTAUDIO_SINT32) {
    Int32 *out = (Int32 *)outBuffer;
    if (info.inFormat == RTAUDIO_SINT8) {
      signed char *in = (signed char *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
          out[info.outOffset[j]] <<= 24;
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_SINT16) {
      Int16 *in = (Int16 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
          out[info.outOffset[j]] <<= 16;
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_SINT24) {
      Int32 *in = (Int32 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
          out[info.outOffset[j]] <<= 8;
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_SINT32) {
      // Channel compensation and/or (de)interleaving only.
      Int32 *in = (Int32 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = in[info.inOffset[j]];
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_FLOAT32) {
      Float32 *in = (Float32 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_FLOAT64) {
      Float64 *in = (Float64 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
  }
  else if (info.outFormat == RTAUDIO_SINT24) {
    Int32 *out = (Int32 *)outBuffer;
    if (info.inFormat == RTAUDIO_SINT8) {
      signed char *in = (signed char *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
          out[info.outOffset[j]] <<= 16;
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_SINT16) {
      Int16 *in = (Int16 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
          out[info.outOffset[j]] <<= 8;
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_SINT24) {
      // Channel compensation and/or (de)interleaving only.
      Int32 *in = (Int32 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = in[info.inOffset[j]];
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_SINT32) {
      Int32 *in = (Int32 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
          out[info.outOffset[j]] >>= 8;
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_FLOAT32) {
      Float32 *in = (Float32 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_FLOAT64) {
      Float64 *in = (Float64 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
  }
  else if (info.outFormat == RTAUDIO_SINT16) {
    Int16 *out = (Int16 *)outBuffer;
    if (info.inFormat == RTAUDIO_SINT8) {
      signed char *in = (signed char *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
          out[info.outOffset[j]] <<= 8;
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_SINT16) {
      // Channel compensation and/or (de)interleaving only.
      Int16 *in = (Int16 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = in[info.inOffset[j]];
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_SINT24) {
      Int32 *in = (Int32 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 8) & 0x0000ffff);
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_SINT32) {
      Int32 *in = (Int32 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_FLOAT32) {
      Float32 *in = (Float32 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_FLOAT64) {
      Float64 *in = (Float64 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
  }
  else if (info.outFormat == RTAUDIO_SINT8) {
    signed char *out = (signed char *)outBuffer;
    if (info.inFormat == RTAUDIO_SINT8) {
      // Channel compensation and/or (de)interleaving only.
      signed char *in = (signed char *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = in[info.inOffset[j]];
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    if (info.inFormat == RTAUDIO_SINT16) {
      Int16 *in = (Int16 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_SINT24) {
      Int32 *in = (Int32 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 16) & 0x000000ff);
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_SINT32) {
      Int32 *in = (Int32 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_FLOAT32) {
      Float32 *in = (Float32 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
    else if (info.inFormat == RTAUDIO_FLOAT64) {
      Float64 *in = (Float64 *)inBuffer;
      for (unsigned int i=0; i<stream_.bufferSize; i++) {
        for (j=0; j<info.channels; j++) {
          out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
        }
        in += info.inJump;
        out += info.outJump;
      }
    }
  }
}

  //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
  //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
  //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }

void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
{
  register char val;
  register char *ptr;

  ptr = buffer;
  if ( format == RTAUDIO_SINT16 ) {
    for ( unsigned int i=0; i<samples; i++ ) {
      // Swap 1st and 2nd bytes.
      val = *(ptr);
      *(ptr) = *(ptr+1);
      *(ptr+1) = val;

      // Increment 2 bytes.
      ptr += 2;
    }
  }
  else if ( format == RTAUDIO_SINT24 ||
            format == RTAUDIO_SINT32 ||
            format == RTAUDIO_FLOAT32 ) {
    for ( unsigned int i=0; i<samples; i++ ) {
      // Swap 1st and 4th bytes.
      val = *(ptr);
      *(ptr) = *(ptr+3);
      *(ptr+3) = val;

      // Swap 2nd and 3rd bytes.
      ptr += 1;
      val = *(ptr);
      *(ptr) = *(ptr+1);
      *(ptr+1) = val;

      // Increment 3 more bytes.
      ptr += 3;
    }
  }
  else if ( format == RTAUDIO_FLOAT64 ) {
    for ( unsigned int i=0; i<samples; i++ ) {
      // Swap 1st and 8th bytes
      val = *(ptr);
      *(ptr) = *(ptr+7);
      *(ptr+7) = val;

      // Swap 2nd and 7th bytes
      ptr += 1;
      val = *(ptr);
      *(ptr) = *(ptr+5);
      *(ptr+5) = val;

      // Swap 3rd and 6th bytes
      ptr += 1;
      val = *(ptr);
      *(ptr) = *(ptr+3);
      *(ptr+3) = val;

      // Swap 4th and 5th bytes
      ptr += 1;
      val = *(ptr);
      *(ptr) = *(ptr+1);
      *(ptr+1) = val;

      // Increment 5 more bytes.
      ptr += 5;
    }
  }
}

  // Indentation settings for Vim and Emacs
  //
  // Local Variables:
  // c-basic-offset: 2
  // indent-tabs-mode: nil
  // End:
  //
  // vim: et sts=2 sw=2