RtAudio.cpp 270 KB
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/************************************************************************/
/*! \class RtAudio
    \brief Realtime audio i/o C++ classes.

    RtAudio provides a common API (Application Programming Interface)
    for realtime audio input/output across Linux (native ALSA, Jack,
    and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
    (DirectSound and ASIO) operating systems.

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

    RtAudio: realtime audio i/o C++ classes
    Copyright (c) 2001-2010 Gary P. Scavone

    Permission is hereby granted, free of charge, to any person
    obtaining a copy of this software and associated documentation files
    (the "Software"), to deal in the Software without restriction,
    including without limitation the rights to use, copy, modify, merge,
    publish, distribute, sublicense, and/or sell copies of the Software,
    and to permit persons to whom the Software is furnished to do so,
    subject to the following conditions:

    The above copyright notice and this permission notice shall be
    included in all copies or substantial portions of the Software.

    Any person wishing to distribute modifications to the Software is
    asked to send the modifications to the original developer so that
    they can be incorporated into the canonical version.  This is,
    however, not a binding provision of this license.

    THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
    EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
    MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
    IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
    ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
    CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
    WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
/************************************************************************/

// RtAudio: Version 4.0.7

#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;
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